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[alsa-devel] [PATCHv3 3/7] ASoC: TWL6030: Manual power-up/down sequences
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

TWL6030 codec device can be powered-up/down through a specific register writes sequence. These sequences can be used when no gpio line is provided for AUDPWRON. When the codec is powered-up in this way, automatic power-down sequence (triggered by thermal shutdown) is not possible. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 112 ++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/twl6030.h | 16 ++++++ 2 files changed, 115 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 3c6c540..f97dec2 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -244,6 +244,88 @@ static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) } } +/* twl6030 codec manual power-up sequence */ +static void twl6030_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6030_NCPENA | TWL6030_NCPOPEN; + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); +} + +/* twl6030 codec manual power-down sequence */ +static void twl6030_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6030_NCPENA | TWL6030_NCPOPEN); + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -480,12 +562,15 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-up sequence latency */ mdelay(16); - } - /* sync registers updated during power-up sequence */ - twl6030_read(codec, TWL6030_REG_NCPCTL); - twl6030_read(codec, TWL6030_REG_LDOCTL); - twl6030_read(codec, TWL6030_REG_LPPLLCTL); + /* sync registers updated during power-up sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6030_power_up(codec); + } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); @@ -502,12 +587,16 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-down sequence latency */ udelay(500); - } - /* sync registers updated during power-down sequence */ - twl6030_read(codec, TWL6030_REG_NCPCTL); - twl6030_read(codec, TWL6030_REG_LDOCTL); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + /* sync registers updated during power-down sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6030_power_down(codec); + } priv->codec_powered = 0; break; @@ -723,9 +812,6 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) goto gpio2_err; priv->codec_powered = 0; - } else { - /* if no gpio is provided, then assume its always on */ - priv->codec_powered = 1; } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index d99d469..7375ae8 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,18 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* NCPCTL (0x05) fields */ + +#define TWL6030_NCPENA 0x01 +#define TWL6030_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6030_LSLDOENA 0x01 +#define TWL6030_HSLDOENA 0x04 +#define TWL6030_REFENA 0x40 +#define TWL6030_OSCENA 0x80 + /* HPPLLCTL (0x07) fields */ #define TWL6030_HPLLENA 0x01 @@ -88,6 +100,10 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +/* ACCCTL (0x2D) fields */ + +#define TWL6030_RESETSPLIT 0x04 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.5.4.3

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[alsa-devel] [PATCHv3 2/7] ASoC: TWL6030: Add twl6030 codec driver
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

Initial version of TWL6030 codec driver. The TWL6030 codec uses a propietary PDM-based digital audio interface. Audio paths supported are: - Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right - Output: Headset Left/Right, Handsfree Left/Right Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 824 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 94 +++++ 4 files changed, 924 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..0effb52 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6030 if TWL6030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 @@ -114,6 +115,9 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TWL4030 tristate +config SND_SOC_TWL6030 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..b70c8a1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -13,6 +13,7 @@ snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-twl6030-objs := twl6030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -50,6 +51,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TWL6030) += snd-soc-twl6030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c new file mode 100644 index 0000000..3c6c540 --- /dev/null +++ b/sound/soc/codecs/twl6030.c @@ -0,0 +1,824 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl6030.h" + +#define TWL6030_RATES (SNDRV_PCM_RATE_48000) +#define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6030_data { + struct snd_soc_codec codec; + int audpwron; + int codec_powered; +}; + +/* + * twl6030 register cache & default register settings + */ +static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6030_ASICID (ro) 0x01 */ + 0x00, /* TWL6030_ASICREV (ro) 0x02 */ + 0x00, /* TWL6030_INTID 0x03 */ + 0x7B, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_NCPCTRL 0x05 */ + 0x00, /* TWL6030_LDOCTL 0x06 */ + 0x00, /* TWL6030_HPPLLCTL 0x07 */ + 0x00, /* TWL6030_LPPLLCTL 0x08 */ + 0x00, /* TWL6030_LPPLLDIV 0x09 */ + 0x00, /* TWL6030_AMICBCTL 0x0A */ + 0x00, /* TWL6030_DMICBCTL 0x0B */ + 0x18, /* TWL6030_MICLCTL 0x0C */ + 0x18, /* TWL6030_MICRCTL 0x0D */ + 0x00, /* TWL6030_MICGAIN 0x0E */ + 0x1B, /* TWL6030_LINEGAIN 0x0F */ + 0x00, /* TWL6030_HSLCTL 0x10 */ + 0x00, /* TWL6030_HSRCTL 0x11 */ + 0x00, /* TWL6030_HSGAIN 0x12 */ + 0x06, /* TWL6030_EARCTL 0x13 */ + 0x00, /* TWL6030_HFLCTL 0x14 */ + 0x03, /* TWL6030_HFLGAIN 0x15 */ + 0x00, /* TWL6030_HFRCTL 0x16 */ + 0x03, /* TWL6030_HFRGAIN 0x17 */ + 0x00, /* TWL6030_VIBCTLL 0x18 */ + 0x00, /* TWL6030_VIBDATL 0x19 */ + 0x00, /* TWL6030_VIBCTLR 0x1A */ + 0x00, /* TWL6030_VIBDATR 0x1B */ + 0x00, /* TWL6030_HKCTL1 0x1C */ + 0x00, /* TWL6030_HKCTL2 0x1D */ + 0x00, /* TWL6030_GPOCTL 0x1E */ + 0x00, /* TWL6030_ALB 0x1F */ + 0x00, /* TWL6030_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6030_TRIM1 0x28 */ + 0x00, /* TWL6030_TRIM2 0x29 */ + 0x00, /* TWL6030_TRIM3 0x2A */ + 0x00, /* TWL6030_HSOTRIM 0x2B */ + 0x00, /* TWL6030_HFOTRIM 0x2C */ + 0x09, /* TWL6030_ACCCTL 0x2D */ + 0x00, /* TWL6030_STATUS (ro) 0x2E */ +}; + +/* + * twl6030 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6030_vio_reg[TWL6030_VIOREGNUM] = { + TWL6030_REG_ASICID, + TWL6030_REG_ASICREV, + TWL6030_REG_INTID, + TWL6030_REG_INTMR, + TWL6030_REG_NCPCTL, + TWL6030_REG_LDOCTL, + TWL6030_REG_AMICBCTL, + TWL6030_REG_DMICBCTL, + TWL6030_REG_HKCTL1, + TWL6030_REG_HKCTL2, + TWL6030_REG_GPOCTL, + TWL6030_REG_TRIM1, + TWL6030_REG_TRIM2, + TWL6030_REG_TRIM3, + TWL6030_REG_HSOTRIM, + TWL6030_REG_HFOTRIM, + TWL6030_REG_ACCCTL, + TWL6030_REG_STATUS, +}; + +/* + * twl6030 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6030_vdd_reg[TWL6030_VDDREGNUM] = { + TWL6030_REG_HPPLLCTL, + TWL6030_REG_LPPLLCTL, + TWL6030_REG_LPPLLDIV, + TWL6030_REG_MICLCTL, + TWL6030_REG_MICRCTL, + TWL6030_REG_MICGAIN, + TWL6030_REG_LINEGAIN, + TWL6030_REG_HSLCTL, + TWL6030_REG_HSRCTL, + TWL6030_REG_HSGAIN, + TWL6030_REG_EARCTL, + TWL6030_REG_HFLCTL, + TWL6030_REG_HFLGAIN, + TWL6030_REG_HFRCTL, + TWL6030_REG_HFRGAIN, + TWL6030_REG_VIBCTLL, + TWL6030_REG_VIBDATL, + TWL6030_REG_VIBCTLR, + TWL6030_REG_VIBDATR, + TWL6030_REG_ALB, + TWL6030_REG_DLB, +}; + +/* + * read twl6030 register cache + */ +static inline unsigned int twl6030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6030 register cache + */ +static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6030 hardware register + */ +static int twl6030_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &value, reg); + twl6030_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6030 register space + */ +static int twl6030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + twl6030_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL6030_MODULE_AUDIO, value, reg); +} + +static void twl6030_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6030_write(codec, TWL6030_REG_ACCCTL, cache[TWL6030_REG_ACCCTL]); + + for (i = 0; i < TWL6030_VIOREGNUM; i++) { + reg = twl6030_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + if ((reg == TWL6030_REG_ASICID) || + (reg == TWL6030_REG_ASICREV) || + (reg == TWL6030_REG_STATUS)) + continue; + twl6030_write(codec, reg, cache[reg]); + } +} + +static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6030_VDDREGNUM; i++) { + reg = twl6030_vdd_reg[i]; + twl6030_write(codec, reg, cache[reg]); + } +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6030_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6030_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6030_enum[] = { + SOC_ENUM_SINGLE(TWL6030_REG_MICLCTL, 3, 3, twl6030_amicl_texts), + SOC_ENUM_SINGLE(TWL6030_REG_MICRCTL, 3, 3, twl6030_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6030_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6030_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6030_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6030_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6030_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6030_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6030_REG_HFLGAIN, TWL6030_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6030_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6030_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6030_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6030_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6030_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6030_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6030_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6030_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6030_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6030_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6030_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6030_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6030_dapm_widgets, + ARRAY_SIZE(twl6030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6030_data *priv = codec->private_data; + int audpwron = priv->audpwron; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* power-up sequence latency */ + mdelay(16); + } + + /* sync registers updated during power-up sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + + /* initialize vdd/vss registers with reg_cache */ + twl6030_init_vdd_regs(codec); + + priv->codec_powered = 1; + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 0); + + /* power-down sequence latency */ + udelay(500); + } + + /* sync registers updated during power-down sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + return 0; +} + +static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 hppllctl, lppllctl; + + hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + hppllctl &= ~TWL6030_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_12000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6030_MCLK_19200KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_26000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6030_MCLK_38400KHZ | + TWL6030_HPLLSQRENA | + TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown sysclk freq %d\n", freq); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + + /* Disable LPPLL and select HPPLL */ + lppllctl = TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static struct snd_soc_dai_ops twl6030_dai_ops = { + .set_sysclk = twl6030_set_dai_sysclk, +}; + +struct snd_soc_dai twl6030_dai = { + .name = "twl6030", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .ops = &twl6030_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6030_dai); + +static int twl6030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6030_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *twl6030_codec; + +static int twl6030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6030_codec); + + codec = twl6030_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6030_snd_controls, + ARRAY_SIZE(twl6030_snd_controls)); + twl6030_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6030 = { + .probe = twl6030_probe, + .remove = twl6030_remove, + .suspend = twl6030_suspend, + .resume = twl6030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6030); + +static int __devinit twl6030_codec_probe(struct platform_device *pdev) +{ + struct twl_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6030_data *priv; + int audpwron; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) + audpwron = twl_codec->audpwron_gpio; + else + audpwron = -EINVAL; + + priv->audpwron = audpwron; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6030_dai.dev = &pdev->dev; + + codec->name = "twl6030"; + codec->owner = THIS_MODULE; + codec->read = twl6030_read_reg_cache; + codec->write = twl6030_write; + codec->set_bias_level = twl6030_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6030_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6030_reg); + codec->reg_cache = kmemdup(twl6030_reg, sizeof(twl6030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (gpio_is_valid(audpwron)) { + ret = gpio_request(audpwron, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } else { + /* if no gpio is provided, then assume its always on */ + priv->codec_powered = 1; + } + + /* init vio registers */ + twl6030_init_vio_regs(codec); + + /* power on device */ + ret = twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto gpio2_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6030_codec = codec; + + ret = snd_soc_register_dai(&twl6030_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6030_codec = NULL; +reg_err: + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); +gpio2_err: + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6030_codec_remove(struct platform_device *pdev) +{ + struct twl6030_data *priv = twl6030_codec->private_data; + int audpwron = priv->audpwron; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + snd_soc_unregister_dai(&twl6030_dai); + snd_soc_unregister_codec(twl6030_codec); + + kfree(twl6030_codec); + twl6030_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int twl6030_codec_suspend(struct platform_device *pdev, + pm_message_t msg) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int twl6030_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define twl6030_codec_suspend NULL +#define twl6030_codec_resume NULL +#endif + +static struct platform_driver twl6030_codec_driver = { + .driver = { + .name = "twl6030_codec", + .owner = THIS_MODULE, + }, + .probe = twl6030_codec_probe, + .remove = __devexit_p(twl6030_codec_remove), + .suspend = twl6030_codec_suspend, + .resume = twl6030_codec_resume, +}; + +static int __init twl6030_codec_init(void) +{ + return platform_driver_register(&twl6030_codec_driver); +} +module_init(twl6030_codec_init); + +static void __exit twl6030_codec_exit(void) +{ + platform_driver_unregister(&twl6030_codec_driver); +} +module_exit(twl6030_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6030 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h new file mode 100644 index 0000000..d99d469 --- /dev/null +++ b/sound/soc/codecs/twl6030.h @@ -0,0 +1,94 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL6030_H__ +#define __TWL6030_H__ + +#define TWL6030_REG_ASICID 0x01 +#define TWL6030_REG_ASICREV 0x02 +#define TWL6030_REG_INTID 0x03 +#define TWL6030_REG_INTMR 0x04 +#define TWL6030_REG_NCPCTL 0x05 +#define TWL6030_REG_LDOCTL 0x06 +#define TWL6030_REG_HPPLLCTL 0x07 +#define TWL6030_REG_LPPLLCTL 0x08 +#define TWL6030_REG_LPPLLDIV 0x09 +#define TWL6030_REG_AMICBCTL 0x0A +#define TWL6030_REG_DMICBCTL 0x0B +#define TWL6030_REG_MICLCTL 0x0C +#define TWL6030_REG_MICRCTL 0x0D +#define TWL6030_REG_MICGAIN 0x0E +#define TWL6030_REG_LINEGAIN 0x0F +#define TWL6030_REG_HSLCTL 0x10 +#define TWL6030_REG_HSRCTL 0x11 +#define TWL6030_REG_HSGAIN 0x12 +#define TWL6030_REG_EARCTL 0x13 +#define TWL6030_REG_HFLCTL 0x14 +#define TWL6030_REG_HFLGAIN 0x15 +#define TWL6030_REG_HFRCTL 0x16 +#define TWL6030_REG_HFRGAIN 0x17 +#define TWL6030_REG_VIBCTLL 0x18 +#define TWL6030_REG_VIBDATL 0x19 +#define TWL6030_REG_VIBCTLR 0x1A +#define TWL6030_REG_VIBDATR 0x1B +#define TWL6030_REG_HKCTL1 0x1C +#define TWL6030_REG_HKCTL2 0x1D +#define TWL6030_REG_GPOCTL 0x1E +#define TWL6030_REG_ALB 0x1F +#define TWL6030_REG_DLB 0x20 +#define TWL6030_REG_TRIM1 0x28 +#define TWL6030_REG_TRIM2 0x29 +#define TWL6030_REG_TRIM3 0x2A +#define TWL6030_REG_HSOTRIM 0x2B +#define TWL6030_REG_HFOTRIM 0x2C +#define TWL6030_REG_ACCCTL 0x2D +#define TWL6030_REG_STATUS 0x2E + +#define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) + +#define TWL6030_VIOREGNUM 18 +#define TWL6030_VDDREGNUM 21 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6030_HPLLENA 0x01 +#define TWL6030_HPLLRST 0x02 +#define TWL6030_HPLLBP 0x04 +#define TWL6030_HPLLSQRENA 0x08 +#define TWL6030_HPLLSQRBP 0x10 +#define TWL6030_MCLK_12000KHZ (0 << 5) +#define TWL6030_MCLK_19200KHZ (1 << 5) +#define TWL6030_MCLK_26000KHZ (2 << 5) +#define TWL6030_MCLK_38400KHZ (3 << 5) +#define TWL6030_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6030_LPLLENA 0x01 +#define TWL6030_LPLLRST 0x02 +#define TWL6030_LPLLSEL 0x04 +#define TWL6030_LPLLFIN 0x08 +#define TWL6030_HPLLSEL 0x10 + +extern struct snd_soc_dai twl6030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6030; + +#endif /* End of __TWL6030_H__ */ -- 1.5.4.3

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[alsa-devel] [PATCHv3 0/7] TWL6030 audio codec initial support
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

Following patch series adds initial support for TWL6030 codec driver. Changes from v2: - CODEC now handles scenarios where no platform data is passed - Removed GPIOLIB dependency of codec driver - Removed magic values, register reads instead - Added runtime constraints depending on sysclk - Added more constraints for switching to low-power mode - Use request_threaded_irq instead (oneshot support required) - wait_for_completion with timeout - Changelog now gives a better explanation of features and corresponding changes Thanks, -Misa --- Misael Lopez Cruz (7): OMAP4: PMIC: Add support for twl6030 codec ASoC: TWL6030: Add twl6030 codec driver ASoC: TWL6030: Manual power-up/down sequences ASoC: TWL6030: Add support for low-power PLL ASoC: TWL6030: Add restrictions for low-power playback mode ASoC: TWL6030: Enable audio interrupt ASoC: TWL6030: Detect power-up sequence completion drivers/mfd/twl-core.c | 15 + include/linux/i2c/twl.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 1242 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 141 +++++ 6 files changed, 1411 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h

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[alsa-devel] [PATCH 18/31] HDA patch_via.c: Add 48K sample rate limit for S/PDIF streams.
by Li Bo 06 Oct '09

06 Oct '09

[ALSA] HDA VIA: Add 48K sample rate limit for S/PDIF streams. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:02.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:04.000000000 +0800 @@ -1228,6 +1228,7 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -1241,6 +1242,7 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, }; static int via_build_controls(struct hda_codec *codec) @@ -2042,6 +2044,7 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -2657,6 +2660,7 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -3137,6 +3141,7 @@ .substreams = 2, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -3581,6 +3586,7 @@ .substreams = 2, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open,

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[alsa-devel] [PATCH 0/31] HDA patch_via.c update
by Li Bo 06 Oct '09

06 Oct '09

Hi! Following are a series of patches for patch_via.c, started from git clone of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 recently. Please have a review and give comments, thank you! Outlook Access Web cannot send plain-text mail, so my gmail is used instead. You can also contact with me with LoganLi(a)viatech.com.cn, or mail to HaraldWelte(a)viatech.com and lydiawang(a)viatech.com.cn.

3 3

[alsa-devel] Bug report, multichannel LPCM over nvidia HDMI not working
by Chris Howells 05 Oct '09

05 Oct '09

Hi, I am trying to get multi-channel LPCM working over HDMI, from an Acer Revo (Nvidia Ion) to a Sony STR-DH800 amp. 2 channel LPCM is working perfectly, as is bitstreaming AC3 and DTS (with xbmc). The latest I've tried is with kernel 2.6.31.1. I am testing with speaker-test. I have found that: 'speaker-test -D hdmi -c2' works perfect. The amp lights up show 2 channel LPCM. Sound is from the front left and front right. http://chrishowells.co.uk/speakertest2ch.jpg 'speaker-test -D hdmi -c6' or (-c8) does not work. The amp realises that there is some PCM sounds, but the amp does not seem to recognise the stream properly, *however*, sound does come out of the left and right. http://chrishowells.co.uk/speakertest6ch.jpg The version of speaker-test is from Alsa 1.0.20. Here's the results of running alsa-info.sh http://www.alsa-project.org/db/?f=d90ca1ce07bf20af3241114b11fb31f79ed63fec The 'Misc = NO_PRESENCE' struck me as being a bit odd. The Revo has been upgraded to the A2 bios (didn't work with A1 either), and the Nvidia drivers are 190.32. I've spent quite a few days trying to get this work, including alsa 1.0.20 and 1.0.21 on ubuntu 9.04 with and without the 8ch patch, Ubuntu karmic beta standard, as well as patching alsa 1.0.20 it with the 8ch patch. Most grateful for any help.

1 0

[alsa-devel] [PATCH] sscape: force AD1848 codec mode on old Soundscape
by Krzysztof Helt 05 Oct '09

05 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Old Soundscape cards (pre PnP) work only with AD1848 codecs. If the CS4231 codec is installed it must be used in AD1848 compatible mode. Also, add gameport support and remove an unused mpu field. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The second version: do not set joystick variable to zeros (false). BTW. The OSS sscape from the 2.6 kernel cannot handle my Media FX card. It seems that only Soundscape PnP may work (after fixing existing bugs). I prefer removing it completely. diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index cf98525..6de56d1 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1639,6 +1639,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. mpu_irq - MPU-401 IRQ # (PnP setup) dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index c739374..e348f8b 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -54,6 +54,7 @@ static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS] __devinitdata; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); @@ -79,6 +80,9 @@ MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); + #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -145,7 +149,6 @@ struct soundscape { struct resource *io_res; struct resource *wss_res; struct snd_wss *chip; - struct snd_mpu401 *mpu; unsigned char midi_vol; }; @@ -815,7 +818,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; - sscape->mpu = mpu; initialise_mpu401(mpu); } @@ -836,12 +838,30 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, register struct soundscape *sscape = get_card_soundscape(card); struct snd_wss *chip; int err; + int codec_type = WSS_HW_DETECT; + + switch (sscape->type) { + case MEDIA_FX: + case SSCAPE: + /* + * There are some freak examples of early Soundscape cards + * with CS4231 instead of AD1848/CS4248. Unfortunately, the + * CS4231 works only in CS4248 compatibility mode on + * these cards so force it. + */ + if (sscape->ic_type != IC_OPUS) + codec_type = WSS_HW_AD1848; + break; - if (sscape->type == SSCAPE_VIVO) + case SSCAPE_VIVO: port += 4; + break; + default: + break; + } err = snd_wss_create(card, port, -1, irq, dma1, dma2, - WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); + codec_type, WSS_HWSHARE_DMA1, &chip); if (!err) { unsigned long flags; struct snd_pcm *pcm; @@ -927,6 +947,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; + int val; const char *name; /* @@ -1026,6 +1047,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); mpu_irq_cfg |= mpu_irq_cfg << 2; + val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7; + if (joystick[dev]) + val |= 8; + sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10); sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT -- 1.6.0.3 ---------------------------------------------------------------------- Nie boisz sie wampirow! A wi�c Graj >>> http://link.interia.pl/f2390

2 1

[alsa-devel] step-by-step usb-snd-audio instructions?
by Anna Keenan 05 Oct '09

05 Oct '09

Hi there ALSA people, I'm very new to Linux, running an eee pc 1000H withLinux Easypeasy 1.5 and the new skype beta. I love my USB headset - I'm frequently on international skype conference calls and on other operating systems I've found that nothing else gives me a clear sound. But I am apparently missing the usb-snd-audio driver, as it doesn't come packaged with Easypeasy, so my beloved headset won't function. Skype audio is generally also really crackly when i'm using the normal microphone and speakers, but i think that's a different problem. Does anyone out there have step-by-step instructions on how to add the snd-usb-audio driver? I've found no user-friendly 'how to' online, despite hours trawling (and posting on) forums... I would *so* love some help or even a response telling where else I should be looking for help... Anna K -- Anna Keenan Youth Climate Advocate +31 6 45043 155 (Netherlands) +61 (0)4 1979 2263 (Global) Skype ID: anna.c.keenan "If you don't change course, you'll end up where you're heading." www.itsgettinghotinhere.org http://climatechangeperspectives.blogspot.com

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 05 Oct '09

05 Oct '09

The following changes since commit 834eb6c599a8efa1fe9b77d469562e0c78c876e1: Mark Brown (1): Merge remote branch 'takashi/fix/asoc' into for-2.6.32 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 Jonathan Cameron (1): ASoC: wm8940: Fix check on error code form snd_soc_codec_set_cache_io Peter Ujfalusi (1): ASoC: Fix SND_SOC_DAPM_LINE handling sound/soc/codecs/wm8940.c | 2 +- sound/soc/soc-dapm.c | 5 +++-- 2 files changed, 4 insertions(+), 3 deletions(-)

1 0

Re: [alsa-devel] Asterisk and ALSA/Jack
by Pavel Hofman 05 Oct '09

05 Oct '09

Fabien COMTE wrote: > Hi, > > I want to use Asterisk with ALSA / Jack audio. > > I tried the next configurations : > > 1 ) app_jack.so -> does not work. > > 2 ) chan_alsa.so without jackd started > my .asoundrc : > pcm.!default > { > type hw > card 1 > } > pcm.jack1 { > type jack > playback_ports { > 0 alsa_pcm:playback_1 > 1 alsa_pcm:playback_2 > } > capture_ports { > 0 alsa_pcm:capture_1 > 1 alsa_pcm:capture_2 > } > } > > pcm.jack1b { > type plug > device jack1b > server linux > > > slave { > pcm "jack1" > rate 8000 > format FLOAT > channels 2 > } > } > > ctl.mixer0 { > type hw > card 1 > } > > in alsa.conf : > input_device=default > output_device=default > > That works fine. > > 3 ) chan_alsa.so with jackd started > Jackd launch command : > jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 > --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 > --outchannels 2 --dither triangular & > > my .asoundrc : > > pcm.!default > { > type plug > slave { > pcm "jack1" > rate 8000 > format FLOAT > channels 2 > } > } > pcm.jack1 { > type jack > playback_ports { > 0 alsa_pcm:playback_1 > 1 alsa_pcm:playback_2 > } > capture_ports { > 0 alsa_pcm:capture_1 > 1 alsa_pcm:capture_2 > } > } > > pcm.jack1b { > type plug > device jack1b > server linux > > > slave { > pcm "jack1" > rate 8000 > format FLOAT > channels 2 > } > } > > > ctl.mixer0 { > type hw > card 1 > } > > in alsa.conf : > input_device=default > output_device=default > > It doesn't work. Asterisk outputs are : > [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open > failed: No such file or directory > [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open > failed: No such file or directory > [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:272 soundcard_init: Problem > opening ALSA I/O devices > > But aplay my_test_file.wav works fine and arecord too... > > Does anyone have an idea ? > Does asterisk run under the same user as the .asoundrc is defined for? Since your card is nb. 1, perhaps asterisk uses a default configuration for card0 which may not be present on your system. Pavel.

1 0

[alsa-devel] [PATCH] sscape: force AD1848 codec mode on old Soundscape
by Krzysztof Helt 05 Oct '09

05 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Old Soundscape cards (pre PnP) work only with AD1848 codecs. If the CS4231 codec is installed it must be used in AD1848 compatible mode. Also, add gameport support and remove an unused mpu field. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Even more OSS code moved into the alsa. I have tested it on early SPEA Media FX card. The only problem is that MIDI playback is not played timely. However, I observe this problem on cards with OPL4 chipset and Dream chipset. I assume it is more general problem with mpu401 and MIDI playback. If Theodore Allen test this on Ensoniq Soundscape card the OSS sscape driver can be removed. diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index cf98525..6de56d1 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1639,6 +1639,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. mpu_irq - MPU-401 IRQ # (PnP setup) dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index c739374..e348f8b 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -54,6 +54,7 @@ static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = false }; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); @@ -79,6 +80,9 @@ MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); + #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -145,7 +149,6 @@ struct soundscape { struct resource *io_res; struct resource *wss_res; struct snd_wss *chip; - struct snd_mpu401 *mpu; unsigned char midi_vol; }; @@ -815,7 +818,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; - sscape->mpu = mpu; initialise_mpu401(mpu); } @@ -836,12 +838,30 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, register struct soundscape *sscape = get_card_soundscape(card); struct snd_wss *chip; int err; + int codec_type = WSS_HW_DETECT; + + switch (sscape->type) { + case MEDIA_FX: + case SSCAPE: + /* + * There are some freak examples of early Soundscape cards + * with CS4231 instead of AD1848/CS4248. Unfortunately, the + * CS4231 works only in CS4248 compatibility mode on + * these cards so force it. + */ + if (sscape->ic_type != IC_OPUS) + codec_type = WSS_HW_AD1848; + break; - if (sscape->type == SSCAPE_VIVO) + case SSCAPE_VIVO: port += 4; + break; + default: + break; + } err = snd_wss_create(card, port, -1, irq, dma1, dma2, - WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); + codec_type, WSS_HWSHARE_DMA1, &chip); if (!err) { unsigned long flags; struct snd_pcm *pcm; @@ -927,6 +947,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; + int val; const char *name; /* @@ -1026,6 +1047,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); mpu_irq_cfg |= mpu_irq_cfg << 2; + val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7; + if (joystick[dev]) + val |= 8; + sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10); sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT -- 1.6.0.3 --------------------------------------------------------------- Przygotuj si� na przysz�o��! Kliknij >>> http://link.interia.pl/f238a

2 1

[alsa-devel] [PATCH] sound: Don't assume i2c device probing always succeeds
by Jean Delvare 05 Oct '09

05 Oct '09

If i2c device probing fails, then there is no driver to dereference after calling i2c_new_device(). Stop assuming that probing will always succeed, to avoid NULL pointer dereferences. We have an easier access to the driver anyway. Reported-by: Tim Shepard <shep(a)alum.mit.edu> Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> --- The code is similar to the one in therm_adt746x, for which Tim reported a real-world oops, so it should be fixed ASAP. sound/aoa/codecs/onyx.c | 4 +++- sound/aoa/codecs/tas.c | 4 +++- sound/ppc/keywest.c | 4 +++- 3 files changed, 9 insertions(+), 3 deletions(-) --- linux-2.6.32-rc1.orig/sound/aoa/codecs/onyx.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/aoa/codecs/onyx.c 2009-09-30 15:13:58.000000000 +0200 @@ -996,6 +996,8 @@ static void onyx_exit_codec(struct aoa_c onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); } +static struct i2c_driver onyx_driver; + static int onyx_create(struct i2c_adapter *adapter, struct device_node *node, int addr) @@ -1027,7 +1029,7 @@ static int onyx_create(struct i2c_adapte * Let i2c-core delete that device on driver removal. * This is safe because i2c-core holds the core_lock mutex for us. */ - list_add_tail(&client->detected, &client->driver->clients); + list_add_tail(&client->detected, &onyx_driver.clients); return 0; } --- linux-2.6.32-rc1.orig/sound/aoa/codecs/tas.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/aoa/codecs/tas.c 2009-09-30 15:13:58.000000000 +0200 @@ -882,6 +882,8 @@ static void tas_exit_codec(struct aoa_co } +static struct i2c_driver tas_driver; + static int tas_create(struct i2c_adapter *adapter, struct device_node *node, int addr) @@ -902,7 +904,7 @@ static int tas_create(struct i2c_adapter * Let i2c-core delete that device on driver removal. * This is safe because i2c-core holds the core_lock mutex for us. */ - list_add_tail(&client->detected, &client->driver->clients); + list_add_tail(&client->detected, &tas_driver.clients); return 0; } --- linux-2.6.32-rc1.orig/sound/ppc/keywest.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/ppc/keywest.c 2009-09-30 15:13:58.000000000 +0200 @@ -40,6 +40,8 @@ static int keywest_probe(struct i2c_clie return 0; } +struct i2c_driver keywest_driver; + /* * This is kind of a hack, best would be to turn powermac to fixed i2c * bus numbers and declare the sound device as part of platform @@ -65,7 +67,7 @@ static int keywest_attach_adapter(struct * This is safe because i2c-core holds the core_lock mutex for us. */ list_add_tail(&keywest_ctx->client->detected, - &keywest_ctx->client->driver->clients); + &keywest_driver.clients); return 0; } -- Jean Delvare

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[alsa-devel] hda-intel via codec on optical sound [PARTIAL SUCCESS]
by AL13N 05 Oct '09

05 Oct '09

forget about all of it, i'm a stupid doofus ------------------------------------------------------- First of all, I HAVE optical sound output. secondly, the hda analyzer still reports the connector as not_present thirdly, i probably have found a bug: * it happens when playing passthrough DD and DTS, as well as when using a52 * what happens is that the front right is playing on the LFE (muffled, but that's probably normal) * so when speakertesting -t wav -c 6 to a52 which points to hw0,1 , my optical surround set recognizes it as dolby sound, no problem. * the sound "front right" as well as the LFE comes from the subwoofer. * keep in mind this also happens in passthrough. What i've done, is i'm using alsa-driver-1.0.21 (after seeing some digital out changes between 19 and 21.) It is weird that it still says not_connected in codec#0, but the "-D iec958:CARD=SB" seems to work fine. can anyone help me find the bug? or tell me this is not a bug? Sincerely, AL13N ----------------------------------------------------------------- I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

1 0

[alsa-devel] hda-intel via codec on optical sound [PARTIAL SUCCESS]
by AL13N 05 Oct '09

05 Oct '09

First of all, I HAVE optical sound output. secondly, the hda analyzer still reports the connector as not_present thirdly, i probably have found a bug: * it happens when playing passthrough DD and DTS, as well as when using a52 * what happens is that the front right is playing on the LFE (muffled, but that's probably normal) * so when speakertesting -t wav -c 6 to a52 which points to hw0,1 , my optical surround set recognizes it as dolby sound, no problem. * the sound "front right" as well as the LFE comes from the subwoofer. * keep in mind this also happens in passthrough. What i've done, is i'm using alsa-driver-1.0.21 (after seeing some digital out changes between 19 and 21.) It is weird that it still says not_connected in codec#0, but the "-D iec958:CARD=SB" seems to work fine. can anyone help me find the bug? or tell me this is not a bug? Sincerely, AL13N ----------------------------------------------------------------- I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

1 0

[alsa-devel] hda-intel via codec on optical sound
by AL13N 04 Oct '09

04 Oct '09

I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

1 0

[alsa-devel] hda-intel via codec on optical sound
by AL13N 04 Oct '09

04 Oct '09

I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

1 0

[alsa-devel] [PATCH] sscape: remove MIDI instances counting with limit ULONG_MAX
by Krzysztof Helt 04 Oct '09

04 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the second version of this patch with improved issues pointed by Takashi. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465c..c739374 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -147,12 +147,6 @@ struct soundscape { struct snd_wss *chip; struct snd_mpu401 *mpu; - /* - * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL - */ - spinlock_t fwlock; - unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +158,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +168,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR "sscape: Failed to allocate " + "%lu bytes for DMA\n", + size); return NULL; } } @@ -482,7 +473,8 @@ static int upload_dma_data(struct soundscape *s, */ spin_unlock_irqrestore(&s->lock, flags); - snd_printk(KERN_ERR "sscape: DMA upload has timed out\n"); + snd_printk(KERN_ERR + "sscape: DMA upload has timed out\n"); ret = -EAGAIN; goto _release_dma; } @@ -504,10 +496,12 @@ static int upload_dma_data(struct soundscape *s, */ ret = 0; if (!obp_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); + snd_printk(KERN_ERR "sscape: No response " + "from on-board processor after upload\n"); ret = -EAGAIN; } else if (!host_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); + snd_printk(KERN_ERR + "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } @@ -536,7 +530,7 @@ static int sscape_upload_bootblock(struct snd_card *card) ret = request_firmware(&init_fw, "scope.cod", card->dev); if (ret < 0) { - snd_printk(KERN_ERR "Error loading scope.cod"); + snd_printk(KERN_ERR "sscape: Error loading scope.cod"); return ret; } ret = upload_dma_data(sscape, init_fw->data, init_fw->size); @@ -554,7 +548,8 @@ static int sscape_upload_bootblock(struct snd_card *card) data &= 0xf; if (ret == 0 && data > 7) { - snd_printk(KERN_ERR "timeout reading firmware version\n"); + snd_printk(KERN_ERR + "sscape: timeout reading firmware version\n"); ret = -EAGAIN; } @@ -575,12 +570,13 @@ static int sscape_upload_microcode(struct snd_card *card, int version) err = request_firmware(&init_fw, name, card->dev); if (err < 0) { - snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + snd_printk(KERN_ERR "sscape: Error loading sndscape.co%d", + version); return err; } err = upload_dma_data(sscape, init_fw->data, init_fw->size); if (err == 0) - snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n", init_fw->size >> 10); release_firmware(init_fw); @@ -750,7 +746,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); if ((inb(wss_io) & 0x80) != 0) goto _done; @@ -774,7 +769,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); /* * SoundScape successfully detected! @@ -794,38 +788,13 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ static int mpu401_open(struct snd_mpu401 * mpu) { - int err; - if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); - err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); + snd_printk(KERN_ERR "sscape: MIDI disabled, " + "please load firmware\n"); + return -ENODEV; } - return err; -} - -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); + return 0; } /* @@ -845,8 +814,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -993,13 +960,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1036,7 +1003,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,8 +1040,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", - wss_port[dev], irq[dev]); + snd_printk(KERN_ERR + "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + wss_port[dev], irq[dev]); goto _release_dma; } strcpy(card->driver, "SoundScape"); @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "sscape: Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "sscape: device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- Zobacz jak pracuje sie na wysokosciach. Kliknij >>> http://link.interia.pl/f2384

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Re: [alsa-devel] Asterisk and ALSA/Jack
by Fabien COMTE 03 Oct '09

03 Oct '09

Hi, I want to use Asterisk with ALSA / Jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd started my .asoundrc : pcm.!default { type hw card 1 } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } pcm.jack1b { type plug device jack1b server linux slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } ctl.mixer0 { type hw card 1 } in alsa.conf : input_device=default output_device=default That works fine. 3 ) chan_alsa.so with jackd started Jackd launch command : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & my .asoundrc : pcm.!default { type plug slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } pcm.jack1b { type plug device jack1b server linux slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } ctl.mixer0 { type hw card 1 } in alsa.conf : input_device=default output_device=default It doesn't work. Asterisk outputs are : [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open failed: No such file or directory [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open failed: No such file or directory [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:272 soundcard_init: Problem opening ALSA I/O devices But aplay my_test_file.wav works fine and arecord too... Does anyone have an idea ? Thanks, Fabien

1 0

Re: [alsa-devel] My Codec: IDT 92HD75B3X5 issues
by Munzir Taha (منذر طه) 02 Oct '09

02 Oct '09

On Yau al-Thulatha 09 Shawwal 1430 5:37:04 pm you wrote: > At Tue, 29 Sep 2009 16:24:29 +0300, > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > On Yau al-Thulatha 09 Shawwal 1430 9:24:19 am Takashi Iwai wrote: > > > At Tue, 29 Sep 2009 07:40:56 +0300, > > > > > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > > > > =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > > > On Yaum al-Ithnain 08 Shawwal 1430 11:41:52 am Takashi Iwai wrote: > > > > > At Mon, 28 Sep 2009 06:58:46 +0300, > > > > > > > > > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > > > > > > > > =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > > > > > Hi sirs, > > > > > > These are the issues I have with my Laptop HP Pavilion dv6 1225ee > > > > > > regarding audio support. > > > > > > > > > > > > 1. No sense for headphone jack (jack_present?), when headphone is > > > > > > plugged (present_int_hp?) > > > > > > > > > > > > 2. There headphone mic jack is not working. The one next to the > > > > > > webcam is always recording even when I change the "Mic Jack" > > > > > > control from "Mic In" to "Line In" > > > > > > > > > > Make sure that you use alsa-driver-snapshot tarball instead of > > > > > 1.0.21 released version. There are many fixes since 1.0.21 > > > > > release. > > > > > > > > > > > > > > > > > > > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-dr > > > > >iver -sn apshot.tar.gz > > > > > > > > Thanks a lot Takashi, things are better now. However, when I tested > > > > the internal mic it's very noisy. I played a bit with the capture and > > > > digital controls but still compared with Windows, it's lame. > > > > > > OK, could you give alsa-info.sh output at this state? > > > Please run with --no-upload option, and attach the generated file. > > > > Attached. > > Try to run: > > % amixer -c0 set Capture cap 12dB > % amixer -c0 set "Digital Mic" 10dB > % amixer -c0 set Digital 0dB > > Any difference? No, still there is a clear shshshshshsh sound in the background. Please check the attachment for a short arecord -f cd without me speaking on the mic and still you can hear this noise very clear. Also attached is the alsa-info report at the time of recording -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

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[alsa-devel] OFFTOPIC atkbd.c: Use 'setkeycodes... (Was: Re: My Codec: IDT 92HD75B3X5 issues)
by Simos Xenitellis 02 Oct '09

02 Oct '09

On Tue, Sep 29, 2009 at 7:40 AM, Munzir Taha (منذر طه) <munzirtaha(a)gmail.com> wrote: > On Yaum al-Ithnain 08 Shawwal 1430 11:41:52 am Takashi Iwai wrote: >> At Mon, 28 Sep 2009 06:58:46 +0300, >> >> Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > =?utf-8?q?_=D8=B7=D9=87?=) wrote: >> > Hi sirs, >> > These are the issues I have with my Laptop HP Pavilion dv6 1225ee >> > regarding audio support. >> > >> > 1. No sense for headphone jack (jack_present?), when headphone is plugged >> > (present_int_hp?) >> > >> > 2. There headphone mic jack is not working. The one next to the webcam is >> > always recording even when I change the "Mic Jack" control from "Mic In" >> > to "Line In" >> >> Make sure that you use alsa-driver-snapshot tarball instead of 1.0.21 >> released version. There are many fixes since 1.0.21 release. > > >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sn >> apshot.tar.gz > > Thanks a lot Takashi, things are better now. However, when I tested the > internal mic it's very noisy. I played a bit with the capture and digital > controls but still compared with Windows, it's lame. > >> > 3. The mute/unmute LED doesn't change color to RED when its muted like in >> > Windows. >> >> This isn't supported yet as we have no hardware information. >> Likely one of GPIO pins corresponds to the LED on/off. You can try >> hda-verb to change the GPIO data bit. > > Ok, I will try it and report back. > >> > Touching that LED shows this in /var/log/messages: >> > >> > Sep 28 06:53:47 dv6 kernel: [ 9197.986789] atkbd.c: Unknown key released >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9197.986791] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> > Sep 28 06:53:47 dv6 kernel: [ 9198.353568] atkbd.c: Unknown key pressed >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9198.353572] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> > Sep 28 06:53:47 dv6 kernel: [ 9198.359205] atkbd.c: Unknown key released >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9198.359208] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> >> It's not about the sound driver but the input layer, as you can find. >> You can defien the scan code 0xe014 by yourself. > > You are right. What's the best component to report this bug against? kernel? Those keys are a 'HAL' thing. Until recently there has been a website by the HAL developer with detailed instructions. Search for 'HAL Quirk Site' on Google to view the site in Google cache. The website was not updated for some time (http://www.mail-archive.com/pm-utils@lists.freedesktop.org/msg01594.html) and being obsolete, it was probably taken down. Your changes would need to make it eventually to the hal-info repository, http://cgit.freedesktop.org/hal-info/ Use the Freedesktop Bugzilla, product 'hal' and component 'hal-info' when reporting, https://bugs.freedesktop.org/enter_bug.cgi?product=hal Simos

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[alsa-devel] [PATCH] Soc: wm8940: Fix check on error code form snd_soc_codec_set_cache_io
by Jonathan Cameron 02 Oct '09

02 Oct '09

Soc: wm8940: Fix check on error code form snd_soc_codec_set_cache_io Fix for typo in commit 8d50e447d19fec64adebeef55f2b60d695435412 ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs Signed-off-by: Jonathan Cameron <jic23(a)cam.ac.uk> --- sound/soc/codecs/wm8940.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index da97aae..1ef2454 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940, codec->reg_cache = &wm8940->reg_cache; ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); - if (ret == 0) { + if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } -- 1.6.3.3

2 1

[alsa-devel] big latency and buffering question
by Achim Herrmann 02 Oct '09

02 Oct '09

Hi all, I'm completely new to sound and ALSA development and I have to develop an application which is capturing periodically a small amount of audio data (e.g. 64 frames), doing some processing and doing playback of the result. I started with ALSA example "pcm.c" to see if ALSA can meet these requirements. During investigation i found out, that up to a choosen buffer size of 140ms, generated sine sound is fine. Below 140ms, sound is stuttering, independent of call method I'm using. For buffer size of 140ms I found out, that target buffer size is 6174 frames (at 44.1kHz sample rate), period size is 3087 frames and loop inside write_and_poll function is calculated every 70 ms and 3087 frames are written. Think this is doublebuffering then, right? But if I now choose a buffersize of 139 ms or lower, target buffer size and period size are always the same. So no double buffering anymore and sound is stuttering. So, currently I'm far away from goal. I'd like to have a buffer size of e.g. 128 frames (2ms). Thus, period size should be 64 frames and loop should be calculated every 1ms. (If necessary I would use Linux-RT) If I choose a buffersize of 2ms, sound is far away from being sine :-) - Can I use ALSA for a application described in first sentence above? - What do I have to change in pcm.c to be able to create a sine signal using a buffersize of e.g. 128 frames and a period size of 64 frames? Any hints are appreciated Thanks in advance Achim

1 1

[alsa-devel] [PATCH] ASoC: Improve the debugfs hierarchy
by Peter Ujfalusi 02 Oct '09

02 Oct '09

Change the way the debugfs entries are created: If the codec->dev is valid, than use: debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/ if the codec->dev is NULL: debugfs/asoc/{codec->name}/ as root for the debugfs entries. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- The patch applies on top of sound-2.6 topic/asoc commit 88439ac793934a47f47ad285656b63d09f5937c8 ASoC: add support for multiple cards/codecs in debugfs sound/soc/soc-core.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e4ab36d..1dec9d2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1256,8 +1256,12 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { char codec_root[128]; - snprintf(codec_root, sizeof(codec_root), - "%s-%s", dev_name(codec->socdev->dev), codec->name); + if (codec->dev) + snprintf(codec_root, sizeof(codec_root), + "%s.%s", codec->name, dev_name(codec->dev)); + else + snprintf(codec_root, sizeof(codec_root), + "%s", codec->name); codec->debugfs_codec_root = debugfs_create_dir(codec_root, debugfs_root); -- 1.6.5.rc2

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 02 Oct '09

02 Oct '09

The following changes since commit c877c25170e2655d519b29e91d6c91d5d1a72a6f: Takashi Iwai (1): ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 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: fix kconfig order of Blackfin drivers Mark Brown (1): Merge remote branch 'takashi/fix/asoc' into for-2.6.32 sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++++++---------------------- 1 files changed, 49 insertions(+), 49 deletions(-)

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[alsa-devel] [PATCH 06/34] don't use __devexit_p to wrap hal2_remove
by Uwe Kleine-König 02 Oct '09

02 Oct '09

The function hal2_remove is defined using __exit, so don't use __devexit_p but __exit_p to wrap it. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/mips/hal2.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index c52691c..72523a7 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -926,7 +926,7 @@ static int __exit hal2_remove(struct platform_device *pdev) static struct platform_driver hal2_driver = { .probe = hal2_probe, - .remove = __devexit_p(hal2_remove), + .remove = __exit_p(hal2_remove), .driver = { .name = "sgihal2", .owner = THIS_MODULE, -- 1.6.4.3

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[alsa-devel] [PATCH] Don't check if period size is equal to buffer size
by Lubomir Rintel 02 Oct '09

02 Oct '09

At least some drivers seem to allow a single period and those which don't would correct us anyway. Moreover, Pulseaudio with tsched sets this frequently. --- aplay/aplay.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index c7c82a1..4fa36cb 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1044,11 +1044,6 @@ static void set_params(void) } snd_pcm_hw_params_get_period_size(params, &chunk_size, 0); snd_pcm_hw_params_get_buffer_size(params, &buffer_size); - if (chunk_size == buffer_size) { - error(_("Can't use period equal to buffer size (%lu == %lu)"), - chunk_size, buffer_size); - exit(EXIT_FAILURE); - } snd_pcm_sw_params_current(handle, swparams); if (avail_min < 0) n = chunk_size; -- 1.6.2.5

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[alsa-devel] [PATCH] sound: Make keywest_driver static
by Jean Delvare 02 Oct '09

02 Oct '09

From: Jean Delvare <khali(a)linux-fr.org> Subject: sound: Make keywest_driver static I can't see any reason for struct i2c_driver keywest_driver to not be static. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/ppc/keywest.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-2.6.32-rc1.orig/sound/ppc/keywest.c 2009-10-01 09:45:18.000000000 +0200 +++ linux-2.6.32-rc1/sound/ppc/keywest.c 2009-10-01 18:05:06.000000000 +0200 @@ -93,7 +93,7 @@ static const struct i2c_device_id keywes { } }; -struct i2c_driver keywest_driver = { +static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, -- Jean Delvare

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[alsa-devel] [PATCH] sscape: remove MIDI instances counting with limit ULONG_MAX
by Krzysztof Helt 02 Oct '09

02 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, remove redundant "sscape:" prefix in messages printed by snd_printk and convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sscape.c | 85 +++++++++++++--------------------------------------- 1 files changed, 21 insertions(+), 64 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465c..ece2bb9 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -149,9 +149,8 @@ struct soundscape { /* * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL + * its firmware. */ - spinlock_t fwlock; unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +163,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +173,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR + "Failed to allocate %lu bytes for DMA\n", + size); return NULL; } } @@ -797,37 +793,13 @@ static int mpu401_open(struct snd_mpu401 * mpu) int err; if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); + snd_printk(KERN_ERR "MIDI disabled, please load firmware\n"); err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); } return err; } -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); -} - /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ @@ -845,8 +817,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -908,21 +878,20 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, err = snd_wss_pcm(chip, 0, &pcm); if (err < 0) { - snd_printk(KERN_ERR "sscape: No PCM device " - "for AD1845 chip\n"); + snd_printk(KERN_ERR "No PCM device for AD1845 chip\n"); goto _error; } err = snd_wss_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR "sscape: No mixer device " + snd_printk(KERN_ERR "No mixer device " "for AD1845 chip\n"); goto _error; } if (chip->hardware != WSS_HW_AD1848) { err = snd_wss_timer(chip, 0, NULL); if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " + snd_printk(KERN_ERR "No timer device " "for AD1845 chip\n"); goto _error; } @@ -932,7 +901,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, err = snd_ctl_add(card, snd_ctl_new1(&midi_mixer_ctl, chip)); if (err < 0) { - snd_printk(KERN_ERR "sscape: Could not create " + snd_printk(KERN_ERR "Could not create " "MIDI mixer control\n"); goto _error; } @@ -968,15 +937,14 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR - "sscape: can't grab port 0x%lx\n", port[dev]); + snd_printk(KERN_ERR "can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; if (sscape->type == SSCAPE_VIVO) { wss_res = request_region(wss_port[dev], 4, "SoundScape"); if (!wss_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%lx\n", + snd_printk(KERN_ERR "can't grab port 0x%lx\n", wss_port[dev]); err = -EBUSY; goto _release_region; @@ -988,18 +956,18 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ err = request_dma(dma[dev], "SoundScape"); if (err < 0) { - snd_printk(KERN_ERR "sscape: can't grab DMA %d\n", dma[dev]); + snd_printk(KERN_ERR "can't grab DMA %d\n", dma[dev]); goto _release_region; } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1030,13 +998,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + snd_printk(KERN_ERR "Invalid IRQ %d\n", irq[dev]); return -ENXIO; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,7 +1041,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + snd_printk(KERN_ERR "No AD1845 device at 0x%lx, IRQ %d\n", wss_port[dev], irq[dev]); goto _release_dma; } @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- 2012 rok wcale nie musi by� tragiczny! Kliknij >>> http://link.interia.pl/f238b

2 1

[alsa-devel] Patches for alsactl init db
by Dan Chen 02 Oct '09

02 Oct '09

Hi, Attached are two patches against alsa-utils.git HEAD that create an intel8x0 init db file for ICH4. This is necessary from bug reports in Debian and Ubuntu that state that External Amplifier must be muted to have audible (analog) playback by default on the Sony VAIO VGN-T350P. Thanks, Dan

3 5

[alsa-devel] Help! Playing multiple audio stream using alsa libraries
by Arjun Sambamoorthy 02 Oct '09

02 Oct '09

Hello all, I am trying to play two audio streams using alsa libraries at the same time, but I am not able to do so. I could play a single audio stream though. I get bi-directional G711 MU Law RTP streams and I am using the sample playback capture code from the tutorial section for playing it. I am using the Non blocking and interleaved mode for playing the streams. The alsa library plays both the audio streams for a few seconds but it is incomprehensible and after few seconds I get the error "write to audio interface failed (broken pipe)". I also tried opening two pcm handles for each stream to the same device ("default"), but the second handle is unusable gives the same error "write to audio interface failed (broken pipe)". I have the same problem playing two audio streams to the /dev/dsp device. I am pretty sure I am not doing it the right way. Any help is highly appreciated or pointer to links that explains how to play multiple streams using alsa. Thanks Arjunen

1 1

[alsa-devel] [PATCH 1/3] ASoC: DaVinci: i2s, reduce underruns by combining into 1 element
by Troy Kisky 01 Oct '09

01 Oct '09

Allow the left and right 16 bit samples to be shifted out as 1 32 bit sample. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> -- This applies to Kevin's temp/asoc branch --- arch/arm/mach-davinci/include/mach/asp.h | 6 ++ sound/soc/davinci/davinci-i2s.c | 74 ++++++++++++++++++++++-------- 2 files changed, 61 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..a3d2aa1 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -51,6 +51,12 @@ struct snd_platform_data { u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ unsigned int codec_fmt; + /* + * Allowing this is more efficient and eliminates left and right swaps + * caused by underruns, but will swap the left and right channels + * when compared to previous behavior. + */ + unsigned disable_channel_combine:1; /* McASP specific fields */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c54..081b2d4 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,6 +97,23 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { .name = "I2S PCM Stereo out", }; @@ -113,6 +130,27 @@ struct davinci_mcbsp_dev { u32 pcr; struct clk *clk; struct davinci_pcm_dma_params *dma_params[2]; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned disable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -359,6 +397,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -388,27 +428,22 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - - dma_params->acnt = dma_params->data_type; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && !dev->disable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -524,7 +559,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) + dev->disable_channel_combine = pdata->disable_channel_combine; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; -- 1.5.6.3

6 28

[alsa-devel] A few fix patches for ASoC
by Takashi Iwai 01 Oct '09

01 Oct '09

Mark, I applied a few fix patches to fix/asoc and topic/asoc branches for build errors I found with cross-build tests in this morning. Check below and sync to your tree. commit bb26276744a80d066681836f4d49c70010b129d6 Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 07:39:45 2009 +0200 ASoC: Fix build errors of wm8711.c with SPI commit c877c25170e2655d519b29e91d6c91d5d1a72a6f Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 08:33:47 2009 +0200 ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 commit 140318aaa924ce9664ff59366993228cf1547f1d Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 08:40:32 2009 +0200 ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c

2 3

[alsa-devel] LEFT or RIGHT Channel in alsa mixer
by Navink Singh 01 Oct '09

01 Oct '09

Hi Developers, I need to configure my audio device such as to use only LEFT or RIGHT Channel. I have carried out following steps to proceed with 1) Opened PCM playback Device 2) Configured its hardware and Software parameters 3) Open mixer 4) Attached Mixer to playback device 5) Registered Mixer 6) Loaded the mixer 7) Get the first element of mixer A) Now when i use following API, snd_mixer_selem_is_playback_mono(aElem); i found mixer is using MONO channel. But i am configuring my PCM Device for 2 channels I am using snd_mixer_selem_set_playback_switch() API to set right or Left channel to MUTE {Does not work} If i set all channels to Mute {Works fine} snd_mixer_selem_set_playback_switch_all (aElem, 0) Also i am able to set SND_MIXER_SCHN_MONO to mute {works fine} snd_mixer_selem_set_playback_switch(aElem, SND_MIXER_SCHN_MONO, 0) B) There is also an API in my MIXER driver to set the right and left balance, but i did not find any API in Application to do so {Is there any such API in Application}. Thanks in Advance, Navin Kumar Singh Tata Consultancy Services DLF Akruti Info Park, PN 28, Rajiv Gandhi Infotech Park Hinjewadi, PHASE II Pune - 411057,Maharashtra India Ph:- +91 20 66752458 Cell:- +919860227357 Mailto: navink.singh(a)tcs.com Website: http://www.tcs.com ____________________________________________ Experience certainty. IT Services Business Solutions Outsourcing ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

1 0

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 01 Oct '09

01 Oct '09

The following changes since commit 140318aaa924ce9664ff59366993228cf1547f1d: Takashi Iwai (1): ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Barry Song (1): ASoC: fix kconfig order of Blackfin drivers Chaithrika U S (1): ASoC: DaVinci: McASP FIFO related updates Lopez Cruz, Misael (1): ASoC: Add PDM DAI format definition Mark Brown (6): ASoC: Clean up WM8974 PLL configuration Merge branch 'upstream/wm8974' into for-2.6.33 ASoC: Factor out analogue platform data from WM8993 Merge remote branch 'takashi/fix/asoc' into for-2.6.32 Merge remote branch 'takashi/topic/asoc' into for-2.6.33 Merge branch 'for-2.6.32' into for-2.6.33 Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs Takashi Iwai (1): ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 include/sound/soc-dai.h | 1 + include/sound/soc.h | 1 + sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++------------------ sound/soc/codecs/wm8974.c | 23 +++++---- sound/soc/codecs/wm8993.c | 36 +++---------- sound/soc/codecs/wm_hubs.c | 35 +++++++++++++ sound/soc/codecs/wm_hubs.h | 5 ++ sound/soc/davinci/davinci-i2s.c | 2 + sound/soc/davinci/davinci-mcasp.c | 17 +++---- sound/soc/davinci/davinci-pcm.c | 21 +++++++- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/pxa/Kconfig | 2 +- sound/soc/soc-core.c | 26 +++++++--- 13 files changed, 161 insertions(+), 107 deletions(-)

2 1

[alsa-devel] Tascam
by humbert.olivier.1＠free.fr 01 Oct '09

01 Oct '09

Hi all in first thanks for your work, it permit us to make music with GNU Linux :) I'm Olivier from France. I work on linuxMAO.org website. It's a french website for music assisted by computer. In this website, i make an article for make work the tascam us-224 on GNU Linux. Sometime I was angry because all is'nt recognize on my sound card. (Like 4 faders for 4 tracks). So I make a letter in what i said something like that : "Hey Tascam, it's cool for you ? For me it's cool, because i buy your product and it's a good one, and now, i play music with my friend and children and many people. But, all doesn't work with my OS, GNU/Linux it's damage because you don't see that many people on linux and other OS, have to go to your concurent (like M-audio or RME) because they're more open than you. blablabla blablabla ..." They don't answer. ... :/ Another boy make another one few month ago. ... They don't answer too. ... :/ ... Last week, a new boy email me that, he was very happy because my website page for the TASCAM us 224 help him to make work on linux. (it was great, i'm happy) So i speak with him and said to him "hey, you can make an email to Tascam (in fact sennheiser in france, the official distributor for tascam products) and say to them "hello boys, can you give us the source code or specifications blablabla" " ... and this one, he answered positively ! The sennheiser man said to us that he speak with japanese guys who said "what they needs?". So, i'm not developper I search for the (one or many) people who made the tascam us 224 (and 122 and 448 i think) driver for ALSA system. because, we've got an open way to make it better. Or another one motived ! The ball is here now. Thanks for read and for your work. Olivier

1 0

[alsa-devel] [PATCH 0/1 - try2] ASoC: debugfs support improvement
by Peter Ujfalusi 01 Oct '09

01 Oct '09

Hello, I have noticed that the support for multiple cards/codecs via the debugfs is kind of broken. The second codec, which tries to create the debugfs files would emit the following messages: ASoC: Failed to create codec register debugfs file Failed to create pop time debugfs file ... The following patch fixes this by adding a directory per codec to host the files. The additional directory is named as: {dev_name(socdev->dev)}-{codec->name} for example: debugfs/asoc/soc-audio.0-twl4030/... So it is easier to find the correct codec we are looking for. Note: The patch accidentally fixes one trailing space in the soc_pcm_apply_symmetry function. --- Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs include/sound/soc.h | 1 + sound/soc/soc-core.c | 28 ++++++++++++++++++++-------- 2 files changed, 21 insertions(+), 8 deletions(-)

2 8

[alsa-devel] [PATCH] ASoC: Use codec mutex in dapm_set_pga()
by ext-eero.nurkkala＠nokia.com 01 Oct '09

01 Oct '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> codec->mutex seems required if widget->* values are altered. Moreover, snd_soc_dapm_put_volsw() may alter widget->saved_value. dapm_set_pga() uses widget->saved_value in a for loop which now has a distant chance of getting out of control. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> --- sound/soc/soc-dapm.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f79711b..d6fb6c4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -349,6 +349,7 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + mutex_lock(&widget->codec->mutex); if (power) { int i; /* power up has happended, increase volume to last level */ @@ -373,6 +374,7 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) } widget->muted = 1; } + mutex_unlock(&widget->codec->mutex); } return 0; } -- 1.6.0.4

3 7

[alsa-devel] adding hrtimer based delay to ioplug ALSA plugin on ARM9 (AT91) platform ?
by Stefan Schoenleitner 01 Oct '09

01 Oct '09

Hi, I'm currently developing an ALSA ioplug plugin which should transfer audio samples over UART to a DSP (I already wrote that in a previous post). The DSP wants to receive 156 - 164 audio samples with a sampling rate of 8 kHz and a format of S16_LE each 20 msec (which is more or less the number of audio samples multiplied by the period length of the sampling rate). My idea is to add a high precision delay to the transfer callback of the ALSA plugin. Unfortunately, I found out that the hrtimer resolution on the system is as low as 10ms: ------------------------------------------------------------------- kernel: hrtimer resolution: 0 sec, 10000000 nsec, loops_per_jiffy: 448512 ------------------------------------------------------------------- * Does this mean that if I use nanosleep for example each delay has to be an integral multiple of the hrtimer precision and for example a delay of 15ms would not be supported (since it is no integral multiple of 10ms) ? * It is possible to increase the precision (e.g. by changing the kernel configuration) ? * Do you know a better solution for the task, where the task is to send and receive audio samples to/from the DSP over UART and the timing is critical ? cheers, Stefan

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[alsa-devel] ioplug plugin programming: HW_PERIOD_BYTES and HW_PERIODS constraints ?
by Stefan Schoenleitner 01 Oct '09

01 Oct '09

Hi, I'm currently developing an ALSA ioplug plugin that should transfer audio samples to a DSP over UART. In my plugin code I set up the hardware constrains (HW_PERIOD_BYTES and HW_PERIODS being among them) and then I utilize the transfer callback to send the audio chunk over UART. The DSP application would like to have exactly 156 - 164 audio samples at a time, where the sampling rate has to be 8 kHz and the format is S16_LE. In between there has to be a delay of 20ms (which is equal to 164 audio frames at a sampling rate of 8 kHz (i.e. 164 * 125us = 20,5 ms). For this reason I configured HW_PERIOD_BYTES with these values. The result is that each time the transfer callback gets called, exactly 312 bytes (thus 164 frames with each one having 2 bytes) are transfered. However, what I do not understand is the setup of HW_PERIODS which is AFAIK the number of periods being transfered at a time ? * What is the influence of this constraint on my ioplug plugin ? * How does it change the way data is being transfered with the transfer callback ? * What would be the correct setup for the mentioned application ? cheers, stefan

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[alsa-devel] [PATCH] ASoC: fix kconfig order of Blackfin drivers
by Mike Frysinger 01 Oct '09

01 Oct '09

From: Barry Song <barry.song(a)analog.com> Some of the Blackfin options don't directly follow the kconfig options they depend on, so kconfig is unable to display the proper tree. So sort the options such they expand/collapse properly. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++++++---------------------- 1 files changed, 49 insertions(+), 49 deletions(-) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index ac927ff..f91dcb1 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -7,15 +7,6 @@ config SND_BF5XX_I2S mode (supports single stereo In/Out). You will also need to select the audio interfaces to support below. -config SND_BF5XX_TDM - tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" - depends on (BLACKFIN && SND_SOC) - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in TDM - mode. - You will also need to select the audio interfaces to support below. - config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" depends on SND_BF5XX_I2S @@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE Enter the GPIO used to control AD73311's SE pin. Acceptable values are 0 to 7 +config SND_BF5XX_TDM + tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" + depends on (BLACKFIN && SND_SOC) + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in TDM + mode. + You will also need to select the audio interfaces to support below. + +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_AD1938 + tristate "SoC AD1938 Audio support for Blackfin" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1938 + help + Say Y if you want to add support for AD1938 codec on Blackfin. + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN @@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT Say y if you want AC97 driver to support up to 5.1 channel audio. this mode will consume much more memory for DMA. +config SND_BF5XX_HAVE_COLD_RESET + bool "BOARD has COLD Reset GPIO" + depends on SND_BF5XX_AC97 + default y if BFIN548_EZKIT + default n if !BFIN548_EZKIT + +config SND_BF5XX_RESET_GPIO_NUM + int "Set a GPIO for cold reset" + depends on SND_BF5XX_HAVE_COLD_RESET + range 0 159 + default 19 if BFIN548_EZKIT + default 5 if BFIN537_STAMP + default 0 + help + Set the correct GPIO for RESET the sound chip. + +config SND_BF5XX_SOC_AD1980 + tristate "SoC AD1980/1 Audio support for BF5xx" + depends on SND_BF5XX_AC97 + select SND_BF5XX_SOC_AC97 + select SND_SOC_AD1980 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_SPORT tristate @@ -88,30 +128,6 @@ 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 - select SND_BF5XX_SOC_AC97 - select SND_SOC_AD1980 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD1938 - tristate "SoC AD1938 Audio support for Blackfin" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1938 - help - Say Y if you want to add support for AD1938 codec on Blackfin. - config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) @@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM default 0 help Set the correct SPORT for sound chip. - -config SND_BF5XX_HAVE_COLD_RESET - bool "BOARD has COLD Reset GPIO" - depends on SND_BF5XX_AC97 - default y if BFIN548_EZKIT - default n if !BFIN548_EZKIT - -config SND_BF5XX_RESET_GPIO_NUM - int "Set a GPIO for cold reset" - depends on SND_BF5XX_HAVE_COLD_RESET - range 0 159 - default 19 if BFIN548_EZKIT - default 5 if BFIN537_STAMP - default 0 - help - Set the correct GPIO for RESET the sound chip. -- 1.6.5.rc2

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[alsa-devel] [PATCH 24/34] don't use __devexit_p to wrap snd_sgio2audio_remove
by Uwe Kleine-König 01 Oct '09

01 Oct '09

The function snd_sgio2audio_remove is defined using __exit, so don't use __devexit_p but __exit_p to wrap it. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Figo.zhang <figo1802(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/mips/sgio2audio.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index e497525..e24ba5a 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -984,7 +984,7 @@ static int __exit snd_sgio2audio_remove(struct platform_device *pdev) static struct platform_driver sgio2audio_driver = { .probe = snd_sgio2audio_probe, - .remove = __devexit_p(snd_sgio2audio_remove), + .remove = __exit_p(snd_sgio2audio_remove), .driver = { .name = "sgio2audio", .owner = THIS_MODULE, -- 1.6.4.3

1 0

[alsa-devel] [PATCH 0/1] ASoC: debugfs support improvement
by Peter Ujfalusi 01 Oct '09

01 Oct '09

Hello, I have noticed that the support for multiple cards/codecs via the debugfs is kind of broken. The second codec, which tries to create the debugfs files would emit the following messages: ASoC: Failed to create codec register debugfs file Failed to create pop time debugfs file ... The following patch fixes this by adding a directory per codec to host the files. The additional directory is named as codec->name, so it is easier to find the correct codec we are looking for. --- Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs include/sound/soc.h | 1 + sound/soc/soc-core.c | 21 ++++++++++++++------- 2 files changed, 15 insertions(+), 7 deletions(-)

2 6

[alsa-devel] [PATCH] sscape: convert to firmware loader framework
by Krzysztof Helt 01 Oct '09

01 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional firmware initialization code has been moved from the OSS driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the second version with fixes to issues pointed by Takashi and two mistakes in the original patch. Documentation/sound/alsa/ALSA-Configuration.txt | 8 +- include/sound/sscape_ioctl.h | 21 -- sound/isa/Kconfig | 8 +- sound/isa/sscape.c | 328 ++++++++--------------- 4 files changed, 116 insertions(+), 249 deletions(-) delete mode 100644 include/sound/sscape_ioctl.h diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1c8eb45..cf98525 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1631,7 +1631,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sscape ----------------- - Module for ENSONIQ SoundScape PnP cards. + Module for ENSONIQ SoundScape cards. port - Port # (PnP setup) wss_port - WSS Port # (PnP setup) @@ -1640,9 +1640,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) - This module supports multiple cards. ISA PnP must be enabled. - You need sscape_ctl tool in alsa-tools package for loading - the microcode. + This module supports multiple cards. + + The driver requires the firmware loader support on kernel. Module snd-sun-amd7930 (on sparc only) -------------------------------------- diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h deleted file mode 100644 index 0d88859..0000000 --- a/include/sound/sscape_ioctl.h +++ /dev/null @@ -1,21 +0,0 @@ -#ifndef SSCAPE_IOCTL_H -#define SSCAPE_IOCTL_H - - -struct sscape_bootblock -{ - unsigned char code[256]; - unsigned version; -}; - -#define SSCAPE_MICROCODE_SIZE 65536 - -struct sscape_microcode -{ - unsigned char __user *code; -}; - -#define SND_SSCAPE_LOAD_BOOTB _IOWR('P', 100, struct sscape_bootblock) -#define SND_SSCAPE_LOAD_MCODE _IOW ('P', 101, struct sscape_microcode) - -#endif diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b90fc16..02fe81c 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -372,9 +372,9 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape driver" - select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB + select FW_LOADER help Say Y here to include support for Ensoniq SoundScape and Ensoniq OEM soundcards. @@ -382,7 +382,11 @@ config SND_SSCAPE The PCM audio is supported on SoundScape Classic, Elite, PnP and VIVO cards. The supported OEM cards are SPEA Media FX and Reveal SC-600. - The MIDI support is very experimental. + The MIDI support is very experimental and requires binary + firmware files called "scope.cod" and "sndscape.co?" where the + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found + in DOS or Windows driver packages. One has to put the firmware + files into the /lib/firmware directory. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b11c35f..1ce465c 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1,5 +1,5 @@ /* - * Low-level ALSA driver for the ENSONIQ SoundScape PnP + * Low-level ALSA driver for the ENSONIQ SoundScape * Copyright (c) by Chris Rankin * * This driver was written in part using information obtained from @@ -25,22 +25,26 @@ #include <linux/err.h> #include <linux/isa.h> #include <linux/delay.h> +#include <linux/firmware.h> #include <linux/pnp.h> #include <linux/spinlock.h> #include <linux/moduleparam.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/hwdep.h> #include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> -#include <sound/sscape_ioctl.h> - MODULE_AUTHOR("Chris Rankin"); -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); MODULE_LICENSE("GPL"); +MODULE_FIRMWARE("sndscape.co0"); +MODULE_FIRMWARE("sndscape.co1"); +MODULE_FIRMWARE("sndscape.co2"); +MODULE_FIRMWARE("sndscape.co3"); +MODULE_FIRMWARE("sndscape.co4"); +MODULE_FIRMWARE("scope.cod"); static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR; @@ -142,14 +146,12 @@ struct soundscape { struct resource *wss_res; struct snd_wss *chip; struct snd_mpu401 *mpu; - struct snd_hwdep *hw; /* * The MIDI device won't work until we've loaded * its firmware via a hardware-dependent device IOCTL */ spinlock_t fwlock; - int hw_in_use; unsigned long midi_usage; unsigned char midi_vol; }; @@ -167,12 +169,6 @@ static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) return (struct soundscape *) (mpu->private_data); } -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw) -{ - return (struct soundscape *) (hw->private_data); -} - - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -393,12 +389,12 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); - if ((x & 0xfe) == 0xfe) + if (x == 0xfe || x == 0xff) return 1; msleep(10); @@ -420,10 +416,10 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; @@ -438,14 +434,14 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) * Upload a byte-stream into the SoundScape using DMA channel A. */ static int upload_dma_data(struct soundscape *s, - const unsigned char __user *data, + const unsigned char *data, size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; spin_lock_irqsave(&s->lock, flags); @@ -458,7 +454,6 @@ static int upload_dma_data(struct soundscape *s, /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMACFG_REG, 0x50); sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); @@ -468,35 +463,17 @@ static int upload_dma_data(struct soundscape *s, sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); /* - * Upload the user's data (firmware?) to the SoundScape + * Upload the firmware to the SoundScape * board through the DMA channel ... */ while (size != 0) { unsigned long len; - /* - * Apparently, copying to/from userspace can sleep. - * We are therefore forbidden from holding any - * spinlocks while we copy ... - */ - spin_unlock_irqrestore(&s->lock, flags); - - /* - * Remember that the data that we want to DMA - * comes from USERSPACE. We have already verified - * the userspace pointer ... - */ len = min(size, dma.bytes); - len -= __copy_from_user(dma.area, data, len); + memcpy(dma.area, data, len); data += len; size -= len; - /* - * Grab that spinlock again, now that we've - * finished copying! - */ - spin_lock_irqsave(&s->lock, flags); - snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { @@ -512,6 +489,7 @@ static int upload_dma_data(struct soundscape *s, } /* while */ set_host_mode_unsafe(s->io_base); + outb(0x0, s->io_base); /* * Boot the board ... (I think) @@ -537,7 +515,7 @@ _release_dma: /* * NOTE!!! We are NOT holding any spinlocks at this point !!! */ - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); free_dmabuf(&dma); return ret; @@ -547,162 +525,69 @@ _release_dma: * Upload the bootblock(?) into the SoundScape. The only * purpose of this block of code seems to be to tell * us which version of the microcode we should be using. - * - * NOTE: The boot-block data resides in USER-SPACE!!! - * However, we have already verified its memory - * addresses by the time we get here. */ -static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb) +static int sscape_upload_bootblock(struct snd_card *card) { + struct soundscape *sscape = get_card_soundscape(card); unsigned long flags; + const struct firmware *init_fw = NULL; int data = 0; int ret; - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); - - spin_lock_irqsave(&sscape->lock, flags); - if (ret == 0) { - data = host_read_ctrl_unsafe(sscape->io_base, 100); - } - set_midi_mode_unsafe(sscape->io_base); - spin_unlock_irqrestore(&sscape->lock, flags); - - if (ret == 0) { - if (data < 0) { - snd_printk(KERN_ERR "sscape: timeout reading firmware version\n"); - ret = -EAGAIN; - } - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { - ret = -EFAULT; - } + ret = request_firmware(&init_fw, "scope.cod", card->dev); + if (ret < 0) { + snd_printk(KERN_ERR "Error loading scope.cod"); + return ret; } + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); - return ret; -} + release_firmware(init_fw); -/* - * Upload the microcode into the SoundScape. The - * microcode is 64K of data, and if we try to copy - * it into a local variable then we will SMASH THE - * KERNEL'S STACK! We therefore leave it in USER - * SPACE, and save ourselves from copying it at all. - */ -static int sscape_upload_microcode(struct soundscape *sscape, - const struct sscape_microcode __user *mc) -{ - unsigned long flags; - char __user *code; - int err; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now. - * - * NOTE: This buffer is 64K long! That's WAY too big to - * copy into a stack-temporary anyway. - */ - if ( get_user(code, &mc->code) || - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) - return -EFAULT; + spin_lock_irqsave(&sscape->lock, flags); + if (ret == 0) + data = host_read_ctrl_unsafe(sscape->io_base, 100); - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) { - snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n"); - } + if (data & 0x10) + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); - spin_lock_irqsave(&sscape->lock, flags); - set_midi_mode_unsafe(sscape->io_base); spin_unlock_irqrestore(&sscape->lock, flags); - initialise_mpu401(sscape->mpu); + data &= 0xf; + if (ret == 0 && data > 7) { + snd_printk(KERN_ERR "timeout reading firmware version\n"); + ret = -EAGAIN; + } - return err; + return (ret == 0) ? data : ret; } /* - * Hardware-specific device functions, to implement special - * IOCTLs for the SoundScape card. This is how we upload - * the microcode into the card, for example, and so we - * must ensure that no two processes can open this device - * simultaneously, and that we can't open it at all if - * someone is using the MIDI device. + * Upload the microcode into the SoundScape. */ -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) +static int sscape_upload_microcode(struct snd_card *card, int version) { - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; + struct soundscape *sscape = get_card_soundscape(card); + const struct firmware *init_fw = NULL; + char name[14]; int err; - spin_lock_irqsave(&sscape->fwlock, flags); + snprintf(name, sizeof(name), "sndscape.co%d", version); - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { - err = -EBUSY; - } else { - sscape->hw_in_use = 1; - err = 0; + err = request_firmware(&init_fw, name, card->dev); + if (err < 0) { + snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + return err; } + err = upload_dma_data(sscape, init_fw->data, init_fw->size); + if (err == 0) + snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + init_fw->size >> 10); - spin_unlock_irqrestore(&sscape->fwlock, flags); - return err; -} - -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) -{ - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - sscape->hw_in_use = 0; - spin_unlock_irqrestore(&sscape->fwlock, flags); - return 0; -} - -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct soundscape *sscape = get_hwdep_soundscape(hw); - int err = -EBUSY; - - switch (cmd) { - case SND_SSCAPE_LOAD_BOOTB: - { - register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now ... - */ - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) - return -EFAULT; - - /* - * Now check that we can write the firmware version number too... - */ - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) - return -EFAULT; - - err = sscape_upload_bootblock(sscape, bb); - } - break; - - case SND_SSCAPE_LOAD_MCODE: - { - register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg; - - err = sscape_upload_microcode(sscape, mc); - } - break; - - default: - err = -EINVAL; - break; - } /* switch */ + release_firmware(init_fw); return err; } - /* * Mixer control for the SoundScape's MIDI device. */ @@ -920,7 +805,7 @@ static int mpu401_open(struct snd_mpu401 * mpu) spin_lock_irqsave(&sscape->fwlock, flags); - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { + if (sscape->midi_usage == ULONG_MAX) { err = -EBUSY; } else { ++(sscape->midi_usage); @@ -1053,13 +938,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, } } - strcpy(card->driver, "SoundScape"); - strcpy(card->shortname, pcm->name); - snprintf(card->longname, sizeof(card->longname), - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", - pcm->name, chip->port, chip->irq, - chip->dma1, chip->dma2); - sscape->chip = chip; } @@ -1162,29 +1040,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) return -ENXIO; } - if (sscape->type != SSCAPE_VIVO) { - /* - * Now create the hardware-specific device so that we can - * load the microcode into the on-board processor. - * We cannot use the MPU-401 MIDI system until this firmware - * has been loaded into the card. - */ - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "firmware device\n"); - goto _release_dma; - } - strlcpy(sscape->hw->name, "SoundScape M68K", - sizeof(sscape->hw->name)); - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; - sscape->hw->ops.open = sscape_hw_open; - sscape->hw->ops.release = sscape_hw_release; - sscape->hw->ops.ioctl = sscape_hw_ioctl; - sscape->hw->private_data = sscape; - } - /* * Tell the on-board devices where their resources are (I think - * I can't be sure without a datasheet ... So many magic values!) @@ -1222,28 +1077,56 @@ static int __devinit create_sscape(int dev, struct snd_card *card) wss_port[dev], irq[dev]); goto _release_dma; } + strcpy(card->driver, "SoundScape"); + strcpy(card->shortname, name); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", + name, sscape->chip->port, sscape->chip->irq, + sscape->chip->dma1, sscape->chip->dma2); + #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%lx\n", - port[dev]); - goto _release_dma; - } + err = sscape_upload_bootblock(card); + if (err >= 0) + err = sscape_upload_microcode(card, err); - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* - * Initialize mixer - */ - sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + if (err == 0) { + err = create_mpu401(card, MIDI_DEVNUM, port[dev], + mpu_irq[dev]); + if (err < 0) { + printk(KERN_ERR "sscape: Failed to create " + "MPU-401 device at 0x%lx\n", + port[dev]); + goto _release_dma; + } + + /* + * Enable the master IRQ ... + */ + sscape_write(sscape, GA_INTENA_REG, 0x80); + + /* + * Initialize mixer + */ + spin_lock_irqsave(&sscape->lock, flags); + sscape->midi_vol = 0; + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_XXX_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_EXTMIDI, 100); + host_write_ctrl_unsafe(sscape->io_base, + 0, 100); + host_write_ctrl_unsafe(sscape->io_base, CMD_ACK, 100); + + set_midi_mode_unsafe(sscape->io_base); + spin_unlock_irqrestore(&sscape->lock, flags); + } } /* @@ -1301,11 +1184,12 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) sscape->type = SSCAPE; dma[dev] &= 0x03; + snd_card_set_dev(card, pdev); + ret = create_sscape(dev, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, pdev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; @@ -1426,12 +1310,12 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, wss_port[idx] = pnp_port_start(dev, 1); dma2[idx] = pnp_dma(dev, 1); } + snd_card_set_dev(card, &pcard->card->dev); ret = create_sscape(idx, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, &pcard->card->dev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; -- 1.6.0.3 ---------------------------------------------------------------------- Nie dla nich ciepla praca za biurkiem. Sprawdz >>> http://link.interia.pl/f2383

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: McASP FIFO related updates
by Chaithrika U S 01 Oct '09

01 Oct '09

The DMA params for McASP with FIFO has been updated so that it works for various FIFO levels. A member- 'fifo_level' has been added to the DMA params data structure. The fifo_level can be adjusted by the tx[rx]_numevt platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This implementation has been tested for numevt values 1, 2, 4, 8. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-i2s.c | 2 ++ sound/soc/davinci/davinci-mcasp.c | 17 +++++++---------- sound/soc/davinci/davinci-pcm.c | 21 ++++++++++++++++++--- sound/soc/davinci/davinci-pcm.h | 1 + 4 files changed, 28 insertions(+), 13 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae7070..2ab8093 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -397,6 +397,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, } dma_params->acnt = dma_params->data_type; + dma_params->fifo_level = 0; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a..50ad051 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -714,16 +714,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; int word_length; - u8 numevt; + u8 fifo_level; davinci_hw_common_param(dev, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - numevt = dev->txnumevt; + fifo_level = dev->txnumevt; else - numevt = dev->rxnumevt; - - if (!numevt) - numevt = 1; + fifo_level = dev->rxnumevt; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -751,12 +748,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (dev->version == MCASP_VERSION_2) { - dma_params->data_type *= numevt; - dma_params->acnt = 4 * numevt; - } else + if (dev->version == MCASP_VERSION_2 && !fifo_level) + dma_params->acnt = 4; + else dma_params->acnt = dma_params->data_type; + dma_params->fifo_level = fifo_level; davinci_config_channel_size(dev, word_length); return 0; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 359e99e..1152d8b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -66,38 +66,53 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_addr_t dma_pos; dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; + unsigned short src_cidx, dst_cidx; unsigned int data_type; unsigned short acnt; unsigned int count; + unsigned int fifo_level; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; + fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; + if (fifo_level) + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 0; + src_cidx = data_type * fifo_level; + dst_cidx = 0; } else { src = prtd->params->dma_addr; dst = dma_pos; src_bidx = 0; dst_bidx = data_type; + src_cidx = 0; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; edma_set_src(lch, src, INCR, W8BIT); edma_set_dest(lch, dst, INCR, W8BIT); - edma_set_src_index(lch, src_bidx, 0); - edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + + edma_set_src_index(lch, src_bidx, src_cidx); + edma_set_dest_index(lch, dst_bidx, dst_cidx); + + if (!fifo_level) + edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + else + edma_set_transfer_params(lch, acnt, fifo_level, count, + fifo_level, ABSYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 8746606..c8b0d2b 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -23,6 +23,7 @@ struct davinci_pcm_dma_params { enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; + unsigned int fifo_level; }; -- 1.5.6

3 3

[alsa-devel] ALC268 - Acer Aspire 5720 - mic source switch disappered
by Maxim Levitsky 30 Sep '09

30 Sep '09

Hi, I have installed 2.6.32-rc1 (well -rc2...) And I notice that 'Input source' is missing. /proc/asound/Intel/codec#0 attached. Best regards, Maxim Levitsky

2 7

[PATCH] ascenario: Add scenario support to alsa-lib
by Stefan Schmidt 30 Sep '09

30 Sep '09

It allows switching audio settings between scenarios or uses-cases like listening to music and answering an incoming phone call. Made of control aliasing for playback, capture master and switch as well as the option to post- and prefix a sequence of control changes avoiding pops and other unwanted noise. Some example programs will be available in alsa-utils. CC: Ian Molton <ian(a)mnementh.co.uk> CC: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Stefan Schmidt <stefan(a)slimlogic.co.uk> --- include/Makefile.am | 2 +- include/ascenario.h | 170 +++++++ src/Makefile.am | 2 +- src/ascenario.c | 1358 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1530 insertions(+), 2 deletions(-) create mode 100644 include/ascenario.h create mode 100644 src/ascenario.c diff --git a/include/Makefile.am b/include/Makefile.am index a291503..572fbc9 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -3,7 +3,7 @@ SUBDIRS = sound sysincludedir = ${includedir}/sys alsaincludedir = ${includedir}/alsa -alsainclude_HEADERS = asoundlib.h asoundef.h \ +alsainclude_HEADERS = asoundlib.h asoundef.h ascenario.h \ version.h global.h input.h output.h error.h \ conf.h control.h iatomic.h diff --git a/include/ascenario.h b/include/ascenario.h new file mode 100644 index 0000000..869f2ea --- /dev/null +++ b/include/ascenario.h @@ -0,0 +1,170 @@ +/* +* ALSA Scenario header file +* +* This library is free software; you can redistribute it and/or modify +* it under the terms of the GNU Lesser General Public License as +* published by the Free Software Foundation; either version 2.1 of +* the License, or (at your option) any later version. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU Lesser General Public License for more details. +* +* You should have received a copy of the GNU Lesser General Public +* License along with this library; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +* Copyright (C) 2008-2009 SlimLogic Ltd +* Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> +* Stefan Schmidt <stefan(a)slimlogic.co.uk> +*/ + +/** + * Scenario IDs + * + * Standard Scenario IDs - Add new scenarios at the end. + */ + +#define SND_SCN_PLAYBACK_SPEAKER "playback speaker" +#define SND_SCN_PLAYBACK_HEADPHONES "playback headphone" +#define SND_SCN_PLAYBACK_HEADSET "playback headset" +#define SND_SCN_PLAYBACK_BLUETOOTH "playback bluetooth" +#define SND_SCN_PLAYBACK_HANDSET "playback handset" +#define SND_SCN_PLAYBACK_GSM "playback gsm" +#define SND_SCN_PLAYBACK_LINE "playback line" + +#define SND_SCN_CAPTURE_MIC "capture mic" +#define SND_SCN_CAPTURE_LINE "capture line" +#define SND_SCN_CAPTURE_HEADSET "capture headset" +#define SND_SCN_CAPTURE_HANDSET "capture handset" +#define SND_SCN_CAPTURE_BLUETOOTH "capture bluetooth" +#define SND_SCN_CAPTURE_GSM "capture gsm" + +#define SND_SCN_PHONECALL_GSM_HANDSET "phonecall gsm handset" +#define SND_SCN_PHONECALL_BT_HANDSET "phonecall bt handset" +#define SND_SCN_PHONECALL_IP_HANDSET "phonecall ip handset" +#define SND_SCN_PHONECALL_GSM_HEADSET "phonecall gsm headset" +#define SND_SCN_PHONECALL_BT_HEADSET "phonecall bt headset" +#define SND_SCN_PHONECALL_IP_HEADSET "phonecall ip headset" + +/** + * QOS + * + * Defines Audio Quality of Service. Systems supporting different types of QoS + * often have lower power consumption on lower quality levels. + */ +#define SND_POWER_QOS_HIFI 0 +#define SND_POWER_QOS_VOICE 1 +#define SND_POWER_QOS_SYSTEM 2 + +struct snd_scenario; + +/* TODO: add notification */ + +/** + * snd_scenario_list - list supported scenarios + * @scn: scenario + * @list: list of supported scenario names. + * + * List supported scenarios for this sound card. + * Returns number of scenarios. + */ +int snd_scenario_list(struct snd_scenario *scn, const char **list[]); + +/** + * snd_scenario_set_scn - set scenario + * @scn: scenario + * @scenario: scenario name + * + * Set new scenario for sound card. + */ +int snd_scenario_set_scn(struct snd_scenario *scn, const char *scenario); + +/** + * snd_scenario_get_scn - get scenario + * @scn: scenario + * + * Get current sound card scenario. + */ +const char *snd_scenario_get_scn(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_playback_volume - get playback volume + * @scn: scenario + * + * Get the master playback volume control name for the current scenario. + */ +int snd_scenario_get_master_playback_volume(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_playback_switch - get playback switch + * @scn: scenario + * + * Get the master playback switch control name for the current scenario. + */ + int snd_scenario_get_master_playback_switch(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_capture_volume - get capture volume + * @scn: scenario + * + * Get the master capture volume control name for the current scenario. + */ +int snd_scenario_get_master_capture_volume(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_capture_switch - get capture switch + * @scn: scenario + * + * Get the master capture switch control name for the current scenario. + */ +int snd_scenario_get_master_capture_switch(struct snd_scenario *scn); + +/** + * snd_scenario_set_qos - set qos + * @qos: qos + * + * Set Quality of Service for this scenario. + */ +int snd_scenario_set_qos(struct snd_scenario *scn, int qos); + +/** + * snd_scenario_get_qos - get qos + * @scn: scenario + * + * Get Quality of Service for this scenario. + */ +int snd_scenario_get_qos(struct snd_scenario *scn); + +/** + * snd_scenario_open - open scenario core + * @card_name: sound card name. + * + * Open scenario manager core for sound card. + */ +struct snd_scenario *snd_scenario_open(const char *card_name); + +/** + * snd_scenario_reload - reload and reparse scenario configuration + * @scn: scenario + * + * Reloads and reparses sound card scenario configuration. + */ +int snd_scenario_reload(struct snd_scenario *scn); + +/** + * snd_scenario_close - close scenario core + * @scn: scenario + * + * Free scenario manager core for sound card. + */ +void snd_scenario_close(struct snd_scenario *scn); + +/** + * snd_scenario_dump - dump + * @card_name: sound card name. + * + * Dump current sound card settings to stdout in scn format. + */ +int snd_scenario_dump(const char *card_name); diff --git a/src/Makefile.am b/src/Makefile.am index 3204fe4..be46cb3 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -14,7 +14,7 @@ SYMFUNCS = endif lib_LTLIBRARIES = libasound.la -libasound_la_SOURCES = conf.c confmisc.c input.c output.c async.c error.c dlmisc.c socket.c shmarea.c userfile.c names.c +libasound_la_SOURCES = conf.c confmisc.c input.c output.c async.c error.c dlmisc.c socket.c shmarea.c userfile.c names.c ascenario.c SUBDIRS=control libasound_la_LIBADD = control/libcontrol.la diff --git a/src/ascenario.c b/src/ascenario.c new file mode 100644 index 0000000..74082d4 --- /dev/null +++ b/src/ascenario.c @@ -0,0 +1,1358 @@ +/* + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + * + * Copyright (C) 2008-2009 SlimLogic Ltd + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> + * Stefan Schmidt <stefan(a)slimlogic.co.uk> + */ + +#define _GNU_SOURCE /* needed of O_NOATIME */ + +#include <stdio.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> +#include <ctype.h> +#include <alsa/asoundlib.h> + +#include "../include/ascenario.h" + +#define PRE_SEQ 0 +#define POST_SEQ 1 +#define MAX_SCN 32 +#define MAX_NAME 64 +#define MAX_FILE 256 +#define MAX_BUF 256 +#define ALSA_SCN_DIR "/etc/alsa/scenario" + +/* + * Stores all scenario settings for 1 kcontrol. Hence we have a + * control_settings for each kcontrol in card. + */ +struct control_settings { + char name[MAX_NAME]; + int id; + snd_ctl_elem_type_t type; + int count; /* 1 = mono, 2 = stereo, etc */ + unsigned short *value; /* kcontrol value 2D array */ +}; + +/* + * If sleep is 0 the element contains the settings in control. Else sleep + * contains the sleep time in micro seconds. + */ +struct sequence_element { + unsigned int sleep; /* Sleep time in msecs if sleep element, else 0 */ + struct control_settings *control; + struct sequence_element *next; /* Pointer to next list element */ +}; + +/* + * Describes default mixers and qos for scenario. + * We have a scenario_info for each scenario loaded. + */ +struct scenario_info { + char *name; + char *file; + char *pre_sequence_file; + char *post_sequence_file; + short playback_volume_id; + short playback_switch_id; + short capture_volume_id; + short capture_switch_id; + int qos; +}; + +/* Describe a snd card and all its scenarios. + */ +struct snd_scenario { + char *card_name; + int current_scenario; + int num_scenarios; /* number of supported scenarios */ + int num_kcontrols; /* number of kcontrols */ + struct sequence_element *pre_seq_list; /* Linked list for pre sequence */ + struct sequence_element *post_seq_list; /* Linked list for post sequence */ + const char **list; + struct scenario_info *scenario; /* var len array of scenario info */ + struct control_settings *control; /* var len array of controls */ +}; + +static void scn_error(const char *fmt,...) +{ + va_list va; + va_start(va, fmt); + fprintf(stderr, "scenario: "); + vfprintf(stderr, fmt, va); + va_end(va); +} + +static void scn_stdout(const char *fmt,...) +{ + va_list va; + va_start(va, fmt); + vfprintf(stdout, fmt, va); + va_end(va); +} + +static inline void set_value(struct snd_scenario *scn, + struct control_settings *control, int count, unsigned short val) +{ + int offset = scn->current_scenario * control->count; + control->value[offset + count] = val; +} + +static inline unsigned short get_value(struct snd_scenario *scn, + struct control_settings *control, int count) +{ + int offset = scn->current_scenario * control->count; + return control->value[offset + count]; +} + +static int dump_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id) +{ + int err, count, i; + snd_ctl_elem_info_t *info; + snd_ctl_elem_type_t type; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + err = snd_ctl_elem_info(handle, info); + if (err < 0) { + scn_stdout("%s: failed to get ctl info\n"); + return err; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + type = snd_ctl_elem_info_get_type(info); + count = snd_ctl_elem_info_get_count(info); + if (count == 0) + return 0; + + scn_stdout("%u:'%s':%d:", + snd_ctl_elem_id_get_numid(id), + snd_ctl_elem_id_get_name(id), count); + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_boolean(control, i)); + scn_stdout("%d", snd_ctl_elem_value_get_boolean(control, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_integer(control, i)); + scn_stdout("%d", snd_ctl_elem_value_get_integer(control, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + for (i = 0; i < count - 1; i++) + scn_stdout("%ld,", + snd_ctl_elem_value_get_integer64(control, i)); + scn_stdout("%ld", + snd_ctl_elem_value_get_integer64(control, i)); + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_enumerated(control, i)); + scn_stdout("%d", + snd_ctl_elem_value_get_enumerated(control, i)); + break; + case SND_CTL_ELEM_TYPE_BYTES: + for (i = 0; i < count - 1; i++) + scn_stdout("%2.2x,", + snd_ctl_elem_value_get_byte(control, i)); + scn_stdout("%2.2x", snd_ctl_elem_value_get_byte(control, i)); + break; + default: + break; + } + scn_stdout("\n"); + return 0; +} + +/* + * Add new kcontrol from sound card into memory database. + */ +static int add_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id, + struct control_settings *control_settings) +{ + int err; + snd_ctl_elem_info_t *info; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + err = snd_ctl_elem_info(handle, info); + if (err < 0) { + scn_stdout("%s: failed to get ctl info\n"); + return err; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + strncpy(control_settings->name, snd_ctl_elem_id_get_name(id), + MAX_NAME); + control_settings->count = snd_ctl_elem_info_get_count(info); + control_settings->type = snd_ctl_elem_info_get_type(info); + control_settings->id = snd_ctl_elem_id_get_numid(id); + return 0; +} + +static int parse_controls(struct snd_scenario *scn, FILE *f) +{ + struct control_settings *control; + char buf[MAX_BUF], name[MAX_NAME]; + int id, count, line = 1, i; + char *name_start, *name_end, *tbuf; + + while (fgets(buf, MAX_BUF, f) != NULL) { + + /* get id */ + tbuf = buf; + id = atoi(tbuf); + if (id == 0) { + scn_error("%s:id == 0 on line %d\n", __func__, line); + return -EINVAL; + } + for (i = 0; i < scn->num_kcontrols; i++) { + if (id == scn->control[i].id) { + control = &scn->control[i]; + goto get_name; + } + } + scn_error("%s:id not found at line %d\n", __func__, line); + return -EINVAL; +get_name: + /* get name start */ + while (*tbuf != 0 && *tbuf != '\'') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + name_start = ++tbuf; + + /* get name end */ + while (*tbuf != 0 && *tbuf != '\'') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + name_end = tbuf++; + + /* copy name */ + if ((name_end - name_start) > MAX_NAME) { + scn_error("%s:name too big at %d chars line %d\n", + __func__, name_end - name_start, line); + return -EINVAL; + } + strncpy(name, name_start, name_end - name_start); + name[name_end - name_start] = 0; + if (strcmp(name, control->name)) { + scn_error("%s: name %s and %s don't match at line %d\n", + __func__, name, control->name, line); + return -EINVAL; + } + + /* get count */ + tbuf++; + count = atoi(tbuf); + if (count == 0) { + scn_error("%s:count == 0 on line %d\n", __func__, + line); + return -EINVAL; + } + if (count != control->count) { + scn_error("%s:count does not match at line %d\n", + __func__, line); + return -EINVAL; + } + + /* get vals */ + control->value = malloc(control->count * scn->num_scenarios * + sizeof(unsigned short)); + if (control->value == NULL) + return -ENOMEM; + + while (*tbuf != 0 && *tbuf != ':') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + tbuf++; + + for (i = 0; i < count; i++) { + set_value(scn, control, i, atoi(tbuf)); + while (*tbuf != 0 && *tbuf != ',') + tbuf++; + + if (*tbuf++ == 0 && i < (count - 1)) + return -EINVAL; + } + line++; + } + + return 0; +} + +static char *get_string (char *buf) +{ + char *str, *end; + + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return NULL; + } + + /* find leading '"' */ + buf++; + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing start '\"'\n", __func__); + return NULL; + } + str = ++buf; + + /* get value */ + while (*buf != 0 && *buf != '"') + buf++; + end = buf; + + /* find '"' terminator */ + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing terminator '\"' %s\n", __func__, buf); + return NULL; + } + + *end = 0; + return strdup(str); +} + +static char *get_control_name (char *buf) +{ + char *str, *end; + + /* find leading '"' */ + buf++; + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing start '\"'\n", __func__); + return NULL; + } + str = ++buf; + + /* get value */ + while (*buf != 0 && *buf != '"') + buf++; + end = buf; + + /* find '"' terminator */ + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing terminator '\"' %s\n", __func__, buf); + return NULL; + } + + *end = 0; + return strdup(str); +} + +static int get_int (char *buf) +{ + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return -EINVAL; + } + buf++; + return atoi(buf); +} + +static int get_enum (char *buf) +{ + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return -EINVAL; + } + buf++; + return 0; /* TODO */ +} + +static void seq_list_append(struct snd_scenario *scn, + struct sequence_element *curr, int position) +{ + struct sequence_element *last, *tmp; + + if (position) { + if (!scn->post_seq_list) + scn->post_seq_list = curr; + + else { + tmp = scn->post_seq_list; + while (tmp) { + last = tmp; + tmp = tmp->next; + } + last->next = curr; + } + } + else { + if (!scn->pre_seq_list) { + scn->pre_seq_list = curr; + } + else { + tmp = scn->pre_seq_list; + while (tmp) { + last = tmp; + tmp = tmp->next; + } + last->next = curr; + } + } +} + +static int parse_sequences(struct snd_scenario *scn, FILE *f, int position) +{ + char buf[MAX_BUF], *tbuf, *control_value; + int control_len, i; + struct sequence_element *curr; + + while (fgets(buf, MAX_BUF, f) != NULL) { + + /* Check for lines with comments and ignore */ + if (buf[0] == '#') + continue; + + /* Parse current line and skip blanks */ + tbuf = buf; + while (isblank(*tbuf)) + tbuf++; + + curr = malloc(sizeof(struct sequence_element)); + if (curr == NULL) + return -ENOMEM; + bzero(curr, sizeof(struct sequence_element)); + + curr->control = malloc(sizeof(struct control_settings)); + if (curr->control == NULL) + return -ENOMEM; + bzero(curr->control, sizeof(struct control_settings)); + + curr->control->value = malloc(curr->control->count * scn->num_scenarios + * sizeof(unsigned short)); + if (curr->control->value == NULL) + return -ENOMEM; + bzero(curr->control->value, curr->control->count * scn->num_scenarios + * sizeof(unsigned short)); + + if (strncmp(tbuf, "kcontrol", 8) == 0) { + strncpy(curr->control->name, get_control_name(tbuf + 8), MAX_NAME); + control_len = strlen(curr->control->name); + /* 11 = 8 from kcontrol + 2 quotes + 1 blank */ + control_value = get_string(tbuf + 11 + control_len); + + for (i = 0; i < scn->num_kcontrols; i++) { + if (strncmp(curr->control->name, scn->control[i].name, + control_len) == 0) { + curr->sleep = 0; + curr->control->id = scn->control[i].id; + curr->control->type = scn->control[i].type; + curr->control->count = scn->control[i].count; + set_value(scn, curr->control, curr->control->count, + atoi(control_value)); + seq_list_append(scn, curr, position); + } + } + + continue; + } + + if (strncmp(tbuf, "msleep", 6) == 0) { + curr->sleep = get_int(tbuf + 6); + + /* Free control elements as we only have a sleep element + * here */ + if (curr->control) { + if (curr->control->value) + free(curr->control->value); + free(curr->control); + } + + seq_list_append(scn, curr, position); + continue; + } + } + + return 0; +} + +/* load scenario i */ +static int read_scenario_file(struct snd_scenario *scn) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + struct scenario_info *info = &scn->scenario[scn->current_scenario]; + + snprintf(filename, MAX_FILE, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->file); + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + scn_error("%s: couldn't open %s\n", __func__, filename); + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = parse_controls(scn, f); + fclose(f); +close: + close(fd); + return ret; +} + +static int read_sequence_file(struct snd_scenario *scn, int position) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + struct scenario_info *info = &scn->scenario[scn->current_scenario]; + + if (position == PRE_SEQ) { + sprintf(filename, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->pre_sequence_file); + } + else { + sprintf(filename, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->post_sequence_file); + } + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = parse_sequences(scn, f, position); + fclose(f); +close: + close(fd); + return ret; +} + +static int parse_scenario(struct snd_scenario *scn, FILE *f, int line_) +{ + struct scenario_info *info; + int line = line_ - 1, id = 0, file = 0; + char buf[MAX_BUF], *tbuf; + + scn->scenario = realloc(scn->scenario, + (scn->num_scenarios + 1) * sizeof(struct scenario_info)); + if (scn->scenario == NULL) + return -ENOMEM; + bzero(scn->scenario, sizeof(struct scenario_info)); + info = scn->scenario + scn->num_scenarios; + + /* Set sequence filename to NULL as it is optional and we want to check + * for NULL to avoid segfaults */ + info->pre_sequence_file = NULL; + info->post_sequence_file = NULL; + + while(fgets(buf, MAX_BUF, f) != NULL) { + + line++; + if (buf[0] == '#') + continue; + + tbuf = buf; + while (isblank(*tbuf)) + tbuf++; + + if (strncmp(tbuf, "Identifier", 10) == 0) { + info->name = get_string(tbuf + 10); + if (info->name == NULL) { + scn_error("%s: failed to get Identifer\n", + __func__); + goto err; + } + id = 1; + continue; + } + + if (strncmp(tbuf, "File", 4) == 0) { + info->file = get_string(tbuf + 4); + if (info->file == NULL) { + scn_error("%s: failed to get File\n", + __func__); + goto err; + } + file = 1; + continue; + } + + if (strncmp(tbuf, "QoS", 3) == 0) { + info->qos = get_enum(tbuf + 3); + if (info->qos < 0) { + scn_error("%s: failed to get QoS\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "MasterPlaybackVolume", 20) == 0) { + info->playback_volume_id = get_int(tbuf + 20); + if (info->playback_volume_id < 0) { + scn_error("%s: failed to get MasterPlaybackVolume\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "MasterPlaybackSwitch", 20) == 0) { + info->playback_switch_id = get_int(tbuf + 20); + if (info->playback_switch_id < 0) { + scn_error("%s: failed to get MasterPlaybackSwitch\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "MasterCaptureVolume", 19) == 0) { + info->capture_volume_id = get_int(tbuf + 19); + if (info->capture_volume_id < 0) { + scn_error("%s: failed to get MasterCaptureVolume\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "MasterCaptureSwitch", 19) == 0) { + info->capture_switch_id = get_int(tbuf + 19); + if (info->capture_switch_id < 0) { + scn_error("%s: failed to get MasterCaptureSwitch\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "PreSequenceFile", 15) == 0) { + info->pre_sequence_file = get_string(tbuf + 15); + if (info->pre_sequence_file == NULL) { + scn_error("%s: failed to get PreSequenceFile\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "PostSequenceFile", 16) == 0) { + info->post_sequence_file = get_string(tbuf + 16); + if (info->post_sequence_file == NULL) { + scn_error("%s: failed to get PostSequenceFile\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "EndSection", 10) == 0) { + break; + } + } + + if (file & id) { + scn->num_scenarios++; + return 0; + } +err: + if (file) { + free(info->file); + info->file = NULL; + } + if (id) { + free(info->name); + info->name = NULL; + } + return -EINVAL; +} + +static int read_master_file(struct snd_scenario *scn, FILE *f) +{ + int line = 0, ret = 0, i; + char buf[MAX_BUF], *tbuf; + + /* parse master config sections */ + while(fgets(buf, MAX_BUF, f) != NULL) { + + if (buf[0] == '#') { + line++; + continue; + } + + if (strncmp(buf, "Section", 7) == 0) { + + tbuf = buf + 7; + while (isblank(*tbuf)) + tbuf++; + + if (strncmp(tbuf, "\"Scenario\"", 10) == 0) { + line = parse_scenario(scn, f, line); + if (line < 0) { + scn_error("%s: failed to parse " + "scenario\n", __func__); + goto err; + } + continue; + } + } + line++; + } + + /* copy ptrs to scenario names */ + scn->list = malloc(scn->num_scenarios * sizeof(char *)); + if (scn->list == NULL) + ret = -ENOMEM; + for (i = 0; i < scn->num_scenarios; i++) + scn->list[i] = scn->scenario[i].name; + +err: + if (ferror(f)) { + scn_error("%s: failed to read master\n", __func__); + return ferror(f); + } + return ret; +} + +/* load scenario i */ +static int import_master_config(struct snd_scenario *scn) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + + sprintf(filename, "%s/%s.conf", ALSA_SCN_DIR, scn->card_name); + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + scn_error("%s: couldn't open %s\n", __func__, filename); + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = read_master_file(scn, f); + fclose(f); +close: + close(fd); + return ret; +} + +/* parse_card_controls + * @scn: scenario + * + * Parse sound card and store control data in memory db. + */ +static int parse_card_controls(struct snd_scenario *scn) +{ + struct control_settings *control; + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + int ret, i; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* open and load snd card */ + ret = snd_ctl_open(&handle, scn->card_name, SND_CTL_READONLY); + if (ret < 0) { + scn_error("%s: control %s open retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + scn->num_kcontrols = snd_ctl_elem_list_get_count(list); + if (scn->num_kcontrols < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, scn->num_kcontrols) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* allocate db memory for controls */ + scn->control = calloc(scn->num_kcontrols, + sizeof(struct control_settings)); + if (scn->control == NULL) { + ret = -ENOMEM; + goto free; + } + control = scn->control; + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < scn->num_kcontrols; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = add_control(handle, id, control++); + if (ret < 0) { + scn_error("%s: failed to add control error %s\n", + __func__, snd_strerror(ret)); + goto close; + } + } +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +/* import_scenario_files - + * @scn: scenario + * + * Read and parse scenario_info files the store in memory. + */ +static int import_scenario_files(struct snd_scenario *scn) +{ + int ret; + + ret = import_master_config(scn); + if (ret < 0) { + scn_error("%s: failed to parse master scenario config\n", + __func__); + return ret; + } + + for (scn->current_scenario = 0; + scn->current_scenario < scn->num_scenarios; + scn->current_scenario++) { + + ret = read_scenario_file(scn); + if (ret < 0) { + scn_error("%s: failed to parse scenario %s\n", + __func__, + scn->scenario[scn->current_scenario].name); + scn->current_scenario = -1; + return ret; + } + + if (scn->scenario[scn->current_scenario].pre_sequence_file != NULL) { + ret = read_sequence_file(scn, PRE_SEQ); + if (ret < 0) { + scn_stdout("Warning: PreSequence file defined but" + " missing in scenario \"%s\"\n", + scn->scenario[scn->current_scenario].name); + } + } + + if (scn->scenario[scn->current_scenario].post_sequence_file != NULL) { + ret = read_sequence_file(scn, POST_SEQ); + if (ret < 0) { + scn_stdout("Warning: PostSequence file defined but" + " missing in scenario \"%s\"\n", + scn->scenario[scn->current_scenario].name); + } + } + + } + return 0; +} + +/* free all resorces */ +static void free_scn(struct snd_scenario *scn) +{ + /* TODO: valgrind to make sure. */ + int i; + + if (scn == NULL) + return; + + if (scn->control) { + if (scn->control->value) + free(scn->control->value); + free(scn->control); + } + + if (scn->list) + free(scn->list); + if (scn->card_name) + free(scn->card_name); + if (scn->pre_seq_list) + free(scn->pre_seq_list); + if (scn->post_seq_list) + free(scn->post_seq_list); + + if (scn->scenario) { + for (i = 0; i < scn->num_scenarios; i++) { + struct scenario_info *info = &scn->scenario[i]; + + if (info->name) + free(info->name); + if (info->file) + free(info->file); + if (info->pre_sequence_file) + free(info->pre_sequence_file); + if (info->post_sequence_file) + free(info->post_sequence_file); + } + free(scn->scenario); + } + free(scn); +} + +/* + * Init sound card scenario db. + */ +struct snd_scenario *snd_scenario_open(const char *card_name) +{ + struct snd_scenario *scn; + int err; + + /* TODO: locking and + * check if card_name scn is already loaded, + * if so reuse to conserve ram. */ + + scn = malloc(sizeof(struct snd_scenario)); + if (scn == NULL) + return NULL; + bzero(scn, sizeof(struct snd_scenario)); + scn->card_name = strdup(card_name); + if (scn->card_name == NULL) { + free(scn); + return NULL; + } + + /* get info about sound card */ + err = parse_card_controls(scn); + if (err < 0) { + free_scn(scn); + return NULL; + } + + /* get info on scenarios and verify against card */ + err = import_scenario_files(scn); + if (err < 0) { + free_scn(scn); + return NULL; + } + + return scn; +} + +/* + * Reload and reparse scenario db. + */ +int snd_scenario_reload(struct snd_scenario *scn) +{ + free_scn(scn); + + scn->num_kcontrols = parse_card_controls(scn); + if (scn->num_kcontrols <= 0) { + free_scn(scn); + return -EINVAL; + } + + scn->num_scenarios = import_scenario_files(scn); + if (scn->num_scenarios <= 0) { + return -EINVAL; + } + + return 0; +} + +void snd_scenario_close(struct snd_scenario *scn) +{ + free_scn(scn); +} + +static int set_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id, + struct snd_scenario *scn) +{ + struct control_settings *setting; + int ret, count, i, idnum; + snd_ctl_elem_info_t *info; + snd_ctl_elem_type_t type; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + ret = snd_ctl_elem_info(handle, info); + if (ret < 0) { + scn_error("%s: failed to get ctl info\n", __func__); + return ret; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + idnum = snd_ctl_elem_id_get_numid(id); + for (i = 0; i < scn->num_kcontrols; i++) { + setting = &scn->control[i]; + if (setting->id == idnum) + goto set_val; + } + scn_error("%s: failed to find control %d\n", __func__, idnum); + return 0; + +set_val: + type = snd_ctl_elem_info_get_type(info); + count = snd_ctl_elem_info_get_count(info); + if (count == 0) + return 0; + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_boolean(control, i, + get_value(scn, setting, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_integer(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_integer64(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_enumerated(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_BYTES: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_byte(control, i, + get_value(scn, setting, i)); + break; + default: + break; + } + + ret = snd_ctl_elem_write(handle, control); + if (ret < 0) { + scn_error("%s: control %s failed: %s\n", __func__, + setting->name, snd_strerror(ret)); + scn_error("%s: count %d type: %d\n", __func__, + count, type); + for (i = 0; i < count; i++) + fprintf(stderr, "%d ", get_value(scn, setting, i)); + return ret; + } + return 0; +} + +static void exec_sequence(struct sequence_element *seq, struct snd_scenario + *scn, snd_ctl_elem_list_t *list, snd_ctl_t *handle) +{ + int count = snd_ctl_elem_list_get_count(list); + while (seq) { + if (seq->sleep) + usleep(seq->sleep); + else { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + int ret, i, numid; + /* Where is id lookup from numid if you need it? */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_list_get_id(list, i, id); + numid = snd_ctl_elem_id_get_numid(id); + if (numid == seq->control->id) { + ret = set_control(handle, id, scn); + if (ret < 0) { + scn_error("%s: failed to set control %s\n", + __func__, scn->card_name); + } + break; + } + } + } + seq = seq->next; + } +} + +int snd_scenario_set_scn(struct snd_scenario *scn, const char *name) +{ + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + snd_ctl_t *handle; + int ret, count, i; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* find scenario name */ + for (i = 0; i < scn->num_scenarios; i++) { + if (!strcmp(scn->scenario[i].name, name)) + goto found; + } + scn_error("%s: scenario %s not found\n", __func__, name); + return -EINVAL; + +found: + /* scenario found - now open card */ + scn->current_scenario = i; + ret = snd_ctl_open(&handle, scn->card_name, 0); + if (ret) { + scn_error("%s: control %s open error: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + count = snd_ctl_elem_list_get_count(list); + if (count < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, count) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* If we have a sequence list that should be executed before the new + * scenario is set do it now */ + if (scn->pre_seq_list) + exec_sequence(scn->pre_seq_list, scn, list, handle); + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = set_control(handle, id, scn); + if (ret < 0) { + scn_error("%s: failed to set control %s\n", __func__, + scn->card_name); + } + } + + /* If we have a sequence list that should be executed after the new + * scenario is set do it now */ + if (scn->post_seq_list) + exec_sequence(scn->post_seq_list, scn, list, handle); + +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +int snd_scenario_dump(const char *card_name) +{ + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + int ret, i, count; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* open and load snd card */ + ret = snd_ctl_open(&handle, card_name, SND_CTL_READONLY); + if (ret < 0) { + scn_error("%s: control %s open retor: %s\n", __func__, card_name, + snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + count = snd_ctl_elem_list_get_count(list); + if (count < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, count) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = dump_control(handle, id); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", + __func__, snd_strerror(ret)); + goto free; + } + } +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +const char *snd_scenario_get_scn(struct snd_scenario *scn) +{ + if (scn->current_scenario > 0 && scn->current_scenario < MAX_SCN) + return scn->scenario[scn->current_scenario].name; + else + return NULL; +} + +int snd_scenario_set_qos(struct snd_scenario *scn, int qos) +{ + /* TODO: change QoS kcontrols */ + scn->scenario[scn->current_scenario].qos = qos; + return 0; +} + +int snd_scenario_get_qos(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].qos; +} + +int snd_scenario_get_master_playback_volume(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].playback_volume_id; +} + +int snd_scenario_get_master_playback_switch(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].playback_switch_id; +} + +int snd_scenario_get_master_capture_volume(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].capture_volume_id; +} + +int snd_scenario_get_master_capture_switch(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].capture_switch_id; +} + +int snd_scenario_list(struct snd_scenario *scn, const char **list[]) +{ + *list = scn->list; + return scn->num_scenarios; +} -- 1.6.4.3 regards Stefan Schmidt

1 0

[PATCH] ascenario: Add scenario support to alsa-lib
by Stefan Schmidt 30 Sep '09

30 Sep '09

It allows switching audio settings between scenarios or uses-cases like listening to music and answering an incoming phone call. Made of control aliasing for playback, capture master and switch as well as the option to post- and prefix a sequence of control changes avoiding pops and other unwanted noise. Some example programs will be available in alsa-utils. CC: Ian Molton <ian(a)mnementh.co.uk> CC: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Stefan Schmidt <stefan(a)slimlogic.co.uk> --- include/Makefile.am | 2 +- include/ascenario.h | 170 +++++++ src/Makefile.am | 2 +- src/ascenario.c | 1360 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1532 insertions(+), 2 deletions(-) create mode 100644 include/ascenario.h create mode 100644 src/ascenario.c diff --git a/include/Makefile.am b/include/Makefile.am index a291503..572fbc9 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -3,7 +3,7 @@ SUBDIRS = sound sysincludedir = ${includedir}/sys alsaincludedir = ${includedir}/alsa -alsainclude_HEADERS = asoundlib.h asoundef.h \ +alsainclude_HEADERS = asoundlib.h asoundef.h ascenario.h \ version.h global.h input.h output.h error.h \ conf.h control.h iatomic.h diff --git a/include/ascenario.h b/include/ascenario.h new file mode 100644 index 0000000..869f2ea --- /dev/null +++ b/include/ascenario.h @@ -0,0 +1,170 @@ +/* +* ALSA Scenario header file +* +* This library is free software; you can redistribute it and/or modify +* it under the terms of the GNU Lesser General Public License as +* published by the Free Software Foundation; either version 2.1 of +* the License, or (at your option) any later version. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU Lesser General Public License for more details. +* +* You should have received a copy of the GNU Lesser General Public +* License along with this library; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +* Copyright (C) 2008-2009 SlimLogic Ltd +* Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> +* Stefan Schmidt <stefan(a)slimlogic.co.uk> +*/ + +/** + * Scenario IDs + * + * Standard Scenario IDs - Add new scenarios at the end. + */ + +#define SND_SCN_PLAYBACK_SPEAKER "playback speaker" +#define SND_SCN_PLAYBACK_HEADPHONES "playback headphone" +#define SND_SCN_PLAYBACK_HEADSET "playback headset" +#define SND_SCN_PLAYBACK_BLUETOOTH "playback bluetooth" +#define SND_SCN_PLAYBACK_HANDSET "playback handset" +#define SND_SCN_PLAYBACK_GSM "playback gsm" +#define SND_SCN_PLAYBACK_LINE "playback line" + +#define SND_SCN_CAPTURE_MIC "capture mic" +#define SND_SCN_CAPTURE_LINE "capture line" +#define SND_SCN_CAPTURE_HEADSET "capture headset" +#define SND_SCN_CAPTURE_HANDSET "capture handset" +#define SND_SCN_CAPTURE_BLUETOOTH "capture bluetooth" +#define SND_SCN_CAPTURE_GSM "capture gsm" + +#define SND_SCN_PHONECALL_GSM_HANDSET "phonecall gsm handset" +#define SND_SCN_PHONECALL_BT_HANDSET "phonecall bt handset" +#define SND_SCN_PHONECALL_IP_HANDSET "phonecall ip handset" +#define SND_SCN_PHONECALL_GSM_HEADSET "phonecall gsm headset" +#define SND_SCN_PHONECALL_BT_HEADSET "phonecall bt headset" +#define SND_SCN_PHONECALL_IP_HEADSET "phonecall ip headset" + +/** + * QOS + * + * Defines Audio Quality of Service. Systems supporting different types of QoS + * often have lower power consumption on lower quality levels. + */ +#define SND_POWER_QOS_HIFI 0 +#define SND_POWER_QOS_VOICE 1 +#define SND_POWER_QOS_SYSTEM 2 + +struct snd_scenario; + +/* TODO: add notification */ + +/** + * snd_scenario_list - list supported scenarios + * @scn: scenario + * @list: list of supported scenario names. + * + * List supported scenarios for this sound card. + * Returns number of scenarios. + */ +int snd_scenario_list(struct snd_scenario *scn, const char **list[]); + +/** + * snd_scenario_set_scn - set scenario + * @scn: scenario + * @scenario: scenario name + * + * Set new scenario for sound card. + */ +int snd_scenario_set_scn(struct snd_scenario *scn, const char *scenario); + +/** + * snd_scenario_get_scn - get scenario + * @scn: scenario + * + * Get current sound card scenario. + */ +const char *snd_scenario_get_scn(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_playback_volume - get playback volume + * @scn: scenario + * + * Get the master playback volume control name for the current scenario. + */ +int snd_scenario_get_master_playback_volume(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_playback_switch - get playback switch + * @scn: scenario + * + * Get the master playback switch control name for the current scenario. + */ + int snd_scenario_get_master_playback_switch(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_capture_volume - get capture volume + * @scn: scenario + * + * Get the master capture volume control name for the current scenario. + */ +int snd_scenario_get_master_capture_volume(struct snd_scenario *scn); + +/** + * snd_scenario_get_master_capture_switch - get capture switch + * @scn: scenario + * + * Get the master capture switch control name for the current scenario. + */ +int snd_scenario_get_master_capture_switch(struct snd_scenario *scn); + +/** + * snd_scenario_set_qos - set qos + * @qos: qos + * + * Set Quality of Service for this scenario. + */ +int snd_scenario_set_qos(struct snd_scenario *scn, int qos); + +/** + * snd_scenario_get_qos - get qos + * @scn: scenario + * + * Get Quality of Service for this scenario. + */ +int snd_scenario_get_qos(struct snd_scenario *scn); + +/** + * snd_scenario_open - open scenario core + * @card_name: sound card name. + * + * Open scenario manager core for sound card. + */ +struct snd_scenario *snd_scenario_open(const char *card_name); + +/** + * snd_scenario_reload - reload and reparse scenario configuration + * @scn: scenario + * + * Reloads and reparses sound card scenario configuration. + */ +int snd_scenario_reload(struct snd_scenario *scn); + +/** + * snd_scenario_close - close scenario core + * @scn: scenario + * + * Free scenario manager core for sound card. + */ +void snd_scenario_close(struct snd_scenario *scn); + +/** + * snd_scenario_dump - dump + * @card_name: sound card name. + * + * Dump current sound card settings to stdout in scn format. + */ +int snd_scenario_dump(const char *card_name); diff --git a/src/Makefile.am b/src/Makefile.am index 3204fe4..be46cb3 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -14,7 +14,7 @@ SYMFUNCS = endif lib_LTLIBRARIES = libasound.la -libasound_la_SOURCES = conf.c confmisc.c input.c output.c async.c error.c dlmisc.c socket.c shmarea.c userfile.c names.c +libasound_la_SOURCES = conf.c confmisc.c input.c output.c async.c error.c dlmisc.c socket.c shmarea.c userfile.c names.c ascenario.c SUBDIRS=control libasound_la_LIBADD = control/libcontrol.la diff --git a/src/ascenario.c b/src/ascenario.c new file mode 100644 index 0000000..99f239e --- /dev/null +++ b/src/ascenario.c @@ -0,0 +1,1360 @@ +/* + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + * + * Copyright (C) 2008-2009 SlimLogic Ltd + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> + * Stefan Schmidt <stefan(a)slimlogic.co.uk> + */ + +#define _GNU_SOURCE /* needed of O_NOATIME */ + +#include <stdio.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> +#include <ctype.h> +#include <alsa/asoundlib.h> + +#include "../include/ascenario.h" + +#define PRE_SEQ 0 +#define POST_SEQ 1 +#define MAX_SCN 32 +#define MAX_NAME 64 +#define MAX_FILE 256 +#define MAX_BUF 256 +#define ALSA_SCN_DIR "/etc/alsa/scenario" + +/* + * Stores all scenario settings for 1 kcontrol. Hence we have a + * control_settings for each kcontrol in card. + */ +struct control_settings { + char name[MAX_NAME]; + int id; + snd_ctl_elem_type_t type; + int count; /* 1 = mono, 2 = stereo, etc */ + unsigned short *value; /* kcontrol value 2D array */ +}; + +/* + * If sleep is 0 the element contains the settings in control. Else sleep + * contains the sleep time in micro seconds. + */ +struct sequence_element { + unsigned int sleep; /* Sleep time in msecs if sleep element, else 0 */ + struct control_settings *control; + struct sequence_element *next; /* Pointer to next list element */ +}; + +/* + * Describes default mixers and qos for scenario. + * We have a scenario_info for each scenario loaded. + */ +struct scenario_info { + char *name; + char *file; + char *pre_sequence_file; + char *post_sequence_file; + short playback_volume_id; + short playback_switch_id; + short capture_volume_id; + short capture_switch_id; + int qos; +}; + +/* Describe a snd card and all its scenarios. + */ +struct snd_scenario { + char *card_name; + int current_scenario; + int num_scenarios; /* number of supported scenarios */ + int num_kcontrols; /* number of kcontrols */ + struct sequence_element *pre_seq_list; /* Linked list for pre sequence */ + struct sequence_element *post_seq_list; /* Linked list for post sequence */ + const char **list; + struct scenario_info *scenario; /* var len array of scenario info */ + struct control_settings *control; /* var len array of controls */ +}; + +static void scn_error(const char *fmt,...) +{ + va_list va; + va_start(va, fmt); + fprintf(stderr, "scenario: "); + vfprintf(stderr, fmt, va); + va_end(va); +} + +static void scn_stdout(const char *fmt,...) +{ + va_list va; + va_start(va, fmt); + vfprintf(stdout, fmt, va); + va_end(va); +} + +static inline void set_value(struct snd_scenario *scn, + struct control_settings *control, int count, unsigned short val) +{ + int offset = scn->current_scenario * control->count; + control->value[offset + count] = val; +} + +static inline unsigned short get_value(struct snd_scenario *scn, + struct control_settings *control, int count) +{ + int offset = scn->current_scenario * control->count; + return control->value[offset + count]; +} + +static int dump_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id) +{ + int err, count, i; + snd_ctl_elem_info_t *info; + snd_ctl_elem_type_t type; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + err = snd_ctl_elem_info(handle, info); + if (err < 0) { + scn_stdout("%s: failed to get ctl info\n"); + return err; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + type = snd_ctl_elem_info_get_type(info); + count = snd_ctl_elem_info_get_count(info); + if (count == 0) + return 0; + + scn_stdout("%u:'%s':%d:", + snd_ctl_elem_id_get_numid(id), + snd_ctl_elem_id_get_name(id), count); + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_boolean(control, i)); + scn_stdout("%d", snd_ctl_elem_value_get_boolean(control, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_integer(control, i)); + scn_stdout("%d", snd_ctl_elem_value_get_integer(control, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + for (i = 0; i < count - 1; i++) + scn_stdout("%ld,", + snd_ctl_elem_value_get_integer64(control, i)); + scn_stdout("%ld", + snd_ctl_elem_value_get_integer64(control, i)); + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + for (i = 0; i < count - 1; i++) + scn_stdout("%d,", + snd_ctl_elem_value_get_enumerated(control, i)); + scn_stdout("%d", + snd_ctl_elem_value_get_enumerated(control, i)); + break; + case SND_CTL_ELEM_TYPE_BYTES: + for (i = 0; i < count - 1; i++) + scn_stdout("%2.2x,", + snd_ctl_elem_value_get_byte(control, i)); + scn_stdout("%2.2x", snd_ctl_elem_value_get_byte(control, i)); + break; + default: + break; + } + scn_stdout("\n"); + return 0; +} + +/* + * Add new kcontrol from sound card into memory database. + */ +static int add_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id, + struct control_settings *control_settings) +{ + int err; + snd_ctl_elem_info_t *info; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + err = snd_ctl_elem_info(handle, info); + if (err < 0) { + scn_stdout("%s: failed to get ctl info\n"); + return err; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + strncpy(control_settings->name, snd_ctl_elem_id_get_name(id), + MAX_NAME); + control_settings->count = snd_ctl_elem_info_get_count(info); + control_settings->type = snd_ctl_elem_info_get_type(info); + control_settings->id = snd_ctl_elem_id_get_numid(id); + return 0; +} + +static int parse_controls(struct snd_scenario *scn, FILE *f) +{ + struct control_settings *control; + char buf[MAX_BUF], name[MAX_NAME]; + int id, count, line = 1, i; + char *name_start, *name_end, *tbuf; + + while (fgets(buf, MAX_BUF, f) != NULL) { + + /* get id */ + tbuf = buf; + id = atoi(tbuf); + if (id == 0) { + scn_error("%s:id == 0 on line %d\n", __func__, line); + return -EINVAL; + } + for (i = 0; i < scn->num_kcontrols; i++) { + if (id == scn->control[i].id) { + control = &scn->control[i]; + goto get_name; + } + } + scn_error("%s:id not found at line %d\n", __func__, line); + return -EINVAL; +get_name: + /* get name start */ + while (*tbuf != 0 && *tbuf != '\'') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + name_start = ++tbuf; + + /* get name end */ + while (*tbuf != 0 && *tbuf != '\'') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + name_end = tbuf++; + + /* copy name */ + if ((name_end - name_start) > MAX_NAME) { + scn_error("%s:name too big at %d chars line %d\n", + __func__, name_end - name_start, line); + return -EINVAL; + } + strncpy(name, name_start, name_end - name_start); + name[name_end - name_start] = 0; + if (strcmp(name, control->name)) { + scn_error("%s: name %s and %s don't match at line %d\n", + __func__, name, control->name, line); + return -EINVAL; + } + + /* get count */ + tbuf++; + count = atoi(tbuf); + if (count == 0) { + scn_error("%s:count == 0 on line %d\n", __func__, + line); + return -EINVAL; + } + if (count != control->count) { + scn_error("%s:count does not match at line %d\n", + __func__, line); + return -EINVAL; + } + + /* get vals */ + control->value = malloc(control->count * scn->num_scenarios * + sizeof(unsigned short)); + if (control->value == NULL) + return -ENOMEM; + + while (*tbuf != 0 && *tbuf != ':') + tbuf++; + if (*tbuf == 0) + return -EINVAL; + tbuf++; + + for (i = 0; i < count; i++) { + set_value(scn, control, i, atoi(tbuf)); + while (*tbuf != 0 && *tbuf != ',') + tbuf++; + + if (*tbuf++ == 0 && i < (count - 1)) + return -EINVAL; + } + line++; + } + + return 0; +} + +static char *get_string (char *buf) +{ + char *str, *end; + + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return NULL; + } + + /* find leading '"' */ + buf++; + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing start '\"'\n", __func__); + return NULL; + } + str = ++buf; + + /* get value */ + while (*buf != 0 && *buf != '"') + buf++; + end = buf; + + /* find '"' terminator */ + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing terminator '\"' %s\n", __func__, buf); + return NULL; + } + + *end = 0; + return strdup(str); +} + +static char *get_control_name (char *buf) +{ + char *str, *end; + + /* find leading '"' */ + buf++; + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing start '\"'\n", __func__); + return NULL; + } + str = ++buf; + + /* get value */ + while (*buf != 0 && *buf != '"') + buf++; + end = buf; + + /* find '"' terminator */ + if (*buf == 0 || *buf != '"') { + scn_error("%s: missing terminator '\"' %s\n", __func__, buf); + return NULL; + } + + *end = 0; + return strdup(str); +} + +static int get_int (char *buf) +{ + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return -EINVAL; + } + buf++; + return atoi(buf); +} + +static int get_enum (char *buf) +{ + /* find '=' */ + while (isblank(*buf)) + buf++; + if (*buf == 0 || *buf != '=') { + scn_error("%s: missing '='\n", __func__); + return -EINVAL; + } + buf++; + return 0; /* TODO */ +} + +static void seq_list_append(struct snd_scenario *scn, + struct sequence_element *curr, int position) +{ + struct sequence_element *last, *tmp; + + if (position) { + if (!scn->post_seq_list) + scn->post_seq_list = curr; + + else { + tmp = scn->post_seq_list; + while (tmp) { + last = tmp; + tmp = tmp->next; + } + last->next = curr; + } + } + else { + if (!scn->pre_seq_list) { + scn->pre_seq_list = curr; + } + else { + tmp = scn->pre_seq_list; + while (tmp) { + last = tmp; + tmp = tmp->next; + } + last->next = curr; + } + } +} + +static int parse_sequences(struct snd_scenario *scn, FILE *f, int position) +{ + char buf[MAX_BUF], *tbuf, *control_value; + int control_len, i; + struct sequence_element *curr; + + while (fgets(buf, MAX_BUF, f) != NULL) { + + /* Check for lines with comments and ignore */ + if (buf[0] == '#') + continue; + + /* Parse current line and skip blanks */ + tbuf = buf; + while (isblank(*tbuf)) + tbuf++; + + curr = malloc(sizeof(struct sequence_element)); + if (curr == NULL) + return -ENOMEM; + bzero(curr, sizeof(struct sequence_element)); + + curr->control = malloc(sizeof(struct control_settings)); + if (curr->control == NULL) + return -ENOMEM; + bzero(curr->control, sizeof(struct control_settings)); + + curr->control->value = malloc(curr->control->count * scn->num_scenarios + * sizeof(unsigned short)); + if (curr->control->value == NULL) + return -ENOMEM; + bzero(curr->control->value, curr->control->count * scn->num_scenarios + * sizeof(unsigned short)); + + if (strncmp(tbuf, "kcontrol", 8) == 0) { + strncpy(curr->control->name, get_control_name(tbuf + 8), MAX_NAME); + control_len = strlen(curr->control->name); + /* 11 = 8 from kcontrol + 2 quotes + 1 blank */ + control_value = get_string(tbuf + 11 + control_len); + + for (i = 0; i < scn->num_kcontrols; i++) { + if (strncmp(curr->control->name, scn->control[i].name, + control_len) == 0) { + curr->sleep = 0; + curr->control->id = scn->control[i].id; + curr->control->type = scn->control[i].type; + curr->control->count = scn->control[i].count; + set_value(scn, curr->control, curr->control->count, + atoi(control_value)); + seq_list_append(scn, curr, position); + } + } + + continue; + } + + if (strncmp(tbuf, "msleep", 6) == 0) { + curr->sleep = get_int(tbuf + 6); + + /* Free control elements as we only have a sleep element + * here */ + if (curr->control) { + if (curr->control->value) + free(curr->control->value); + free(curr->control); + } + + seq_list_append(scn, curr, position); + continue; + } + } + + return 0; +} + +/* load scenario i */ +static int read_scenario_file(struct snd_scenario *scn) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + struct scenario_info *info = &scn->scenario[scn->current_scenario]; + + snprintf(filename, MAX_FILE, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->file); + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + scn_error("%s: couldn't open %s\n", __func__, filename); + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = parse_controls(scn, f); + fclose(f); +close: + close(fd); + return ret; +} + +static int read_sequence_file(struct snd_scenario *scn, int position) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + struct scenario_info *info = &scn->scenario[scn->current_scenario]; + + if (position == PRE_SEQ) { + sprintf(filename, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->pre_sequence_file); + } + else { + sprintf(filename, "%s/%s/%s", ALSA_SCN_DIR, scn->card_name, + info->post_sequence_file); + } + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = parse_sequences(scn, f, position); + fclose(f); +close: + close(fd); + return ret; +} + +static int parse_scenario(struct snd_scenario *scn, FILE *f, int line_) +{ + struct scenario_info *info; + int line = line_ - 1, id = 0, file = 0, len, offset; + char buf[MAX_BUF], *tbuf; + + offset = sizeof(scn->scenario); + scn->scenario = realloc(scn->scenario, + (scn->num_scenarios + 1) * sizeof(struct scenario_info)); + len = sizeof(scn->scenario) - offset; + if (scn->scenario == NULL) + return -ENOMEM; + bzero(scn->scenario + offset, len); + info = scn->scenario + scn->num_scenarios; + + /* Set sequence filename to NULL as it is optional and we want to check + * for NULL to avoid segfaults */ + info->pre_sequence_file = NULL; + info->post_sequence_file = NULL; + + while(fgets(buf, MAX_BUF, f) != NULL) { + + line++; + if (buf[0] == '#') + continue; + + tbuf = buf; + while (isblank(*tbuf)) + tbuf++; + + if (strncmp(tbuf, "Identifier", 10) == 0) { + info->name = get_string(tbuf + 10); + if (info->name == NULL) { + scn_error("%s: failed to get Identifer\n", + __func__); + goto err; + } + id = 1; + continue; + } + + if (strncmp(tbuf, "File", 4) == 0) { + info->file = get_string(tbuf + 4); + if (info->file == NULL) { + scn_error("%s: failed to get File\n", + __func__); + goto err; + } + file = 1; + continue; + } + + if (strncmp(tbuf, "QoS", 3) == 0) { + info->qos = get_enum(tbuf + 3); + if (info->qos < 0) { + scn_error("%s: failed to get QoS\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "Master Playback Volume", 22) == 0) { + info->playback_volume_id = get_int(tbuf + 22); + if (info->playback_volume_id < 0) { + scn_error("%s: failed to get Master Playback Volume\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "Master Playback Switch", 22) == 0) { + info->playback_switch_id = get_int(tbuf + 22); + if (info->playback_switch_id < 0) { + scn_error("%s: failed to get Master Playback Switch\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "Master Capture Volume", 21) == 0) { + info->capture_volume_id = get_int(tbuf + 21); + if (info->capture_volume_id < 0) { + scn_error("%s: failed to get Master Capture Volume\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "Master Capture Switch", 21) == 0) { + info->capture_switch_id = get_int(tbuf + 21); + if (info->capture_switch_id < 0) { + scn_error("%s: failed to get Master Capture Switch\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "PreSequenceFile", 15) == 0) { + info->pre_sequence_file = get_string(tbuf + 15); + if (info->pre_sequence_file == NULL) { + scn_error("%s: failed to get PreSequenceFile\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "PostSequenceFile", 16) == 0) { + info->post_sequence_file = get_string(tbuf + 16); + if (info->post_sequence_file == NULL) { + scn_error("%s: failed to get PostSequenceFile\n", + __func__); + goto err; + } + continue; + } + + if (strncmp(tbuf, "EndSection", 10) == 0) { + break; + } + } + + if (file & id) { + scn->num_scenarios++; + return 0; + } +err: + if (file) { + free(info->file); + info->file = NULL; + } + if (id) { + free(info->name); + info->name = NULL; + } + return -EINVAL; +} + +static int read_master_file(struct snd_scenario *scn, FILE *f) +{ + int line = 0, ret = 0, i; + char buf[MAX_BUF], *tbuf; + + /* parse master config sections */ + while(fgets(buf, MAX_BUF, f) != NULL) { + + if (buf[0] == '#') { + line++; + continue; + } + + if (strncmp(buf, "Section", 7) == 0) { + + tbuf = buf + 7; + while (isblank(*tbuf)) + tbuf++; + + if (strncmp(tbuf, "\"Scenario\"", 10) == 0) { + line = parse_scenario(scn, f, line); + if (line < 0) { + scn_error("%s: failed to parse " + "scenario\n", __func__); + goto err; + } + continue; + } + } + line++; + } + + /* copy ptrs to scenario names */ + scn->list = malloc(scn->num_scenarios * sizeof(char *)); + if (scn->list == NULL) + ret = -ENOMEM; + for (i = 0; i < scn->num_scenarios; i++) + scn->list[i] = scn->scenario[i].name; + +err: + if (ferror(f)) { + scn_error("%s: failed to read master\n", __func__); + return ferror(f); + } + return ret; +} + +/* load scenario i */ +static int import_master_config(struct snd_scenario *scn) +{ + int fd, ret; + FILE *f; + char filename[MAX_FILE]; + + sprintf(filename, "%s/%s.conf", ALSA_SCN_DIR, scn->card_name); + + fd = open(filename, O_RDONLY | O_NOATIME); + if (fd < 0) { + scn_error("%s: couldn't open %s\n", __func__, filename); + return fd; + } + + f = fdopen(fd, "r"); + if (f == NULL) { + ret = errno; + goto close; + } + + ret = read_master_file(scn, f); + fclose(f); +close: + close(fd); + return ret; +} + +/* parse_card_controls + * @scn: scenario + * + * Parse sound card and store control data in memory db. + */ +static int parse_card_controls(struct snd_scenario *scn) +{ + struct control_settings *control; + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + int ret, i; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* open and load snd card */ + ret = snd_ctl_open(&handle, scn->card_name, SND_CTL_READONLY); + if (ret < 0) { + scn_error("%s: control %s open retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + scn->num_kcontrols = snd_ctl_elem_list_get_count(list); + if (scn->num_kcontrols < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, scn->num_kcontrols) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* allocate db memory for controls */ + scn->control = calloc(scn->num_kcontrols, + sizeof(struct control_settings)); + if (scn->control == NULL) { + ret = -ENOMEM; + goto free; + } + control = scn->control; + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < scn->num_kcontrols; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = add_control(handle, id, control++); + if (ret < 0) { + scn_error("%s: failed to add control error %s\n", + __func__, snd_strerror(ret)); + goto close; + } + } +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +/* import_scenario_files - + * @scn: scenario + * + * Read and parse scenario_info files the store in memory. + */ +static int import_scenario_files(struct snd_scenario *scn) +{ + int ret; + + ret = import_master_config(scn); + if (ret < 0) { + scn_error("%s: failed to parse master scenario config\n", + __func__); + return ret; + } + + for (scn->current_scenario = 0; + scn->current_scenario < scn->num_scenarios; + scn->current_scenario++) { + + ret = read_scenario_file(scn); + if (ret < 0) { + scn_error("%s: failed to parse scenario %s\n", + __func__, + scn->scenario[scn->current_scenario].name); + scn->current_scenario = -1; + return ret; + } + + if (scn->scenario[scn->current_scenario].pre_sequence_file != NULL) { + ret = read_sequence_file(scn, PRE_SEQ); + if (ret < 0) { + scn_stdout("Warning: PreSequence file defined but" + " missing in scenario \"%s\"\n", + scn->scenario[scn->current_scenario].name); + } + } + + if (scn->scenario[scn->current_scenario].post_sequence_file != NULL) { + ret = read_sequence_file(scn, POST_SEQ); + if (ret < 0) { + scn_stdout("Warning: PostSequence file defined but" + " missing in scenario \"%s\"\n", + scn->scenario[scn->current_scenario].name); + } + } + + } + return 0; +} + +/* free all resorces */ +static void free_scn(struct snd_scenario *scn) +{ + /* TODO: valgrind to make sure. */ + int i; + + if (scn == NULL) + return; + + if (scn->control) { + if (scn->control->value) + free(scn->control->value); + free(scn->control); + } + + if (scn->list) + free(scn->list); + if (scn->card_name) + free(scn->card_name); + if (scn->pre_seq_list) + free(scn->pre_seq_list); + if (scn->post_seq_list) + free(scn->post_seq_list); + + if (scn->scenario) { + for (i = 0; i < scn->num_scenarios; i++) { + struct scenario_info *info = &scn->scenario[i]; + + if (info->name) + free(info->name); + if (info->file) + free(info->file); + if (info->pre_sequence_file) + free(info->pre_sequence_file); + if (info->post_sequence_file) + free(info->post_sequence_file); + } + free(scn->scenario); + } + free(scn); +} + +/* + * Init sound card scenario db. + */ +struct snd_scenario *snd_scenario_open(const char *card_name) +{ + struct snd_scenario *scn; + int err; + + /* TODO: locking and + * check if card_name scn is already loaded, + * if so reuse to conserve ram. */ + + scn = malloc(sizeof(struct snd_scenario)); + if (scn == NULL) + return NULL; + bzero(scn, sizeof(struct snd_scenario)); + scn->card_name = strdup(card_name); + if (scn->card_name == NULL) { + free(scn); + return NULL; + } + + /* get info about sound card */ + err = parse_card_controls(scn); + if (err < 0) { + free_scn(scn); + return NULL; + } + + /* get info on scenarios and verify against card */ + err = import_scenario_files(scn); + if (err < 0) { + free_scn(scn); + return NULL; + } + + return scn; +} + +/* + * Reload and reparse scenario db. + */ +int snd_scenario_reload(struct snd_scenario *scn) +{ + free_scn(scn); + + scn->num_kcontrols = parse_card_controls(scn); + if (scn->num_kcontrols <= 0) { + free_scn(scn); + return -EINVAL; + } + + scn->num_scenarios = import_scenario_files(scn); + if (scn->num_scenarios <= 0) { + return -EINVAL; + } + + return 0; +} + +void snd_scenario_close(struct snd_scenario *scn) +{ + free_scn(scn); +} + +static int set_control(snd_ctl_t *handle, snd_ctl_elem_id_t *id, + struct snd_scenario *scn) +{ + struct control_settings *setting; + int ret, count, i, idnum; + snd_ctl_elem_info_t *info; + snd_ctl_elem_type_t type; + snd_ctl_elem_value_t *control; + + snd_ctl_elem_info_alloca(&info); + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_info_set_id(info, id); + ret = snd_ctl_elem_info(handle, info); + if (ret < 0) { + scn_error("%s: failed to get ctl info\n", __func__); + return ret; + } + + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_read(handle, control); + + idnum = snd_ctl_elem_id_get_numid(id); + for (i = 0; i < scn->num_kcontrols; i++) { + setting = &scn->control[i]; + if (setting->id == idnum) + goto set_val; + } + scn_error("%s: failed to find control %d\n", __func__, idnum); + return 0; + +set_val: + type = snd_ctl_elem_info_get_type(info); + count = snd_ctl_elem_info_get_count(info); + if (count == 0) + return 0; + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_boolean(control, i, + get_value(scn, setting, i)); + break; + case SND_CTL_ELEM_TYPE_INTEGER: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_integer(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_integer64(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_enumerated(control, i, + get_value(scn, setting, i)); + + break; + case SND_CTL_ELEM_TYPE_BYTES: + for (i = 0; i < count; i++) + snd_ctl_elem_value_set_byte(control, i, + get_value(scn, setting, i)); + break; + default: + break; + } + + ret = snd_ctl_elem_write(handle, control); + if (ret < 0) { + scn_error("%s: control %s failed: %s\n", __func__, + setting->name, snd_strerror(ret)); + scn_error("%s: count %d type: %d\n", __func__, + count, type); + for (i = 0; i < count; i++) + fprintf(stderr, "%d ", get_value(scn, setting, i)); + return ret; + } + return 0; +} + +static void exec_sequence(struct sequence_element *seq, struct snd_scenario + *scn, snd_ctl_elem_list_t *list, snd_ctl_t *handle) +{ + int count = snd_ctl_elem_list_get_count(list); + while (seq) { + if (seq->sleep) + usleep(seq->sleep); + else { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + int ret, i, numid; + /* Where is id lookup from numid if you need it? */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_list_get_id(list, i, id); + numid = snd_ctl_elem_id_get_numid(id); + if (numid == seq->control->id) { + ret = set_control(handle, id, scn); + if (ret < 0) { + scn_error("%s: failed to set control %s\n", + __func__, scn->card_name); + } + break; + } + } + } + seq = seq->next; + } +} + +int snd_scenario_set_scn(struct snd_scenario *scn, const char *name) +{ + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + snd_ctl_t *handle; + int ret, count, i; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* find scenario name */ + for (i = 0; i < scn->num_scenarios; i++) { + if (!strcmp(scn->scenario[i].name, name)) + goto found; + } + scn_error("%s: scenario %s not found\n", __func__, name); + return -EINVAL; + +found: + /* scenario found - now open card */ + scn->current_scenario = i; + ret = snd_ctl_open(&handle, scn->card_name, 0); + if (ret) { + scn_error("%s: control %s open error: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + scn->card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + count = snd_ctl_elem_list_get_count(list); + if (count < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, count) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* If we have a sequence list that should be executed before the new + * scenario is set do it now */ + if (scn->pre_seq_list) + exec_sequence(scn->pre_seq_list, scn, list, handle); + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = set_control(handle, id, scn); + if (ret < 0) { + scn_error("%s: failed to set control %s\n", __func__, + scn->card_name); + } + } + + /* If we have a sequence list that should be executed after the new + * scenario is set do it now */ + if (scn->post_seq_list) + exec_sequence(scn->post_seq_list, scn, list, handle); + +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +int snd_scenario_dump(const char *card_name) +{ + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_elem_list_t *list; + int ret, i, count; + + snd_ctl_card_info_alloca(&info); + snd_ctl_elem_list_alloca(&list); + + /* open and load snd card */ + ret = snd_ctl_open(&handle, card_name, SND_CTL_READONLY); + if (ret < 0) { + scn_error("%s: control %s open retor: %s\n", __func__, card_name, + snd_strerror(ret)); + return ret; + } + + ret = snd_ctl_card_info(handle, info); + if (ret < 0) { + scn_error("%s :control %s local retor: %s\n", __func__, + card_name, snd_strerror(ret)); + goto close; + } + + ret = snd_ctl_elem_list(handle, list); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto close; + } + + count = snd_ctl_elem_list_get_count(list); + if (count < 0) { + ret = 0; + goto close; + } + + snd_ctl_elem_list_set_offset(list, 0); + if (snd_ctl_elem_list_alloc_space(list, count) < 0) { + scn_error("%s: not enough memory...\n", __func__); + ret = -ENOMEM; + goto close; + } + if ((ret = snd_ctl_elem_list(handle, list)) < 0) { + scn_error("%s: cannot determine controls: %s\n", __func__, + snd_strerror(ret)); + goto free; + } + + /* iterate through each kcontrol and add to db */ + for (i = 0; i < count; ++i) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_list_get_id(list, i, id); + + ret = dump_control(handle, id); + if (ret < 0) { + scn_error("%s: cannot determine controls: %s\n", + __func__, snd_strerror(ret)); + goto free; + } + } +free: + snd_ctl_elem_list_free_space(list); +close: + snd_ctl_close(handle); + return ret; +} + +const char *snd_scenario_get_scn(struct snd_scenario *scn) +{ + if (scn->current_scenario > 0 && scn->current_scenario < MAX_SCN) + return scn->scenario[scn->current_scenario].name; + else + return NULL; +} + +int snd_scenario_set_qos(struct snd_scenario *scn, int qos) +{ + /* TODO: change QoS kcontrols */ + scn->scenario[scn->current_scenario].qos = qos; + return 0; +} + +int snd_scenario_get_qos(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].qos; +} + +int snd_scenario_get_master_playback_volume(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].playback_volume_id; +} + +int snd_scenario_get_master_playback_switch(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].playback_switch_id; +} + +int snd_scenario_get_master_capture_volume(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].capture_volume_id; +} + +int snd_scenario_get_master_capture_switch(struct snd_scenario *scn) +{ + return scn->scenario[scn->current_scenario].capture_switch_id; +} + +int snd_scenario_list(struct snd_scenario *scn, const char **list[]) +{ + *list = scn->list; + return scn->num_scenarios; +} -- 1.6.4.3 regards Stefan Schmidt

1 0

[alsa-devel] Some bugs with Auzentech Prelude
by Laurent Léonard 30 Sep '09

30 Sep '09

Hi, I just bought an Auzentech Prelude and compiled a 2.6.31.1 kernel to test it. Analog and digital outputs are working but I experience some problems: - There are some lags when I play long tracks on Amarok. - Sometimes the track suddenly play slower for a time and go back, or not, and I have to pause and play to fix it. - Sometimes there is noise (only) for a time when I play a track, then it go back after a while. - I was unable to play an audio file correctly with VLC, the sound is very saccaded. - No sound at all with Flash and Iceweasel. Are they known bugs ? Are there plans to fix them ? Thank you, -- Laurent Léonard

1 0

[alsa-devel] [PATCH 0/3] Usage of AM_SILENT_RULES and small cleanup
by Stefan Schmidt 30 Sep '09

30 Sep '09

Hello. While working with alsa-lib to get scenarios in shape and submitted later I added support for the new automake feature AM_SILENT_RULES and did some small cleanups. [PATCH 1/3] cvscompile: Remove in favour of gitcompile. I see no reason to keep this around when CVS is no longer used and gitcompile does replace it. Something I miss here? [PATCH 2/3] configure.in: Add m4 check for new AM_SILENT_RULES The kbuild like output makes it a lot easier to follow the compile and linking process, at least in my opinion. It is available with automake 1.11 and we use a m4 macro here to only use it when it is available. [PATCH 3/3] control: Remove unused variable. Found while really seeing what is going on thanks to AM_SILENT_RULES: ;) There are also some unused paramter and a function around. I don't know how you guys prefer to handle this in the ALSA project so I did not prepare patches for this yet. Let me know if you want some for it. pcm_route.c: In function ‘snd_pcm_route_convert1_many’: pcm_route.c:350: warning: ‘src’ may be used uninitialized in this function pcm_rate_linear.c: In function ‘get_supported_rates’: pcm_rate_linear.c:408: warning: unused parameter ‘rate’ pcm_rate_linear.c: In function ‘linear_dump’: pcm_rate_linear.c:416: warning: unused parameter ‘rate’ pcm_rate_linear.c: In function ‘_snd_pcm_rate_linear_open’: pcm_rate_linear.c:435: warning: unused parameter ‘version’ alisp.c:237: warning: ‘incref_tree_explicit’ defined but not used regards Stefan Schmidt

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[alsa-devel] [PATCH 1/2] pcm_rate_linear: Annotate unused function parameter to avoid compiler warnings.
by Stefan Schmidt 30 Sep '09

30 Sep '09

From: Stefan Schmidt <stefan(a)slimlogic.co.uk> Signed-off-by: Stefan Schmidt <stefan(a)slimlogic.co.uk> --- src/pcm/pcm_rate_linear.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_rate_linear.c b/src/pcm/pcm_rate_linear.c index 8b2d2d0..7481b38 100644 --- a/src/pcm/pcm_rate_linear.c +++ b/src/pcm/pcm_rate_linear.c @@ -405,15 +405,15 @@ static void linear_close(void *obj) free(obj); } -static int get_supported_rates(void *rate, unsigned int *rate_min, - unsigned int *rate_max) +static int get_supported_rates(ATTRIBUTE_UNUSED void *rate, + unsigned int *rate_min, unsigned int *rate_max) { *rate_min = SND_PCM_PLUGIN_RATE_MIN; *rate_max = SND_PCM_PLUGIN_RATE_MAX; return 0; } -static void linear_dump(void *rate, snd_output_t *out) +static void linear_dump(ATTRIBUTE_UNUSED void *rate, snd_output_t *out) { snd_output_printf(out, "Converter: linear-interpolation\n"); } @@ -432,7 +432,8 @@ static const snd_pcm_rate_ops_t linear_ops = { .dump = linear_dump, }; -int SND_PCM_RATE_PLUGIN_ENTRY(linear) (unsigned int version, void **objp, snd_pcm_rate_ops_t *ops) +int SND_PCM_RATE_PLUGIN_ENTRY(linear) (ATTRIBUTE_UNUSED unsigned int version, + void **objp, snd_pcm_rate_ops_t *ops) { struct rate_linear *rate; -- 1.6.4.3

1 1

[alsa-devel] [PATCH 0/1] ASoC: DAPM: Fix for line widget handling
by Peter Ujfalusi 30 Sep '09

30 Sep '09

Hello, The following patch fixes the handling of the line widget, which can be input or output. Additional check is added to is_connected_output_ep and is_connected_input_ep functions when the line widget is checked to handle only the appropriate line widget. Without the patch these two functions would pick up either line widgets, which confused DAPM. Note: checkpach complains for the long lines, but I feel that it is appropriate here to use the long lines. --- Peter Ujfalusi (1): ASoC: Fix SND_SOC_DAPM_LINE handling sound/soc/soc-dapm.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-)

2 2

[alsa-devel] [PATCH 1/1] Echoaudio: fix broken Mia driver
by Giuliano Pochini 30 Sep '09

30 Sep '09

This change: http://git.alsa-project.org/?p=alsa-kernel.git;a=blobdiff;f=sound/pci/echoa… broke the Mia driver because it removed the line-out control interface from all vmixer cards. But the Mia has that (undocumented!!) control and after the change it is impossible to raise the volume. I tested the driver with my IndigoIOx that was supposed to have the same set of controls... sorry. I hope it's not too late to merge this patch in linux-2.6.32. ---------- Re-enable the line-out control for the Mia card. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -u alsa-driver-1.0.21/alsa-kernel/pci/echoaudio__orig/echoaudio.c alsa-driver-1.0.21/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.21/alsa-kernel/pci/echoaudio__orig/echoaudio.c 2009-08-28 13:41:39.000000000 +0200 +++ alsa-driver-1.0.21/alsa-kernel/pci/echoaudio/echoaudio.c 2009-09-29 21:55:48.000000000 +0200 @@ -950,7 +950,7 @@ Control interface ******************************************************************************/ -#ifndef ECHOCARD_HAS_VMIXER +#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN) /******************* PCM output volume *******************/ static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol, @@ -1003,6 +1003,19 @@ return changed; } +#ifdef ECHOCARD_HAS_LINE_OUT_GAIN +/* On the Mia this one controls the line-out volume */ +static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = { + .name = "Line Playback Volume", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = snd_echo_output_gain_info, + .get = snd_echo_output_gain_get, + .put = snd_echo_output_gain_put, + .tlv = {.p = db_scale_output_gain}, +}; +#else static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1012,9 +1025,10 @@ .put = snd_echo_output_gain_put, .tlv = {.p = db_scale_output_gain}, }; - #endif +#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */ + #ifdef ECHOCARD_HAS_INPUT_GAIN @@ -2030,10 +2044,14 @@ snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip); if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0) goto ctl_error; -#else - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0) +#ifdef ECHOCARD_HAS_LINE_OUT_GAIN + if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0) goto ctl_error; #endif +#else /* ECHOCARD_HAS_VMIXER */ + if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0) + goto ctl_error; +#endif /* ECHOCARD_HAS_VMIXER */ #ifdef ECHOCARD_HAS_INPUT_GAIN if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0) diff -u alsa-driver-1.0.21/alsa-kernel/pci/echoaudio__orig/mia.c alsa-driver-1.0.21/alsa-kernel/pci/echoaudio/mia.c --- alsa-driver-1.0.21/alsa-kernel/pci/echoaudio__orig/mia.c 2009-08-28 13:41:39.000000000 +0200 +++ alsa-driver-1.0.21/alsa-kernel/pci/echoaudio/mia.c 2009-09-29 21:55:48.000000000 +0200 @@ -29,6 +29,7 @@ #define ECHOCARD_HAS_ADAT FALSE #define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 #define ECHOCARD_HAS_MIDI +#define ECHOCARD_HAS_LINE_OUT_GAIN /* Pipe indexes */ #define PX_ANALOG_OUT 0 /* 8 */ -- Giuliano.

2 2

[alsa-devel] alsa duplex test application
by 罗富冲 30 Sep '09

30 Sep '09

Hi all, I want to write a sound card Duplex test Application(playing what we are recording) refer to alsa-utils, and i got Segmentation fault. Accessoriesis my application for test,can anyone help me to find out what is the problem, or show me a application which can work. Sorry for my poor english, and waiting for your reply.

2 1

[alsa-devel] My Codec: IDT 92HD75B3X5 issues
by Munzir Taha (منذر طه) 29 Sep '09

29 Sep '09

Hi sirs, These are the issues I have with my Laptop HP Pavilion dv6 1225ee regarding audio support. 1. No sense for headphone jack (jack_present?), when headphone is plugged (present_int_hp?) 2. There headphone mic jack is not working. The one next to the webcam is always recording even when I change the "Mic Jack" control from "Mic In" to "Line In" 3. The mute/unmute LED doesn't change color to RED when its muted like in Windows. Touching that LED shows this in /var/log/messages: Sep 28 06:53:47 dv6 kernel: [ 9197.986789] atkbd.c: Unknown key released (translated set 2, code 0x94 on isa0060/serio0). Sep 28 06:53:47 dv6 kernel: [ 9197.986791] atkbd.c: Use 'setkeycodes e014 <keycode>' to make it known. Sep 28 06:53:47 dv6 kernel: [ 9198.353568] atkbd.c: Unknown key pressed (translated set 2, code 0x94 on isa0060/serio0). Sep 28 06:53:47 dv6 kernel: [ 9198.353572] atkbd.c: Use 'setkeycodes e014 <keycode>' to make it known. Sep 28 06:53:47 dv6 kernel: [ 9198.359205] atkbd.c: Unknown key released (translated set 2, code 0x94 on isa0060/serio0). Sep 28 06:53:47 dv6 kernel: [ 9198.359208] atkbd.c: Use 'setkeycodes e014 <keycode>' to make it known. alsa info is at: http://www.alsa-project.org/db/?f=8f8518c634bc6880a1633e025fe7a7f975f3ed28 I also played with different models like hp-m4 HP mini 1000 hp-dv5 HP dv series hp-hdx HP HDX series hp-dv4-1222nr HP dv4-1222nr (with LED support) but nothing helps. -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

2 6

[alsa-devel] ASoC: SND_SOC_DAPM_LINE behavior
by Peter Ujfalusi 29 Sep '09

29 Sep '09

Hello, as I'm writing a new codec driver I have noticed the following (using 2.6.31 kernel). The codec itself is really simple, it is only capable of playback and it can bypass audio from line-in to the output. The DAPM routing is something like this in the codec driver: |DAC|------------->| | |Playback Mixer|->|OUTPUT| |INPUT|->|Bypass|->| | |SWITCH| Now if in the machine driver I create the following DAPM widget: SND_SOC_DAPM_LINE("Line In", NULL), and than connect this DAPM_LINE to the codec's INPUT (LINE-IN): {"LINE-IN", NULL, "Line In"}, Than the codec bias level would be always in ON state, regardless of the state of the Bypass Switch (it is off by default). I can use the snd_soc_dapm_disable_pin(codec, "Line In") in the machine driver to disable the pin, which moves the codec to STANDBY state. Now, if I change the SND_SOC_DAPM_LINE to SND_SOC_DAPM_MIC("Line In", NULL), in the machine driver, than the codec bias would be STANDBY as long as the Bypass switches are off, if they changed to on, than the codec bias goes to ON, as expected. Is this expected behavior? -- Péter

2 2

[alsa-devel] hda-intel: low volume with nvidia mcp7a hdmi
by Halim Sahin 29 Sep '09

29 Sep '09

Hello, The volume of the mentioned card is really too low. used kernel: 2.6.31-rc6 from opensuse factory. lspci reports: 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) Subsystem: nVidia Corporation Device cb84 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Ste Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at fad78000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3ho Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Output from /proc/asound/cards: 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfad78000 irq 21 I didn't give any cmdline params to snd-ihda-intel. Any ideas how to get more power out of this card? regards halim

2 2

[alsa-devel] [PATCH] ASoC: Consider the codec->active also when changing bias level
by Peter Ujfalusi 29 Sep '09

29 Sep '09

Take into account the codec->active (playback/capture activity) when changing the bias level. Fixes the following scenario: if during active playback/capture the DAPM routes has been modified in a way, that there will be no complete route (for example muting the output) DAPM would set the codec into BIAS_STANDBY, which could disable the clocks from the codec (codec is master), which breaks the playback with error since the clocks are missing from the serial bus and no data is shifted out/in. With this patch during active use (playback/capture) the DAPM will not set the bias to STANDBY, thus keeping the playback/capture running. When no playback/capture is running, the DAPM will handle the complete paths in a same way it used to: setting the bias to ON/STANDBY based on complete paths. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9babda5..567be82 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -977,7 +977,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; power = w->power_check(w); - if (power) + if (power || codec->active) sys_power = 1; if (w->power == power) -- 1.6.5.rc1

2 3

[alsa-devel] Alsa not working when linked to a library
by Fortini Matteo 29 Sep '09

29 Sep '09

Hi, I'm experiencing a very strange behavior. I'm using alsa 1.0.18 on a powerpc platform. If I run the following simple code from the main of a .c file, it works perfectly, but if I put into a shared .so library (libaudio.so), then I link the main with -laudio -lasound, then the hwparams structure is not filled with anything, while the alsa functions don't return any error. Do you have any clue why this could happen? TIA ############################################################ int play_sample(char *pcm_name, char *samplename) { int res; snd_pcm_t *pcm_handle; snd_pcm_hw_params_t *hwparams; const char *name= (pcm_name != NULL) ? pcm_name : "speaker"; if ( (res = snd_pcm_open(&pcm_handle, name, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { fprintf(stderr, "Error opening PCM device %s\n", pcm_name); return 1; } else { printf ("snd_pcm_open returned %d\n", res); } snd_pcm_hw_params_malloc(&hwparams); if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) { fprintf(stderr, "Can not configure this PCM device.\n"); return 1; snd_pcm_close(pcm_handle); } int main (void) { play_sample (NULL, "test_sound.wav); return 0; } ############################################################

2 2

Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 29 Sep '09

29 Sep '09

Peter, I'm getting a XUNDFLSTAT (indicates a real underflow condition) and XEMPTYEOF (Transmit buffer empty at end of frame) in MCBSPLP_IRQSTATUS_REG. Does this tell you anything? Rick

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
by Miguel de Barros 29 Sep '09

29 Sep '09

Hello, I would like to submit this patch, it will fix the bug I've submited : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614 Thanks, Miguel de Barros

2 3

[alsa-devel] [PATCH] snd-usb-us122l: added support for US-144
by Eric Luiggi 29 Sep '09

29 Sep '09

Hi Tobias, I'd like to get my US-144 interface working on jaunty and am willing to help out with testing for this US-144 solution. Please send patch instructions. Thanks, Eric

2 1

Re: [alsa-devel] [PATCH] sscape: convert to firmware loader framework
by krzysztof.h1＠poczta.fm 29 Sep '09

29 Sep '09

Please drop this patch. It does not compile because it includes removed sscape_ioctl.h. I will send updated patch. Best regards, Krzysztof "Krzysztof Helt" <krzysztof.h1(a)poczta.fm> pisze: > The conversion solves the problem that firmware size was set to 64KB > while non PnP cards have 128KB firmware files. > > An additional post firmware initialization code has been moved > from the OSS driver. > > --- > Now, the MIDI is audible but not correct. It is not played timely. > The sscape_ctl alsa-tool is not needed any more. > > > Documentation/sound/alsa/ALSA-Configuration.txt | 8 +- > include/sound/sscape_ioctl.h | 21 -- > sound/isa/Kconfig | 8 +- > sound/isa/sscape.c | 305 > +++++++---------------- > 4 files changed, 101 insertions(+), 241 deletions(-) > delete mode 100644 include/sound/sscape_ioctl.h > > diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt > b/Documentation/sound/alsa/ALSA-Configuration.txt > index 1c8eb45..cf98525 100644 > --- a/Documentation/sound/alsa/ALSA-Configuration.txt > +++ b/Documentation/sound/alsa/ALSA-Configuration.txt > @@ -1631,7 +1631,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' > prefix. This was removed. > Module snd-sscape > ----------------- > > - Module for ENSONIQ SoundScape PnP cards. > + Module for ENSONIQ SoundScape cards. > > port - Port # (PnP setup) > wss_port - WSS Port # (PnP setup) > @@ -1640,9 +1640,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' > prefix. This was removed. > dma - DMA # (PnP setup) > dma2 - 2nd DMA # (PnP setup, -1 to disable) > > - This module supports multiple cards. ISA PnP must be enabled. > - You need sscape_ctl tool in alsa-tools package for loading > - the microcode. > + This module supports multiple cards. > + > + The driver requires the firmware loader support on kernel. > > Module snd-sun-amd7930 (on sparc only) > -------------------------------------- > diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h > deleted file mode 100644 > index 0d88859..0000000 > --- a/include/sound/sscape_ioctl.h > +++ /dev/null > @@ -1,21 +0,0 @@ > -#ifndef SSCAPE_IOCTL_H > -#define SSCAPE_IOCTL_H > - > - > -struct sscape_bootblock > -{ > - unsigned char code[256]; > - unsigned version; > -}; > - > -#define SSCAPE_MICROCODE_SIZE 65536 > - > -struct sscape_microcode > -{ > - unsigned char __user *code; > -}; > - > -#define SND_SSCAPE_LOAD_BOOTB _IOWR('P', 100, struct sscape_bootblock) > -#define SND_SSCAPE_LOAD_MCODE _IOW ('P', 101, struct sscape_microcode) > - > -#endif > diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig > index b90fc16..02fe81c 100644 > --- a/sound/isa/Kconfig > +++ b/sound/isa/Kconfig > @@ -372,9 +372,9 @@ config SND_SGALAXY > > config SND_SSCAPE > tristate "Ensoniq SoundScape driver" > - select SND_HWDEP > select SND_MPU401_UART > select SND_WSS_LIB > + select FW_LOADER > help > Say Y here to include support for Ensoniq SoundScape > and Ensoniq OEM soundcards. > @@ -382,7 +382,11 @@ config SND_SSCAPE > The PCM audio is supported on SoundScape Classic, Elite, PnP > and VIVO cards. The supported OEM cards are SPEA Media FX and > Reveal SC-600. > - The MIDI support is very experimental. > + The MIDI support is very experimental and requires binary > + firmware files called "scope.cod" and "sndscape.co?" where the > + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found > + in DOS or Windows driver packages. One has to put the firmware > + files into the /lib/firmware directory. > > To compile this driver as a module, choose M here: the module > will be called snd-sscape. > diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c > index b11c35f..08389a2 100644 > --- a/sound/isa/sscape.c > +++ b/sound/isa/sscape.c > @@ -1,5 +1,5 @@ > /* > - * Low-level ALSA driver for the ENSONIQ SoundScape PnP > + * Low-level ALSA driver for the ENSONIQ SoundScape > * Copyright (c) by Chris Rankin > * > * This driver was written in part using information obtained from > @@ -25,12 +25,12 @@ > #include <linux/err.h> > #include <linux/isa.h> > #include <linux/delay.h> > +#include <linux/firmware.h> > #include <linux/pnp.h> > #include <linux/spinlock.h> > #include <linux/moduleparam.h> > #include <asm/dma.h> > #include <sound/core.h> > -#include <sound/hwdep.h> > #include <sound/wss.h> > #include <sound/mpu401.h> > #include <sound/initval.h> > @@ -39,7 +39,7 @@ > > > MODULE_AUTHOR("Chris Rankin"); > -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); > +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); > MODULE_LICENSE("GPL"); > > static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; > @@ -142,14 +142,12 @@ struct soundscape { > struct resource *wss_res; > struct snd_wss *chip; > struct snd_mpu401 *mpu; > - struct snd_hwdep *hw; > > /* > * The MIDI device won't work until we've loaded > * its firmware via a hardware-dependent device IOCTL > */ > spinlock_t fwlock; > - int hw_in_use; > unsigned long midi_usage; > unsigned char midi_vol; > }; > @@ -167,12 +165,6 @@ static inline struct soundscape > *get_mpu401_soundscape(struct snd_mpu401 * mpu) > return (struct soundscape *) (mpu->private_data); > } > > -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * > hw) > -{ > - return (struct soundscape *) (hw->private_data); > -} > - > - > /* > * Allocates some kernel memory that we can use for DMA. > * I think this means that the memory has to map to > @@ -438,14 +430,14 @@ static int host_startup_ack(struct soundscape *s, > unsigned timeout) > * Upload a byte-stream into the SoundScape using DMA channel A. > */ > static int upload_dma_data(struct soundscape *s, > - const unsigned char __user *data, > + const unsigned char *data, > size_t size) > { > unsigned long flags; > struct snd_dma_buffer dma; > int ret; > > - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) > + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) > return -ENOMEM; > > spin_lock_irqsave(&s->lock, flags); > @@ -468,35 +460,17 @@ static int upload_dma_data(struct soundscape *s, > sscape_write_unsafe(s->io_base, GA_HMCTL_REG, > sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); > > /* > - * Upload the user's data (firmware?) to the SoundScape > + * Upload the firmware to the SoundScape > * board through the DMA channel ... > */ > while (size != 0) { > unsigned long len; > > - /* > - * Apparently, copying to/from userspace can sleep. > - * We are therefore forbidden from holding any > - * spinlocks while we copy ... > - */ > - spin_unlock_irqrestore(&s->lock, flags); > - > - /* > - * Remember that the data that we want to DMA > - * comes from USERSPACE. We have already verified > - * the userspace pointer ... > - */ > len = min(size, dma.bytes); > - len -= __copy_from_user(dma.area, data, len); > + memcpy(dma.area, data, len); > data += len; > size -= len; > > - /* > - * Grab that spinlock again, now that we've > - * finished copying! > - */ > - spin_lock_irqsave(&s->lock, flags); > - > snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); > sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); > if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { > @@ -512,6 +486,7 @@ static int upload_dma_data(struct soundscape *s, > } /* while */ > > set_host_mode_unsafe(s->io_base); > + outb(0x0, s->io_base); > > /* > * Boot the board ... (I think) > @@ -537,7 +512,7 @@ _release_dma: > /* > * NOTE!!! We are NOT holding any spinlocks at this point !!! > */ > - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); > + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); > free_dmabuf(&dma); > > return ret; > @@ -547,162 +522,70 @@ _release_dma: > * Upload the bootblock(?) into the SoundScape. The only > * purpose of this block of code seems to be to tell > * us which version of the microcode we should be using. > - * > - * NOTE: The boot-block data resides in USER-SPACE!!! > - * However, we have already verified its memory > - * addresses by the time we get here. > */ > -static int sscape_upload_bootblock(struct soundscape *sscape, struct > sscape_bootblock __user *bb) > +static int sscape_upload_bootblock(struct snd_card *card) > { > + struct soundscape *sscape = get_card_soundscape(card); > unsigned long flags; > + const struct firmware *init_fw = NULL; > int data = 0; > int ret; > > - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); > - > - spin_lock_irqsave(&sscape->lock, flags); > - if (ret == 0) { > - data = host_read_ctrl_unsafe(sscape->io_base, 100); > - } > - set_midi_mode_unsafe(sscape->io_base); > - spin_unlock_irqrestore(&sscape->lock, flags); > - > - if (ret == 0) { > - if (data < 0) { > - snd_printk(KERN_ERR "sscape: timeout reading firmware versionn"); > - ret = -EAGAIN; > - } > - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { > - ret = -EFAULT; > - } > + ret = request_firmware(&init_fw, "scope.cod", card->dev); > + if (ret < 0) { > + snd_printk(KERN_ERR "Error loading scope.cod"); > + return ret; > } > + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); > > - return ret; > -} > + release_firmware(init_fw); > > -/* > - * Upload the microcode into the SoundScape. The > - * microcode is 64K of data, and if we try to copy > - * it into a local variable then we will SMASH THE > - * KERNEL'S STACK! We therefore leave it in USER > - * SPACE, and save ourselves from copying it at all. > - */ > -static int sscape_upload_microcode(struct soundscape *sscape, > - const struct sscape_microcode __user > *mc) > -{ > - unsigned long flags; > - char __user *code; > - int err; > - > - /* > - * We are going to have to copy this data into a special > - * DMA-able buffer before we can upload it. We shall therefore > - * just check that the data pointer is valid for now. > - * > - * NOTE: This buffer is 64K long! That's WAY too big to > - * copy into a stack-temporary anyway. > - */ > - if ( get_user(code, &mc->code) || > - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) > - return -EFAULT; > + spin_lock_irqsave(&sscape->lock, flags); > + if (ret == 0) > + data = host_read_ctrl_unsafe(sscape->io_base, 100); > > - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) > { > - snd_printk(KERN_INFO "sscape: MIDI firmware loadedn"); > - } > + if (data & 0x10) > + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); > > - spin_lock_irqsave(&sscape->lock, flags); > - set_midi_mode_unsafe(sscape->io_base); > spin_unlock_irqrestore(&sscape->lock, flags); > > - initialise_mpu401(sscape->mpu); > + data &= 0xf; > + if (ret == 0 && data > 7) { > + snd_printk(KERN_ERR > + "sscape: timeout reading firmware versionn"); > + ret = -EAGAIN; > + } > > - return err; > + return (ret == 0) ? data : ret; > } > > /* > - * Hardware-specific device functions, to implement special > - * IOCTLs for the SoundScape card. This is how we upload > - * the microcode into the card, for example, and so we > - * must ensure that no two processes can open this device > - * simultaneously, and that we can't open it at all if > - * someone is using the MIDI device. > + * Upload the microcode into the SoundScape. > */ > -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) > +static int sscape_upload_microcode(struct snd_card *card, int version) > { > - register struct soundscape *sscape = get_hwdep_soundscape(hw); > - unsigned long flags; > + struct soundscape *sscape = get_card_soundscape(card); > + const struct firmware *init_fw = NULL; > + char name[14] = "sndscape.co0"; > int err; > > - spin_lock_irqsave(&sscape->fwlock, flags); > + name[11] = '0' + version; > > - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { > - err = -EBUSY; > - } else { > - sscape->hw_in_use = 1; > - err = 0; > + err = request_firmware(&init_fw, name, card->dev); > + if (err < 0) { > + snd_printk(KERN_ERR "Error loading sndscape.co0"); > + return err; > } > + err = upload_dma_data(sscape, init_fw->data, init_fw->size); > + if (err == 0) > + snd_printk(KERN_INFO "MIDI firmware loaded %d KBsn", > + init_fw->size >> 10); > > - spin_unlock_irqrestore(&sscape->fwlock, flags); > - return err; > -} > - > -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) > -{ > - register struct soundscape *sscape = get_hwdep_soundscape(hw); > - unsigned long flags; > - > - spin_lock_irqsave(&sscape->fwlock, flags); > - sscape->hw_in_use = 0; > - spin_unlock_irqrestore(&sscape->fwlock, flags); > - return 0; > -} > - > -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, > - unsigned int cmd, unsigned long arg) > -{ > - struct soundscape *sscape = get_hwdep_soundscape(hw); > - int err = -EBUSY; > - > - switch (cmd) { > - case SND_SSCAPE_LOAD_BOOTB: > - { > - register struct sscape_bootblock __user *bb = (struct sscape_bootblock > __user *) arg; > - > - /* > - * We are going to have to copy this data into a special > - * DMA-able buffer before we can upload it. We shall therefore > - * just check that the data pointer is valid for now ... > - */ > - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) > - return -EFAULT; > - > - /* > - * Now check that we can write the firmware version number too... > - */ > - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) > - return -EFAULT; > - > - err = sscape_upload_bootblock(sscape, bb); > - } > - break; > - > - case SND_SSCAPE_LOAD_MCODE: > - { > - register const struct sscape_microcode __user *mc = (const struct > sscape_microcode __user *) arg; > - > - err = sscape_upload_microcode(sscape, mc); > - } > - break; > - > - default: > - err = -EINVAL; > - break; > - } /* switch */ > + release_firmware(init_fw); > > return err; > } > > - > /* > * Mixer control for the SoundScape's MIDI device. > */ > @@ -920,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 * mpu) > > spin_lock_irqsave(&sscape->fwlock, flags); > > - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { > + if (sscape->midi_usage == ULONG_MAX) { > err = -EBUSY; > } else { > ++(sscape->midi_usage); > @@ -1053,13 +936,6 @@ static int __devinit create_ad1845(struct snd_card > *card, unsigned port, > } > } > > - strcpy(card->driver, "SoundScape"); > - strcpy(card->shortname, pcm->name); > - snprintf(card->longname, sizeof(card->longname), > - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %dn", > - pcm->name, chip->port, chip->irq, > - chip->dma1, chip->dma2); > - > sscape->chip = chip; > } > > @@ -1162,29 +1038,6 @@ static int __devinit create_sscape(int dev, struct > snd_card *card) > return -ENXIO; > } > > - if (sscape->type != SSCAPE_VIVO) { > - /* > - * Now create the hardware-specific device so that we can > - * load the microcode into the on-board processor. > - * We cannot use the MPU-401 MIDI system until this firmware > - * has been loaded into the card. > - */ > - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); > - if (err < 0) { > - printk(KERN_ERR "sscape: Failed to create " > - "firmware devicen"); > - goto _release_dma; > - } > - strlcpy(sscape->hw->name, "SoundScape M68K", > - sizeof(sscape->hw->name)); > - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '�'; > - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; > - sscape->hw->ops.open = sscape_hw_open; > - sscape->hw->ops.release = sscape_hw_release; > - sscape->hw->ops.ioctl = sscape_hw_ioctl; > - sscape->hw->private_data = sscape; > - } > - > /* > * Tell the on-board devices where their resources are (I think - > * I can't be sure without a datasheet ... So many magic values!) > @@ -1222,28 +1075,51 @@ static int __devinit create_sscape(int dev, struct > snd_card *card) > wss_port[dev], irq[dev]); > goto _release_dma; > } > + strcpy(card->driver, "SoundScape"); > + strcpy(card->shortname, name); > + snprintf(card->longname, sizeof(card->longname), > + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %dn", > + name, sscape->chip->port, sscape->chip->irq, > + sscape->chip->dma1, sscape->chip->dma2); > + > #define MIDI_DEVNUM 0 > if (sscape->type != SSCAPE_VIVO) { > - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); > - if (err < 0) { > - printk(KERN_ERR "sscape: Failed to create " > - "MPU-401 device at 0x%lxn", > - port[dev]); > - goto _release_dma; > - } > + err = sscape_upload_bootblock(card); > + if (err >= 0) > + err = sscape_upload_microcode(card, err); > + > + if (err == 0) { > + err = create_mpu401(card, MIDI_DEVNUM, port[dev], > + mpu_irq[dev]); > + if (err < 0) { > + printk(KERN_ERR "sscape: Failed to create " > + "MPU-401 device at 0x%lxn", > + port[dev]); > + goto _release_dma; > + } > > - /* > - * Enable the master IRQ ... > - */ > - sscape_write(sscape, GA_INTENA_REG, 0x80); > - > - /* > - * Initialize mixer > - */ > - sscape->midi_vol = 0; > - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); > - host_write_ctrl_unsafe(sscape->io_base, 0, 100); > - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); > + /* > + * Initialize mixer > + */ > + spin_lock_irqsave(&sscape->lock, flags); > + sscape->midi_vol = 0; > + host_write_ctrl_unsafe(sscape->io_base, > + CMD_SET_MIDI_VOL, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + sscape->midi_vol, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + CMD_XXX_MIDI_VOL, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + sscape->midi_vol, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + CMD_SET_EXTMIDI, 100); > + host_write_ctrl_unsafe(sscape->io_base, > + 0, 100); > + host_write_ctrl_unsafe(sscape->io_base, 80, 1); > + > + set_midi_mode_unsafe(sscape->io_base); > + spin_unlock_irqrestore(&sscape->lock, flags); > + } > } > > /* > @@ -1301,11 +1177,12 @@ static int __devinit snd_sscape_probe(struct > device *pdev, unsigned int dev) > sscape->type = SSCAPE; > > dma[dev] &= 0x03; > + snd_card_set_dev(card, pdev); > + > ret = create_sscape(dev, card); > if (ret < 0) > goto _release_card; > > - snd_card_set_dev(card, pdev); > if ((ret = snd_card_register(card)) < 0) { > printk(KERN_ERR "sscape: Failed to register sound cardn"); > goto _release_card; > @@ -1426,12 +1303,12 @@ static int __devinit sscape_pnp_detect(struct > pnp_card_link *pcard, > wss_port[idx] = pnp_port_start(dev, 1); > dma2[idx] = pnp_dma(dev, 1); > } > + snd_card_set_dev(card, &pcard->card->dev); > > ret = create_sscape(idx, card); > if (ret < 0) > goto _release_card; > > - snd_card_set_dev(card, &pcard->card->dev); > if ((ret = snd_card_register(card)) < 0) { > printk(KERN_ERR "sscape: Failed to register sound cardn"); > goto _release_card; > -- > 1.6.0.3 > > ---------------------------------------------------------------------- > Co godzine losujemy 15 000 z ---------------------------------------------------------------------- Praca za granic�? Zobacz oferty! http://link.interia.pl/f2331

2 1

[alsa-devel] [PATCH] sscape: convert to firmware loader framework
by Krzysztof Helt 29 Sep '09

29 Sep '09

The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional post firmware initialization code has been moved from the OSS driver. --- Now, the MIDI is audible but not correct. It is not played timely. The sscape_ctl alsa-tool is not needed any more. Documentation/sound/alsa/ALSA-Configuration.txt | 8 +- include/sound/sscape_ioctl.h | 21 -- sound/isa/Kconfig | 8 +- sound/isa/sscape.c | 305 +++++++---------------- 4 files changed, 101 insertions(+), 241 deletions(-) delete mode 100644 include/sound/sscape_ioctl.h diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1c8eb45..cf98525 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1631,7 +1631,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sscape ----------------- - Module for ENSONIQ SoundScape PnP cards. + Module for ENSONIQ SoundScape cards. port - Port # (PnP setup) wss_port - WSS Port # (PnP setup) @@ -1640,9 +1640,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) - This module supports multiple cards. ISA PnP must be enabled. - You need sscape_ctl tool in alsa-tools package for loading - the microcode. + This module supports multiple cards. + + The driver requires the firmware loader support on kernel. Module snd-sun-amd7930 (on sparc only) -------------------------------------- diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h deleted file mode 100644 index 0d88859..0000000 --- a/include/sound/sscape_ioctl.h +++ /dev/null @@ -1,21 +0,0 @@ -#ifndef SSCAPE_IOCTL_H -#define SSCAPE_IOCTL_H - - -struct sscape_bootblock -{ - unsigned char code[256]; - unsigned version; -}; - -#define SSCAPE_MICROCODE_SIZE 65536 - -struct sscape_microcode -{ - unsigned char __user *code; -}; - -#define SND_SSCAPE_LOAD_BOOTB _IOWR('P', 100, struct sscape_bootblock) -#define SND_SSCAPE_LOAD_MCODE _IOW ('P', 101, struct sscape_microcode) - -#endif diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b90fc16..02fe81c 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -372,9 +372,9 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape driver" - select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB + select FW_LOADER help Say Y here to include support for Ensoniq SoundScape and Ensoniq OEM soundcards. @@ -382,7 +382,11 @@ config SND_SSCAPE The PCM audio is supported on SoundScape Classic, Elite, PnP and VIVO cards. The supported OEM cards are SPEA Media FX and Reveal SC-600. - The MIDI support is very experimental. + The MIDI support is very experimental and requires binary + firmware files called "scope.cod" and "sndscape.co?" where the + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found + in DOS or Windows driver packages. One has to put the firmware + files into the /lib/firmware directory. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b11c35f..08389a2 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1,5 +1,5 @@ /* - * Low-level ALSA driver for the ENSONIQ SoundScape PnP + * Low-level ALSA driver for the ENSONIQ SoundScape * Copyright (c) by Chris Rankin * * This driver was written in part using information obtained from @@ -25,12 +25,12 @@ #include <linux/err.h> #include <linux/isa.h> #include <linux/delay.h> +#include <linux/firmware.h> #include <linux/pnp.h> #include <linux/spinlock.h> #include <linux/moduleparam.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/hwdep.h> #include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> @@ -39,7 +39,7 @@ MODULE_AUTHOR("Chris Rankin"); -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); MODULE_LICENSE("GPL"); static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; @@ -142,14 +142,12 @@ struct soundscape { struct resource *wss_res; struct snd_wss *chip; struct snd_mpu401 *mpu; - struct snd_hwdep *hw; /* * The MIDI device won't work until we've loaded * its firmware via a hardware-dependent device IOCTL */ spinlock_t fwlock; - int hw_in_use; unsigned long midi_usage; unsigned char midi_vol; }; @@ -167,12 +165,6 @@ static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) return (struct soundscape *) (mpu->private_data); } -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw) -{ - return (struct soundscape *) (hw->private_data); -} - - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -438,14 +430,14 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) * Upload a byte-stream into the SoundScape using DMA channel A. */ static int upload_dma_data(struct soundscape *s, - const unsigned char __user *data, + const unsigned char *data, size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; spin_lock_irqsave(&s->lock, flags); @@ -468,35 +460,17 @@ static int upload_dma_data(struct soundscape *s, sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); /* - * Upload the user's data (firmware?) to the SoundScape + * Upload the firmware to the SoundScape * board through the DMA channel ... */ while (size != 0) { unsigned long len; - /* - * Apparently, copying to/from userspace can sleep. - * We are therefore forbidden from holding any - * spinlocks while we copy ... - */ - spin_unlock_irqrestore(&s->lock, flags); - - /* - * Remember that the data that we want to DMA - * comes from USERSPACE. We have already verified - * the userspace pointer ... - */ len = min(size, dma.bytes); - len -= __copy_from_user(dma.area, data, len); + memcpy(dma.area, data, len); data += len; size -= len; - /* - * Grab that spinlock again, now that we've - * finished copying! - */ - spin_lock_irqsave(&s->lock, flags); - snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { @@ -512,6 +486,7 @@ static int upload_dma_data(struct soundscape *s, } /* while */ set_host_mode_unsafe(s->io_base); + outb(0x0, s->io_base); /* * Boot the board ... (I think) @@ -537,7 +512,7 @@ _release_dma: /* * NOTE!!! We are NOT holding any spinlocks at this point !!! */ - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); free_dmabuf(&dma); return ret; @@ -547,162 +522,70 @@ _release_dma: * Upload the bootblock(?) into the SoundScape. The only * purpose of this block of code seems to be to tell * us which version of the microcode we should be using. - * - * NOTE: The boot-block data resides in USER-SPACE!!! - * However, we have already verified its memory - * addresses by the time we get here. */ -static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb) +static int sscape_upload_bootblock(struct snd_card *card) { + struct soundscape *sscape = get_card_soundscape(card); unsigned long flags; + const struct firmware *init_fw = NULL; int data = 0; int ret; - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); - - spin_lock_irqsave(&sscape->lock, flags); - if (ret == 0) { - data = host_read_ctrl_unsafe(sscape->io_base, 100); - } - set_midi_mode_unsafe(sscape->io_base); - spin_unlock_irqrestore(&sscape->lock, flags); - - if (ret == 0) { - if (data < 0) { - snd_printk(KERN_ERR "sscape: timeout reading firmware version\n"); - ret = -EAGAIN; - } - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { - ret = -EFAULT; - } + ret = request_firmware(&init_fw, "scope.cod", card->dev); + if (ret < 0) { + snd_printk(KERN_ERR "Error loading scope.cod"); + return ret; } + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); - return ret; -} + release_firmware(init_fw); -/* - * Upload the microcode into the SoundScape. The - * microcode is 64K of data, and if we try to copy - * it into a local variable then we will SMASH THE - * KERNEL'S STACK! We therefore leave it in USER - * SPACE, and save ourselves from copying it at all. - */ -static int sscape_upload_microcode(struct soundscape *sscape, - const struct sscape_microcode __user *mc) -{ - unsigned long flags; - char __user *code; - int err; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now. - * - * NOTE: This buffer is 64K long! That's WAY too big to - * copy into a stack-temporary anyway. - */ - if ( get_user(code, &mc->code) || - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) - return -EFAULT; + spin_lock_irqsave(&sscape->lock, flags); + if (ret == 0) + data = host_read_ctrl_unsafe(sscape->io_base, 100); - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) { - snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n"); - } + if (data & 0x10) + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); - spin_lock_irqsave(&sscape->lock, flags); - set_midi_mode_unsafe(sscape->io_base); spin_unlock_irqrestore(&sscape->lock, flags); - initialise_mpu401(sscape->mpu); + data &= 0xf; + if (ret == 0 && data > 7) { + snd_printk(KERN_ERR + "sscape: timeout reading firmware version\n"); + ret = -EAGAIN; + } - return err; + return (ret == 0) ? data : ret; } /* - * Hardware-specific device functions, to implement special - * IOCTLs for the SoundScape card. This is how we upload - * the microcode into the card, for example, and so we - * must ensure that no two processes can open this device - * simultaneously, and that we can't open it at all if - * someone is using the MIDI device. + * Upload the microcode into the SoundScape. */ -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) +static int sscape_upload_microcode(struct snd_card *card, int version) { - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; + struct soundscape *sscape = get_card_soundscape(card); + const struct firmware *init_fw = NULL; + char name[14] = "sndscape.co0"; int err; - spin_lock_irqsave(&sscape->fwlock, flags); + name[11] = '0' + version; - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { - err = -EBUSY; - } else { - sscape->hw_in_use = 1; - err = 0; + err = request_firmware(&init_fw, name, card->dev); + if (err < 0) { + snd_printk(KERN_ERR "Error loading sndscape.co0"); + return err; } + err = upload_dma_data(sscape, init_fw->data, init_fw->size); + if (err == 0) + snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + init_fw->size >> 10); - spin_unlock_irqrestore(&sscape->fwlock, flags); - return err; -} - -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) -{ - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - sscape->hw_in_use = 0; - spin_unlock_irqrestore(&sscape->fwlock, flags); - return 0; -} - -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct soundscape *sscape = get_hwdep_soundscape(hw); - int err = -EBUSY; - - switch (cmd) { - case SND_SSCAPE_LOAD_BOOTB: - { - register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now ... - */ - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) - return -EFAULT; - - /* - * Now check that we can write the firmware version number too... - */ - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) - return -EFAULT; - - err = sscape_upload_bootblock(sscape, bb); - } - break; - - case SND_SSCAPE_LOAD_MCODE: - { - register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg; - - err = sscape_upload_microcode(sscape, mc); - } - break; - - default: - err = -EINVAL; - break; - } /* switch */ + release_firmware(init_fw); return err; } - /* * Mixer control for the SoundScape's MIDI device. */ @@ -920,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 * mpu) spin_lock_irqsave(&sscape->fwlock, flags); - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { + if (sscape->midi_usage == ULONG_MAX) { err = -EBUSY; } else { ++(sscape->midi_usage); @@ -1053,13 +936,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, } } - strcpy(card->driver, "SoundScape"); - strcpy(card->shortname, pcm->name); - snprintf(card->longname, sizeof(card->longname), - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", - pcm->name, chip->port, chip->irq, - chip->dma1, chip->dma2); - sscape->chip = chip; } @@ -1162,29 +1038,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) return -ENXIO; } - if (sscape->type != SSCAPE_VIVO) { - /* - * Now create the hardware-specific device so that we can - * load the microcode into the on-board processor. - * We cannot use the MPU-401 MIDI system until this firmware - * has been loaded into the card. - */ - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "firmware device\n"); - goto _release_dma; - } - strlcpy(sscape->hw->name, "SoundScape M68K", - sizeof(sscape->hw->name)); - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; - sscape->hw->ops.open = sscape_hw_open; - sscape->hw->ops.release = sscape_hw_release; - sscape->hw->ops.ioctl = sscape_hw_ioctl; - sscape->hw->private_data = sscape; - } - /* * Tell the on-board devices where their resources are (I think - * I can't be sure without a datasheet ... So many magic values!) @@ -1222,28 +1075,51 @@ static int __devinit create_sscape(int dev, struct snd_card *card) wss_port[dev], irq[dev]); goto _release_dma; } + strcpy(card->driver, "SoundScape"); + strcpy(card->shortname, name); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", + name, sscape->chip->port, sscape->chip->irq, + sscape->chip->dma1, sscape->chip->dma2); + #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%lx\n", - port[dev]); - goto _release_dma; - } + err = sscape_upload_bootblock(card); + if (err >= 0) + err = sscape_upload_microcode(card, err); + + if (err == 0) { + err = create_mpu401(card, MIDI_DEVNUM, port[dev], + mpu_irq[dev]); + if (err < 0) { + printk(KERN_ERR "sscape: Failed to create " + "MPU-401 device at 0x%lx\n", + port[dev]); + goto _release_dma; + } - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* - * Initialize mixer - */ - sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + /* + * Initialize mixer + */ + spin_lock_irqsave(&sscape->lock, flags); + sscape->midi_vol = 0; + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_XXX_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_EXTMIDI, 100); + host_write_ctrl_unsafe(sscape->io_base, + 0, 100); + host_write_ctrl_unsafe(sscape->io_base, 80, 1); + + set_midi_mode_unsafe(sscape->io_base); + spin_unlock_irqrestore(&sscape->lock, flags); + } } /* @@ -1301,11 +1177,12 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) sscape->type = SSCAPE; dma[dev] &= 0x03; + snd_card_set_dev(card, pdev); + ret = create_sscape(dev, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, pdev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; @@ -1426,12 +1303,12 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, wss_port[idx] = pnp_port_start(dev, 1); dma2[idx] = pnp_dma(dev, 1); } + snd_card_set_dev(card, &pcard->card->dev); ret = create_sscape(idx, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, &pcard->card->dev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; -- 1.6.0.3 ---------------------------------------------------------------------- Co godzine losujemy 15 000 z�! http://link.interia.pl/f2376

1 0

[alsa-devel] [PATCHv2 0/7] TWL6030 audio codec initial support
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

Following patch series adds initial support for TWL6030 codec driver. TWL6030 codec depends on TWL6030_CORE, which is not accepted yet. The main intention then is to get comments, I'll repost the patches of this series when TWL6030_CORE support gets accepted and rework patches as needed. Changes from v1: - Renamed dai format for PDM as SND_SOC_DAIFMT_PDM - CODEC driver registered as a platform driver - AUDPWRON gpio line handled in the CODEC driver - Handle scenario when no AUDPWRON gpio line is provided - Corrected names of some widgets - Avoid multiple calls to power functions in set_bias_level Thanks, -Misa --- Misael Lopez Cruz (7): OMAP4: PMIC: Add support for twl6030 codec ASoC: Add PDM DAI format definition ASoC: TWL6030: Add twl6030 codec driver ASoC: TWL6030: Manual power-up/down sequences ASoC: TWL6030: Add support for low-power mode ASoC: TWL6030: Enable audio interrupt ASoC: TWL6030: Power-up seq completion through audio interrupt drivers/mfd/twl-core.c | 15 + include/linux/i2c/twl.h | 11 + include/sound/soc-dai.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 1179 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 137 +++++ 7 files changed, 1349 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h

3 3

Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 28 Sep '09

28 Sep '09

Hi Peter, Thanks for the help. I don't believe it's possible for me to try the master/slave switch. Please take a look at http://www.efn.org/~rick/pub/wm8731.jpg Note that DACLRC and ADCLRC are connected. This requires me to run as slave, right? Our bit clock line is buffered so I don't think that will be a problem. Rick > > Yes, I had tried 8 before and just tried it now. Same result. It > > really seems like there is a DMA issue. Can you think of anything > > else to try before I dive into the DMA code? > > > Ok, that wasn't the reason. > > Have you tried to reverse roles of codec and omap? I.e. by using > the codec as a master and omap as a slave with SND_SOC_DAIFMT_CBM_CFM. > > I don't think there is any HW reason as you have seen the bit-clock and > frame sync signals toggling but I remember one case few years back > where codec had slight too high capacitive load on the bit-clock line. > That was then causing unstable frame sync as it was internally derived > directly from the bit-clock pin without any buffer. Bit-clock signal > looked just fine on oscilloscope but wasn't well enough for internal > circuits of the cpu. > > So if assuming similar problem here, I could imagine that if the FSX is > used similar way directly inside the omap and if signal is not well > enough, the McBSP and DMA may not be running. > > This migh be a bit far reason to look for but at least easy to try by > switching the roles as then the codec is driving the both bit-clock and > fs. > >

2 1

[alsa-devel] [PATCHv2 1/7] OMAP4: PMIC: Add support for twl6030 codec
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

In order to have TWL6030 CODEC driver as a platform driver, codec data should be passed through twl_platform_data structure. For twl6030 audio codec, the following data is required: - audpwron_gpio: gpio line used to power-up/down the codec. A low-to-high transition powers codec up. Setting audpwron_gpio to a negative value means that codec will use manual power sequence instead of automatic sequence - naudint_irq: irq line for audio interrupt. twl6030 drives NAUDINT line to low when an interrupt (codec ready, plug insertion/removal, etc) is detected Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- drivers/mfd/twl-core.c | 15 +++++++++++++++ include/linux/i2c/twl.h | 11 +++++++++++ 2 files changed, 26 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index af4cf47..e9d1ff1 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -108,6 +108,12 @@ #define twl_has_mmc() false #endif +#if defined(CONFIG_SND_SOC_TWL6030) +#define twl_has_codec() true +#else +#define twl_has_codec() false +#endif + #if defined(CONFIG_TWL4030_USB) || defined(CONFIG_TWL4030_USB_MODULE) #define twl_has_usb() true #else @@ -190,6 +196,7 @@ #define BCI_SUB_CHIP_ID SUB_CHIP_ID1 #define GPIO_SUB_CHIP_ID 0 /* NOT SUPPORTED IN TWL6030 */ #define KEYPAD_SUB_CHIP_ID 0 /* ADDED FOR COMPILATION ONLY */ +#define CODEC_SUB_CHIP_ID SUB_CHIP_ID3 /* subchip/slave 0 0x48 - POWER */ #define TWL6030_BASEADD_RTC 0x0000 @@ -632,6 +639,14 @@ add_children(struct twl_platform_data *pdata, unsigned long features) if (IS_ERR(child)) return PTR_ERR(child); } + + if (twl_has_codec() && pdata->codec) { + child = add_child(CODEC_SUB_CHIP_ID, "twl6030_codec", + pdata->codec, sizeof(*pdata->codec), false, + 0, 0); + if (IS_ERR(child)) + return PTR_ERR(child); + } #endif if (twl_has_usb() && pdata->usb) { diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index b687a8b..d8a8823 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -472,6 +472,16 @@ struct twl_usb_data { enum twl_usb_mode usb_mode; }; +struct snd_soc_codec; + +struct twl_codec_data { + struct snd_soc_codec *codec; + + /* twl6030 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ +}; + struct twl_platform_data { unsigned irq_base, irq_end; struct twl_bci_platform_data *bci; @@ -479,6 +489,7 @@ struct twl_platform_data { struct twl_madc_platform_data *madc; struct twl_keypad_data *keypad; struct twl_usb_data *usb; + struct twl_codec_data *codec; /* LDO regulators common to TWL4030/TWL6030 */ struct regulator_init_data *vdac; -- 1.5.4.3

2 1

[alsa-devel] snd-hda-intel and MSI
by Elimar Riesebieter 28 Sep '09

28 Sep '09

Hi folks, are there any contradictions to enable-msi=1 by default? There are users with chips like an 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Hewlett-Packard Company Device 30f7 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 33 Region 0: Memory at dc500000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel which only works with MSI. And it is somewhat hard to find the appropriate Documantation in HD-Audio.txt. 2nd it will be very hard for Distributers to maintain modul options by chip ID's. Thanks for cooperation. Elimar (one of Debian's alsa maintainers) -- Do you smell something burning or ist it me?

5 15

[alsa-devel] [PATCHv2 2/7] ASoC: Add PDM DAI format definition
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

Add DAI format definition for PDM interfaces. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc-dai.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..9239cb0 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -30,6 +30,7 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ #define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ +#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */ /* left and right justified also known as MSB and LSB respectively */ #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J -- 1.5.4.3

2 1

[alsa-devel] [PATCHv2 7/7] ASoC: TWL6030: Power-up seq completion through audio interrupt
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

When the codec is powered-up through external AUDPWRON line it starts its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. CODEC driver starts a wait_for_completion just after AUDPWRON line transitions from low to high. It's signaled as complete when servicing READYINT interrupt. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 46 ++++++++++++++++++++++++++++--------------- 1 files changed, 30 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 5cf2099..f8bd8ee 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -48,6 +48,7 @@ struct twl6030_data { int codec_powered; unsigned int sysclk; struct work_struct audint_work; + struct completion ready_completion; }; /* @@ -58,7 +59,7 @@ static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { 0x4B, /* TWL6030_ASICID (ro) 0x01 */ 0x00, /* TWL6030_ASICREV (ro) 0x02 */ 0x00, /* TWL6030_INTID 0x03 */ - 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_INTMR 0x04 */ 0x00, /* TWL6030_NCPCTRL 0x05 */ 0x00, /* TWL6030_LDOCTL 0x06 */ 0x00, /* TWL6030_HPPLLCTL 0x07 */ @@ -186,6 +187,23 @@ static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, } /* + * read from twl6030 hardware register + */ +static int twl6030_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg > TWL6030_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &value, reg); + twl6030_write_reg_cache(codec, reg, value); + + return value; +} + +/* * write to the twl6030 register space */ static int twl6030_write(struct snd_soc_codec *codec, @@ -372,7 +390,8 @@ void twl6030_naudint_work(struct work_struct *work) dev_alert(codec->dev, "vib drivers over current detection\n"); break; case TWL6030_READYINT: - dev_alert(codec->dev, "codec is ready\n"); + priv->codec_powered = 1; + complete(&priv->ready_completion); break; default: dev_err(codec->dev, "unknown audio interrupt %d\n", intid); @@ -617,25 +636,21 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* use AUDPWRON line */ gpio_set_value(audpwron_gpio, 1); - /* power-up sequence latency */ - mdelay(16); + /* wait for ready interrupt */ + wait_for_completion(&priv->ready_completion); /* sync registers updated during power-up sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, - 0x81); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, - 0x45); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, - 0x01); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); } else { /* use manual power-up sequence */ twl6030_power_up(codec); + priv->codec_powered = 1; } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); - - priv->codec_powered = 1; break; case SND_SOC_BIAS_OFF: if (!priv->codec_powered) @@ -649,10 +664,8 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, udelay(500); /* sync registers updated during power-down sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, - 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, - 0x00); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); } else { @@ -1061,6 +1074,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) /* audio interrupt */ INIT_WORK(&priv->audint_work, twl6030_naudint_work); + init_completion(&priv->ready_completion); ret = request_irq(naudint_irq, twl6030_naudint_handler, -- 1.5.4.3

2 1

[alsa-devel] [PATCHv2 6/7] ASoC: TWL6030: Enable audio interrupt
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

NAUDINT interrupt line is provided by the TWL6030 codec to signal externally events like headset plug/unplug, hook, power-up sequence completion, etc. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 78 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl6030.h | 10 ++++++ 2 files changed, 87 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 032619d..5cf2099 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -47,6 +47,7 @@ struct twl6030_data { struct snd_soc_codec codec; int codec_powered; unsigned int sysclk; + struct work_struct audint_work; }; /* @@ -329,6 +330,58 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +/* audio interrupt handler */ +irqreturn_t twl6030_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6030_data *priv = codec->private_data; + + schedule_work(&priv->audint_work); + + /* disable audint irq to let workqueue to execute */ + disable_irq_nosync(irq); + + return IRQ_HANDLED; +} + +void twl6030_naudint_work(struct work_struct *work) +{ + struct twl_codec_data *twl_codec; + struct snd_soc_codec *codec; + struct twl6030_data *priv; + u8 intid; + + priv = container_of(work, struct twl6030_data, audint_work); + codec = &priv->codec; + twl_codec = codec->control_data; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &intid, TWL6030_REG_INTID); + + switch (intid) { + case TWL6030_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6030_PLUGINT: + case TWL6030_UNPLUGINT: + case TWL6030_HOOKINT: + break; + case TWL6030_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6030_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6030_READYINT: + dev_alert(codec->dev, "codec is ready\n"); + break; + default: + dev_err(codec->dev, "unknown audio interrupt %d\n", intid); + break; + } + + enable_irq(twl_codec->naudint_irq); +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -610,6 +663,7 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, priv->codec_powered = 0; break; } + codec->bias_level = level; return 0; @@ -954,8 +1008,15 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) struct twl6030_data *priv; struct snd_soc_codec *codec; int audpwron_gpio = twl_codec->audpwron_gpio; + int naudint_irq = twl_codec->naudint_irq; int ret = 0; + /* prerequisites */ + if (!naudint_irq) { + dev_err(&pdev->dev, "no audio interrupt irq supplied\n"); + return -EINVAL; + } + priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); if (priv == NULL) return -ENOMEM; @@ -998,6 +1059,17 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) priv->codec_powered = 0; } + /* audio interrupt */ + INIT_WORK(&priv->audint_work, twl6030_naudint_work); + + ret = request_irq(naudint_irq, + twl6030_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_DISABLED, + "twl6030-codec", + codec); + if (ret) + goto gpio2_err; + /* init vio registers */ twl6030_init_vio_regs(codec); @@ -1006,7 +1078,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(codec); if (ret) - goto gpio2_err; + goto reg_err; twl6030_codec = codec; @@ -1019,6 +1091,8 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) dai_err: snd_soc_unregister_codec(codec); twl6030_codec = NULL; +reg_err: + free_irq(naudint_irq, twl_codec); gpio2_err: if (gpio_is_valid(audpwron_gpio)) gpio_free(audpwron_gpio); @@ -1036,6 +1110,8 @@ static int __devexit twl6030_codec_remove(struct platform_device *pdev) if (gpio_is_valid(twl_codec->audpwron_gpio)) gpio_free(twl_codec->audpwron_gpio); + free_irq(twl_codec->naudint_irq, twl6030_codec); + snd_soc_unregister_dai(&twl6030_dai); snd_soc_unregister_codec(twl6030_codec); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 15d3e1b..8a106f2 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,16 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* INTID (0x03) fields */ + +#define TWL6030_THINT 0x01 +#define TWL6030_PLUGINT 0x02 +#define TWL6030_UNPLUGINT 0x04 +#define TWL6030_HOOKINT 0x08 +#define TWL6030_HFINT 0x10 +#define TWL6030_VIBINT 0x20 +#define TWL6030_READYINT 0x40 + /* NCPCTL (0x05) fields */ #define TWL6030_NCPENA 0x01 -- 1.5.4.3

2 1

[alsa-devel] [PATCHv2 4/7] ASoC: TWL6030: Manual power-up/down sequences
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

TWL6030 codec device can be powered-up/down through a specific register writes sequence. These sequences can be used when no gpio line is provided for AUDPWRON. When the codec is powered-up in this way, automatic power-down sequence (triggered by thermal shutdown) is not possible. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 117 +++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/twl6030.h | 16 ++++++ 2 files changed, 120 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 92797e7..f1e333f 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -223,6 +223,88 @@ static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) } } +/* twl6030 codec manual power-up sequence */ +static void twl6030_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6030_NCPENA | TWL6030_NCPOPEN; + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); +} + +/* twl6030 codec manual power-down sequence */ +static void twl6030_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6030_NCPENA | TWL6030_NCPOPEN); + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -460,12 +542,18 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-up sequence latency */ mdelay(16); - } - /* sync registers updated during power-up sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); + /* sync registers updated during power-up sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, + 0x81); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, + 0x45); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, + 0x01); + } else { + /* use manual power-up sequence */ + twl6030_power_up(codec); + } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); @@ -482,12 +570,18 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-down sequence latency */ udelay(500); - } - /* sync registers updated during power-down sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + /* sync registers updated during power-down sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, + 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, + 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6030_power_down(codec); + } priv->codec_powered = 0; break; @@ -746,9 +840,6 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) goto gpio2_err; priv->codec_powered = 0; - } else { - /* if no gpio is provided, then assume its always on */ - priv->codec_powered = 1; } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index d99d469..7375ae8 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,18 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* NCPCTL (0x05) fields */ + +#define TWL6030_NCPENA 0x01 +#define TWL6030_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6030_LSLDOENA 0x01 +#define TWL6030_HSLDOENA 0x04 +#define TWL6030_REFENA 0x40 +#define TWL6030_OSCENA 0x80 + /* HPPLLCTL (0x07) fields */ #define TWL6030_HPLLENA 0x01 @@ -88,6 +100,10 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +/* ACCCTL (0x2D) fields */ + +#define TWL6030_RESETSPLIT 0x04 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.5.4.3

2 1

[alsa-devel] [PATCHv2 3/7] ASoC: TWL6030: Add twl6030 codec driver
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

Initial version of TWL6030 codec driver. The TWL6030 codec uses a propietary PDM-based digital audio interface. TWL6030 codec has two power modes: low-power and high-performance, this initial version only supports high-performance mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 842 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 94 +++++ 4 files changed, 942 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..fd9f8c1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6030 if TWL6030_CORE && GPIOLIB select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 @@ -114,6 +115,9 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TWL4030 tristate +config SND_SOC_TWL6030 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..b70c8a1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -13,6 +13,7 @@ snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-twl6030-objs := twl6030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -50,6 +51,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TWL6030) += snd-soc-twl6030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c new file mode 100644 index 0000000..92797e7 --- /dev/null +++ b/sound/soc/codecs/twl6030.c @@ -0,0 +1,842 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl6030.h" + +#define TWL6030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6030_data { + struct snd_soc_codec codec; + int codec_powered; +}; + +/* + * twl6030 register cache & default register settings + */ +static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6030_ASICID (ro) 0x01 */ + 0x00, /* TWL6030_ASICREV (ro) 0x02 */ + 0x00, /* TWL6030_INTID 0x03 */ + 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_NCPCTRL 0x05 */ + 0x00, /* TWL6030_LDOCTL 0x06 */ + 0x00, /* TWL6030_HPPLLCTL 0x07 */ + 0x00, /* TWL6030_LPPLLCTL 0x08 */ + 0x00, /* TWL6030_LPPLLDIV 0x09 */ + 0x00, /* TWL6030_AMICBCTL 0x0A */ + 0x00, /* TWL6030_DMICBCTL 0x0B */ + 0x18, /* TWL6030_MICLCTL 0x0C */ + 0x18, /* TWL6030_MICRCTL 0x0D */ + 0x00, /* TWL6030_MICGAIN 0x0E */ + 0x1B, /* TWL6030_LINEGAIN 0x0F */ + 0x00, /* TWL6030_HSLCTL 0x10 */ + 0x00, /* TWL6030_HSRCTL 0x11 */ + 0x00, /* TWL6030_HSGAIN 0x12 */ + 0x06, /* TWL6030_EARCTL 0x13 */ + 0x00, /* TWL6030_HFLCTL 0x14 */ + 0x03, /* TWL6030_HFLGAIN 0x15 */ + 0x00, /* TWL6030_HFRCTL 0x16 */ + 0x03, /* TWL6030_HFRGAIN 0x17 */ + 0x00, /* TWL6030_VIBCTLL 0x18 */ + 0x00, /* TWL6030_VIBDATL 0x19 */ + 0x00, /* TWL6030_VIBCTLR 0x1A */ + 0x00, /* TWL6030_VIBDATR 0x1B */ + 0x00, /* TWL6030_HKCTL1 0x1C */ + 0x00, /* TWL6030_HKCTL2 0x1D */ + 0x00, /* TWL6030_GPOCTL 0x1E */ + 0x00, /* TWL6030_ALB 0x1F */ + 0x00, /* TWL6030_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6030_TRIM1 0x28 */ + 0x00, /* TWL6030_TRIM2 0x29 */ + 0x00, /* TWL6030_TRIM3 0x2A */ + 0x00, /* TWL6030_HSOTRIM 0x2B */ + 0x00, /* TWL6030_HFOTRIM 0x2C */ + 0x09, /* TWL6030_ACCCTL 0x2D */ + 0x00, /* TWL6030_STATUS (ro) 0x2E */ +}; + +/* + * twl6030 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6030_vio_reg[TWL6030_VIOREGNUM] = { + TWL6030_REG_ASICID, + TWL6030_REG_ASICREV, + TWL6030_REG_INTID, + TWL6030_REG_INTMR, + TWL6030_REG_NCPCTL, + TWL6030_REG_LDOCTL, + TWL6030_REG_AMICBCTL, + TWL6030_REG_DMICBCTL, + TWL6030_REG_HKCTL1, + TWL6030_REG_HKCTL2, + TWL6030_REG_GPOCTL, + TWL6030_REG_TRIM1, + TWL6030_REG_TRIM2, + TWL6030_REG_TRIM3, + TWL6030_REG_HSOTRIM, + TWL6030_REG_HFOTRIM, + TWL6030_REG_ACCCTL, + TWL6030_REG_STATUS, +}; + +/* + * twl6030 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6030_vdd_reg[TWL6030_VDDREGNUM] = { + TWL6030_REG_HPPLLCTL, + TWL6030_REG_LPPLLCTL, + TWL6030_REG_LPPLLDIV, + TWL6030_REG_MICLCTL, + TWL6030_REG_MICRCTL, + TWL6030_REG_MICGAIN, + TWL6030_REG_LINEGAIN, + TWL6030_REG_HSLCTL, + TWL6030_REG_HSRCTL, + TWL6030_REG_HSGAIN, + TWL6030_REG_EARCTL, + TWL6030_REG_HFLCTL, + TWL6030_REG_HFLGAIN, + TWL6030_REG_HFRCTL, + TWL6030_REG_HFRGAIN, + TWL6030_REG_VIBCTLL, + TWL6030_REG_VIBDATL, + TWL6030_REG_VIBCTLR, + TWL6030_REG_VIBDATR, + TWL6030_REG_ALB, + TWL6030_REG_DLB, +}; + +/* + * read twl6030 register cache + */ +static inline unsigned int twl6030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6030 register cache + */ +static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the twl6030 register space + */ +static int twl6030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if ((reg < TWL6030_REG_INTID) || (reg > TWL6030_REG_ACCCTL)) + /* read-only registers */ + return -EIO; + + twl6030_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL6030_MODULE_AUDIO, value, reg); +} + +static void twl6030_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6030_write(codec, TWL6030_REG_ACCCTL, cache[TWL6030_REG_ACCCTL]); + + /* avoid read-only registers (ASICID, ASICREV, STATUS) */ + for (i = 2; i < (TWL6030_VIOREGNUM - 1); i++) { + reg = twl6030_vio_reg[i]; + twl6030_write(codec, reg, cache[reg]); + } +} + +static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6030_VDDREGNUM; i++) { + reg = twl6030_vdd_reg[i]; + twl6030_write(codec, reg, cache[reg]); + } +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6030_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6030_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6030_enum[] = { + SOC_ENUM_SINGLE(TWL6030_REG_MICLCTL, 3, 3, twl6030_amicl_texts), + SOC_ENUM_SINGLE(TWL6030_REG_MICRCTL, 3, 3, twl6030_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6030_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6030_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6030_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6030_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6030_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6030_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6030_REG_HFLGAIN, TWL6030_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6030_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6030_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6030_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6030_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6030_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6030_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6030_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6030_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6030_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6030_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6030_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6030_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6030_dapm_widgets, + ARRAY_SIZE(twl6030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl_codec_data *twl_codec = codec->control_data; + struct twl6030_data *priv = codec->private_data; + int audpwron_gpio = twl_codec->audpwron_gpio; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + return 0; + + if (gpio_is_valid(audpwron_gpio)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron_gpio, 1); + + /* power-up sequence latency */ + mdelay(16); + } + + /* sync registers updated during power-up sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); + + /* initialize vdd/vss registers with reg_cache */ + twl6030_init_vdd_regs(codec); + + priv->codec_powered = 1; + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + return 0; + + if (gpio_is_valid(audpwron_gpio)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron_gpio, 0); + + /* power-down sequence latency */ + udelay(500); + } + + /* sync registers updated during power-down sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + + priv->codec_powered = 0; + break; + } + codec->bias_level = level; + + return 0; +} + +static int twl6030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int rate, format; + + /* hardware dai (McPDM) requires bit stream of twice + * the actual rate, application processor (audio backend) + * should upsample accordingly + */ + rate = params_rate(params); + switch (rate) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unknown rate %d\n", rate); + return -EINVAL; + } + + /* the sample lenght handled by codec at A-D and D-A + * conversion is 16-bits, but the dai requires 32-bits + */ + format = params_format(params); + if (format != SNDRV_PCM_FORMAT_S32_LE) { + dev_err(codec->dev, "unknown format %d\n", format); + return -EINVAL; + } + + return 0; +} + +static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 hppll, lppll; + + hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + hppll &= TWL6030_HPLLRST; + + switch (freq) { + case 12000000: + /* MCLK input, PLL enabled */ + hppll = TWL6030_MCLK_12000KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLENA; + break; + case 19200000: + /* MCLK input, PLL disabled */ + hppll = TWL6030_MCLK_19200KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLBP; + break; + case 26000000: + /* MCLK input, PLL enabled */ + hppll = TWL6030_MCLK_26000KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer input, PLL disabled */ + hppll = TWL6030_MCLK_38400KHZ + | TWL6030_HPLLSQRENA + | TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + + /* Disable LPPLL and select HPPLL */ + lppll = TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + return 0; +} + +static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + int format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* propietary pdm dai format */ + if (format != SND_SOC_DAIFMT_PDM) + return -EINVAL; + + return 0; +} + +static struct snd_soc_dai_ops twl6030_dai_ops = { + .hw_params = twl6030_hw_params, + .set_sysclk = twl6030_set_dai_sysclk, + .set_fmt = twl6030_set_dai_fmt, +}; + +struct snd_soc_dai twl6030_dai = { + .name = "twl6030", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .ops = &twl6030_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6030_dai); + +static int twl6030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6030_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *twl6030_codec; + +static int twl6030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6030_codec); + + codec = twl6030_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6030_snd_controls, + ARRAY_SIZE(twl6030_snd_controls)); + twl6030_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6030 = { + .probe = twl6030_probe, + .remove = twl6030_remove, + .suspend = twl6030_suspend, + .resume = twl6030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6030); + +static int __devinit twl6030_codec_probe(struct platform_device *pdev) +{ + struct twl_codec_data *twl_codec = pdev->dev.platform_data; + struct twl6030_data *priv; + struct snd_soc_codec *codec; + int audpwron_gpio = twl_codec->audpwron_gpio; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + codec = &priv->codec; + twl_codec->codec = codec; + codec->dev = &pdev->dev; + twl6030_dai.dev = &pdev->dev; + + codec->name = "twl6030"; + codec->owner = THIS_MODULE; + codec->read = twl6030_read_reg_cache; + codec->write = twl6030_write; + codec->set_bias_level = twl6030_set_bias_level; + codec->private_data = priv; + codec->control_data = twl_codec; + codec->dai = &twl6030_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6030_reg); + codec->reg_cache = kmemdup(twl6030_reg, sizeof(twl6030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (gpio_is_valid(audpwron_gpio)) { + ret = gpio_request(audpwron_gpio, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron_gpio, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } else { + /* if no gpio is provided, then assume its always on */ + priv->codec_powered = 1; + } + + /* init vio registers */ + twl6030_init_vio_regs(codec); + + /* power on device */ + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_register_codec(codec); + if (ret) + goto gpio2_err; + + twl6030_codec = codec; + + ret = snd_soc_register_dai(&twl6030_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6030_codec = NULL; +gpio2_err: + if (gpio_is_valid(audpwron_gpio)) + gpio_free(audpwron_gpio); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6030_codec_remove(struct platform_device *pdev) +{ + struct twl_codec_data *twl_codec = pdev->dev.platform_data; + + if (gpio_is_valid(twl_codec->audpwron_gpio)) + gpio_free(twl_codec->audpwron_gpio); + + snd_soc_unregister_dai(&twl6030_dai); + snd_soc_unregister_codec(twl6030_codec); + + kfree(twl6030_codec); + twl6030_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int twl6030_codec_suspend(struct platform_device *pdev, + pm_message_t msg) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int twl6030_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define twl6030_codec_suspend NULL +#define twl6030_codec_resume NULL +#endif + +static struct platform_driver twl6030_codec_driver = { + .driver = { + .name = "twl6030_codec", + .owner = THIS_MODULE, + }, + .probe = twl6030_codec_probe, + .remove = __devexit_p(twl6030_codec_remove), + .suspend = twl6030_codec_suspend, + .resume = twl6030_codec_resume, +}; + +static int __init twl6030_codec_init(void) +{ + return platform_driver_register(&twl6030_codec_driver); +} +module_init(twl6030_codec_init); + +static void __exit twl6030_codec_exit(void) +{ + platform_driver_unregister(&twl6030_codec_driver); +} +module_exit(twl6030_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6030 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h new file mode 100644 index 0000000..d99d469 --- /dev/null +++ b/sound/soc/codecs/twl6030.h @@ -0,0 +1,94 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL6030_H__ +#define __TWL6030_H__ + +#define TWL6030_REG_ASICID 0x01 +#define TWL6030_REG_ASICREV 0x02 +#define TWL6030_REG_INTID 0x03 +#define TWL6030_REG_INTMR 0x04 +#define TWL6030_REG_NCPCTL 0x05 +#define TWL6030_REG_LDOCTL 0x06 +#define TWL6030_REG_HPPLLCTL 0x07 +#define TWL6030_REG_LPPLLCTL 0x08 +#define TWL6030_REG_LPPLLDIV 0x09 +#define TWL6030_REG_AMICBCTL 0x0A +#define TWL6030_REG_DMICBCTL 0x0B +#define TWL6030_REG_MICLCTL 0x0C +#define TWL6030_REG_MICRCTL 0x0D +#define TWL6030_REG_MICGAIN 0x0E +#define TWL6030_REG_LINEGAIN 0x0F +#define TWL6030_REG_HSLCTL 0x10 +#define TWL6030_REG_HSRCTL 0x11 +#define TWL6030_REG_HSGAIN 0x12 +#define TWL6030_REG_EARCTL 0x13 +#define TWL6030_REG_HFLCTL 0x14 +#define TWL6030_REG_HFLGAIN 0x15 +#define TWL6030_REG_HFRCTL 0x16 +#define TWL6030_REG_HFRGAIN 0x17 +#define TWL6030_REG_VIBCTLL 0x18 +#define TWL6030_REG_VIBDATL 0x19 +#define TWL6030_REG_VIBCTLR 0x1A +#define TWL6030_REG_VIBDATR 0x1B +#define TWL6030_REG_HKCTL1 0x1C +#define TWL6030_REG_HKCTL2 0x1D +#define TWL6030_REG_GPOCTL 0x1E +#define TWL6030_REG_ALB 0x1F +#define TWL6030_REG_DLB 0x20 +#define TWL6030_REG_TRIM1 0x28 +#define TWL6030_REG_TRIM2 0x29 +#define TWL6030_REG_TRIM3 0x2A +#define TWL6030_REG_HSOTRIM 0x2B +#define TWL6030_REG_HFOTRIM 0x2C +#define TWL6030_REG_ACCCTL 0x2D +#define TWL6030_REG_STATUS 0x2E + +#define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) + +#define TWL6030_VIOREGNUM 18 +#define TWL6030_VDDREGNUM 21 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6030_HPLLENA 0x01 +#define TWL6030_HPLLRST 0x02 +#define TWL6030_HPLLBP 0x04 +#define TWL6030_HPLLSQRENA 0x08 +#define TWL6030_HPLLSQRBP 0x10 +#define TWL6030_MCLK_12000KHZ (0 << 5) +#define TWL6030_MCLK_19200KHZ (1 << 5) +#define TWL6030_MCLK_26000KHZ (2 << 5) +#define TWL6030_MCLK_38400KHZ (3 << 5) +#define TWL6030_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6030_LPLLENA 0x01 +#define TWL6030_LPLLRST 0x02 +#define TWL6030_LPLLSEL 0x04 +#define TWL6030_LPLLFIN 0x08 +#define TWL6030_HPLLSEL 0x10 + +extern struct snd_soc_dai twl6030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6030; + +#endif /* End of __TWL6030_H__ */ -- 1.5.4.3

2 1

[alsa-devel] [PATCH 0/3] Add quirks for Vaio VGN
by Pascal Terjan 28 Sep '09

28 Sep '09

Here are some quirks for 3 Vaio VGN model reported on launchpad to need model=vaio Pascal Terjan (3): [ALSA] Add quirk for Sony Vaio VGN-SZ650N [ALSA] Add quirk for Sony Vaio VGN-AR51M [ALSA] Add quirk for Sony Vaio VGN-FZ15G sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-)

2 7

[alsa-devel] ctl_elems hooks with dmix plugin
by George Kuriakose 28 Sep '09

28 Sep '09

Hi I was trying to use ctl_elems hooks with dmix plugin but figured that this combination doesn't work. Is there any workaround for this. I am trying the following. There are two softvol devices s0 and s1 going through dmix plugins. I wan't to mute s0 when any app opens s1 and unmute s0 when s1 is closed. I am using the below asoundrc file which results in "No card for this PCM" . Any workarounds or any minor change in libasound which will make this work? Thanks George asoundrc: pcm.softvol0 { type softvol slave { pcm "dmix:1" } control { name "SoftControl0" card 1 } } pcm.softvol1 { type softvol slave { pcm "dmix:1" } control { name "SoftControl1" card 1 } } pcm.s0 { type plug slave.pcm "softvol0" } pcm.s1 { type plug slave.pcm "softvol1" } pcm.s1hook { type hooks # PCM with hooks slave.pcm "s1" hooks.0 { type ctl_elems hook_args [ { name "SoftControl0" preserve true lock true optional true value [ 0 ] } { name "SoftControl1" preserve true lock true optional true value [ 100 ] } ] } }

2 4

[alsa-devel] [PATCH] sscape: add supoort for SPEA Media FX/Reveal SC-600
by Krzysztof Helt 28 Sep '09

28 Sep '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Move code from the OSS sscape driver in order to support old Soundscape OEM models. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Only PCM works. sound/isa/Kconfig | 6 ++- sound/isa/sscape.c | 116 ++++++++++++++++++++++++++++++++++++--------------- 2 files changed, 86 insertions(+), 36 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 51a7e37..b90fc16 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -377,10 +377,12 @@ config SND_SSCAPE select SND_WSS_LIB help Say Y here to include support for Ensoniq SoundScape - soundcards. + and Ensoniq OEM soundcards. The PCM audio is supported on SoundScape Classic, Elite, PnP - and VIVO cards. The MIDI support is very experimental. + and VIVO cards. The supported OEM cards are SPEA Media FX and + Reveal SC-600. + The MIDI support is very experimental. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 6618712..b11c35f 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -127,7 +127,8 @@ enum GA_REG { enum card_type { - SSCAPE, + MEDIA_FX, /* Sequoia S-1000 */ + SSCAPE, /* Sequoia S-2000 */ SSCAPE_PNP, SSCAPE_VIVO, }; @@ -784,20 +785,25 @@ static struct snd_kcontrol_new midi_mixer_ctl = { * These IRQs are encoded as bit patterns so that they can be * written to the control registers. */ -static unsigned __devinit get_irq_config(int irq) +static unsigned __devinit get_irq_config(int sscape_type, int irq) { static const int valid_irq[] = { 9, 5, 7, 10 }; + static const int old_irq[] = { 9, 7, 5, 15 }; unsigned cfg; - for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) { - if (irq == valid_irq[cfg]) - return cfg; - } /* for */ + if (sscape_type == MEDIA_FX) { + for (cfg = 0; cfg < ARRAY_SIZE(old_irq); ++cfg) + if (irq == old_irq[cfg]) + return cfg; + } else { + for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) + if (irq == valid_irq[cfg]) + return cfg; + } return INVALID_IRQ; } - /* * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. @@ -842,11 +848,39 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e) goto _done; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + if (s->ic_type == IC_OPUS) + activate_ad1845_unsafe(s->io_base); if (s->type == SSCAPE_VIVO) wss_io += 4; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + + /* wait for WSS codec */ + for (d = 0; d < 500; d++) { + if ((inb(wss_io) & 0x80) == 0) + break; + spin_unlock_irqrestore(&s->lock, flags); + msleep(1); + spin_lock_irqsave(&s->lock, flags); + } + snd_printd(KERN_INFO "init delay = %d ms\n", d); + + if ((inb(wss_io) & 0x80) != 0) + goto _done; + + if (inb(wss_io + 2) == 0xff) + goto _done; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d); + + if ((inb(wss_io) & 0x80) != 0) + s->type = MEDIA_FX; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { if ((inb(wss_io) & 0x80) == 0) @@ -954,9 +988,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, if (sscape->type == SSCAPE_VIVO) port += 4; - if (dma1 == dma2) - dma2 = -1; - err = snd_wss_create(card, port, -1, irq, dma1, dma2, WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); if (!err) { @@ -1051,21 +1082,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; - - /* - * Check that the user didn't pass us garbage data ... - */ - irq_cfg = get_irq_config(irq[dev]); - if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; - } - - mpu_irq_cfg = get_irq_config(mpu_irq[dev]); - if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; - } + const char *name; /* * Grab IO ports that we will need to probe so that we @@ -1109,8 +1126,41 @@ static int __devinit create_sscape(int dev, struct snd_card *card) goto _release_dma; } - printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n", - sscape->io_base, irq[dev], dma[dev]); + switch (sscape->type) { + case MEDIA_FX: + name = "MediaFX/SoundFX"; + break; + case SSCAPE: + name = "Soundscape"; + break; + case SSCAPE_PNP: + name = "Soundscape PnP"; + break; + case SSCAPE_VIVO: + name = "Soundscape VIVO"; + break; + default: + name = "unknown Soundscape"; + break; + } + + printk(KERN_INFO "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", + name, sscape->io_base, irq[dev], dma[dev]); + + /* + * Check that the user didn't pass us garbage data ... + */ + irq_cfg = get_irq_config(sscape->type, irq[dev]); + if (irq_cfg == INVALID_IRQ) { + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + return -ENXIO; + } + + mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); + if (mpu_irq_cfg == INVALID_IRQ) { + printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + return -ENXIO; + } if (sscape->type != SSCAPE_VIVO) { /* @@ -1141,8 +1191,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - activate_ad1845_unsafe(sscape->io_base); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1151,12 +1199,12 @@ static int __devinit create_sscape(int dev, struct snd_card *card) * Enable and configure the DMA channels ... */ sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50); - dma_cfg = (sscape->ic_type == IC_ODIE ? 0x70 : 0x40); + dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70); sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg); sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); - sscape_write_unsafe(sscape->io_base, - GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg); + mpu_irq_cfg |= mpu_irq_cfg << 2; + sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); -- 1.6.0.3 ---------------------------------------------------------------------- Kawalerka za 79 000 zl! Sprawdz >>> http://link.interia.pl/f2348

2 1

[alsa-devel] ASoC update for 2.6.33
by Mark Brown 28 Sep '09

28 Sep '09

The following changes since commit 2c9ee33d37a6f3808c4319f0849671fafa8a6f4c: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Graeme Gregory (1): ASoC: pxa-ssp increase max_channels to 8 sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-)

2 1

[alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 28 Sep '09

28 Sep '09

Tested this on my Zotac IONITX motherboard (N330 model). Dmesg output: [ 16.464533] ALSA hda_codec.c:3081: hda_codec: model '3stack-6ch' is selected for config 174b:437b (Zotac IONITX) Patch: --- alsa-kernel/pci/hda/patch_realtek.c.orig 2009-09-24 14:50:45.000000000 -0700 +++ alsa-kernel/pci/hda/patch_realtek.c 2009-09-24 14:38:07.000000000 -0700 @@ -16724,6 +16724,7 @@ static struct snd_pci_quirk alc662_cfg_t ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x174b, 0x437b, "Zotac IONITX", ALC662_3ST_6ch), {} }; Thanks, Gus ------------------------- Message sent via Atmail Open - http://atmail.org/

2 1

[alsa-devel] No sound after returning from suspend-to-ram
by Ozan Çağlayan 28 Sep '09

28 Sep '09

Hi, A user complains that after returning from suspend-to-ram, he isn't able to get any sound from his laptop until rebooting. I delayed reporting this one because I once thought that it may be related to Pulseaudio or some kind of mixer level problem but now I tried to suspect the kernel and the drivers as well. This is 2.6.30.5 with alsa-* 1.0.21a. I have dmesg and alsa-info outputs before and after s2ram: dmesg before s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4454 alsa-info before s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4455 dmesg after s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4456 alsa-info after s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4457 Note that a traceback is dumped in dmesg during resume. Maybe it will give a clue about the problem, dunno. And there's something like hda_codec: invalid CONNECT_LIST verb 12[2]:2100 in regular dmesg output, i wonder what does that mean too. Any ideas? Thanks, Ozan Caglayan

2 5

[alsa-devel] Problem with mplayer and OSS on at91sam9g20(kernel 2.6.30)
by Sedji Gaouaou 28 Sep '09

28 Sep '09

Hi, We've been doing some tests here and we noticed a problem when using mplayer that we didn't have previously(on 2.6.27). While playing audio through OSS emulation on 2.6.30 / at91sam9g20ek there seems to be issue with hardware parameters. The sound does not play and a bunch of error messages are comming on the console. Here is the test case on angstrom: -- mplayer output sample ------------------------- mplayer -quiet myfile.mp3 MPlayer 1.0rc1-4.1.2 (C) 2000-2006 MPlayer Team CPU: ARM Playing /media/sda/audio/01.mp3. [..] AUDIO: 44100 Hz, 2 ch, s16le, 192.0 kbit/13.61% (ratio: 24000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) [AO OSS]setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-( -- kernel output sample ------------------------- Atmel_ssc_dai: Entering atmel_ssc_hw_params() natmel_ssc_hw_params: RCMR=00010420 RFMR=001f008f TCMR=4a010404 TFMR=001f008f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f008f TCMR=4a010404 TFMR=001f008f atmel_ssc_dai,hw_params: SSC initialized Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f018f TCMR=4a010404 TFMR=001f018f atmel_ssc_dai,hw_params: SSC initialized asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed [..] asoc: machine hw_params failed MPlayer interrupted by signal 2 in module: ao2_init atmel_ssc_shutdown: transmit disabled SSC_SR=0x000004cc atmel_ssc_dau: Stopping clock soc-audio soc-audio: shutdown pop wq checking: Playback status: inactive waiting: yes pop wq D1 WM8731 Playback pop wq D3 WM8731 Playback -- 2 error messages to note ------------------------- mplayer: "[AO OSS]setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(" kernel: "asoc: machine hw_params failed" => please note also that it is working correctly using the alsa output: mplayer -ao alsa myfile.mp3 Here is the log while using the alsa output (it is working in this case): root@at91sam:~$ mplayer -quiet -ao alsa /media/sda/audio/01.mp3 MPlayer 1.0rc1-4.1.2 (C) 2000-2006 MPlayer Team CPU: ARM Playing /media/sda/audio/01.mp3. Audio file file format detected. Clip info: Title: Angel Artist: Massive Attack Album: Mezzanine Year: 1998 Comment: Ripped by Beto Track: 1 Genre: Rock ========================================================================== Forced audio codec: mad Opening audio decoder: [libmad] libmad mpeg audio decoder AUDIO: 44100 Hz, 2 ch, s16le, 192.0 kbit/13.61% (ratio: 24000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) ========================================================================== atmel_ssc_startup: SSC_SR=0x204 asoc: WM8731 <-> atmel-ssc0 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f018f TCMR=19010404 TFMR=001f018f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized transmit enabled SSC_SR=0x000104cf AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... [..] atmel_ssc_shutdown: transmit disabled SSC_SR=0x000104c3 atmel_ssc_dau: Stopping clock soc-audio soc-audio: shutdown Exiting... (Quit) pop wq checking: Playback status: inactive waiting: yes pop wq D1 WM8731 Playback pop wq D3 WM8731 Playback egards, Sedji

3 4

[alsa-devel] [PATCH 0/1] alsa-lib: pcm_hw: Always use delay ioctl in snd_pcm_delay()
by Kai Vehmanen 28 Sep '09

28 Sep '09

Hello all, this is a updated version of the alsa-lib patch I mentioned in the earlier thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-August/020377.html That was meant to just describe the possible problem, but now I'm more confident that this should be considered for merging. The related alsa-kernel patch is 'ALSA: Add extra delay count in PCM', which added the 'runtime->delay' property. Please review if this patch makes sense. Kai Vehmanen (1): pcm_hw: Always use delay ioctl in snd_pcm_delay() src/pcm/pcm_hw.c | 22 ---------------------- 1 files changed, 0 insertions(+), 22 deletions(-)

3 2

Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 28 Sep '09

28 Sep '09

Hi Peter, > Yes it is indicating so. I read through this thread I didn't find any > simple reason why it's not working. Your setup is similar with the > Pandora, i.e. McBSP functional clock is coming the codec and codec is > slave. Yes, I pretty much used the pandora as an example to write mine. > The McBSP is operating if the FSX and CLKX are toggling in this setup. > Is it so that you don't get any DMA interrupts or just few of them? I don't get any DMA int's: ~# cat /proc/interrupts CPU0 11: 0 INTC prcm 12: 0 INTC DMA 25: 0 INTC dispc 37: 5003 INTC gp timer 56: 3425 INTC i2c_omap 62: 0 INTC McBSP 63: 0 INTC McBSP > Your CLKGDV divisor value 256 means too low bit clock and sample rate > for both external 12.288 MHz and internal 96 MHz but still the DMA > should be running (if there is no bug with this divisor value). > Can you try divisor value 8 does it work then? For 48 kHz sample rate > with I2S you need a bit clock of 48 kHz*2*16 = 1.536 MHz and this you > get by dividing the 12.288 MHz with 8. > Yes, I had tried 8 before and just tried it now. Same result. It really seems like there is a DMA issue. Can you think of anything else to try before I dive into the DMA code? Rick

2 1

Re: [alsa-devel] Problems with Intel HDA driver
by Munzir Taha (منذر طه) 28 Sep '09

28 Sep '09

Hi Takashi, Sorry, the correct link to alsa-info report is http://www.alsa-project.org/db/?f=2e50beab11390b9687490feae899217710e81afa On Yaum al-Ithnain 24 Ramadan 1430 4:23:05 pm Takashi Iwai wrote: > At Mon, 14 Sep 2009 15:15:53 +0200, > > I wrote: > > At Mon, 14 Sep 2009 16:15:02 +0300, > > > > Ozan Çağlayan wrote: > > > Takashi Iwai wrote On 14-09-2009 15:36: > > > > Yeah, I think we can add EAPD handling as default for IDT92HD71xx > > > > codecs. I've never seen machines that can't live with GPIO0 bit up. > > > > > > enable_msi doesn't seem to affect the auto-muting problem. I played > > > with every single node in hda-analyzer but can't get auto-mute work. I > > > can manually mute the HP output and the internal speakers by > > > checking/unchecking OUT checkboxes on relevant NIDs. There's a 8xn > > > matrix for GPIO settings, I played with them as well to set unsol but > > > nope. > > > > > > I want to remember that dell-m4-1 makes automute work but breaks the > > > mixer channels: > > > > > > http://www.spinics.net/lists/alsa-devel/msg26714.html > > > > > > Err, I know that I didn't answer the last question in the thread as I'm > > > a little bit lost in the NID pool and didn't have time to investigate > > > further. > > > > Could you check whether this still valid? > > > > Basically the model option doesn't change the auto-mute behavior. > > It just changes which pin to be detected and which pin to mute. > > So, if model=dell-m4-1 makes the auto-muting working, it means > > that the BIOS setup is broken. > > I looked through alsa-info output now, and found the culprit. > NID 0x0f is set as a line-out, and currently the driver doesn't > don't do auto-muting for line-out unless a headphone is present. > Changing it to a headphone should fix. Or you can give hp_detect=1 > hint via sysfs and reconfigure. > > > Takashi -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

2 6

[alsa-devel] Problems with Intel HDA driver
by Andrius Štikonas 27 Sep '09

27 Sep '09

On my new laptop there were no sound with snd-hda-intel driver, so I tried to change model parameter. It seems that with model=hp-m4 my soundcard mostly works (except that main speakers still play when I plug headphones). I've attached my alsa-info.txt script. -- Andrius Štikonas

4 11

[PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model modified: Documentation/HD-Audio-Models.txt modified: pci/hda/patch_analog.c
by Miguel de Barros 27 Sep '09

27 Sep '09

Signed-off-by: Miguel de Barros <miguel.de.barros(a)bluewin.ch> diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt index abffd41..9e850ed 100644 --- a/Documentation/HD-Audio-Models.txt +++ b/Documentation/HD-Audio-Models.txt @@ -208,6 +208,7 @@ AD1884A / AD1883 / AD1984A / AD1984B laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart AD1884 ====== diff --git a/pci/hda/patch_analog.c b/pci/hda/patch_analog.c index 215e72a..7df8a08 100644 --- a/pci/hda/patch_analog.c +++ b/pci/hda/patch_analog.c @@ -4032,6 +4032,143 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1984a_touchsmart_capture_source = { + .num_items = 3, + .items = { + { "Mix", 0x3 }, /* Mixer */ + { "Mic", 0x4 }, /* port-E */ + { "Internal Mic", 0x5 }, /* Digital Mic */ + }, +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4176,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4185,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4198,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4282,22 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984a_touchsmart_capture_source; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; -- 1.6.0.4 --=-w/QltLfzo6NHdpL3XZCU--

1 0

[PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
by Miguel de Barros 27 Sep '09

27 Sep '09

Reference: ALSA bug #0004614 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614 port-A (0x11) - front hp-out port-D (0x12) - rear line out port-E (0x1c) - front mic-in port-F (0x16) - Internal speakers digital-mic (0x17) - Internal mic init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware Signed-off-by: Miguel de Barros <miguel.de.barros(a)bluewin.ch> diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt index abffd41..9e850ed 100644 --- a/Documentation/HD-Audio-Models.txt +++ b/Documentation/HD-Audio-Models.txt @@ -208,6 +208,7 @@ AD1884A / AD1883 / AD1984A / AD1984B laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart AD1884 ====== diff --git a/pci/hda/patch_analog.c b/pci/hda/patch_analog.c index 215e72a..2d603f6 100644 --- a/pci/hda/patch_analog.c +++ b/pci/hda/patch_analog.c @@ -4032,6 +4032,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { } /* end */ +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4160,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; -- 1.6.0.4 --=-J56U1/hvhPtbcxv8DR5c--

1 0

Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 26 Sep '09

26 Sep '09

Hi Peter, I forgot to tell you that if I deliberately misconfigure the McBXP_CLKS pin so that it's an output, the FSX -> DACLRC/ADCLRC signal stops. This should indicate that the CLKOUT signal is actually making it to the CLKS pin on the OMAP. Right? Rick

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#527074: /etc/init.d/alsa-utils: `/etc/init.d/alsa-utils stop` triggers amixer error
by Elimar Riesebieter 26 Sep '09

26 Sep '09

* Piotr Lewandowski [090925 20:54 +0200] > Hi alsa-devel@! > > One of the maintainers of alsa-utils package in Debian advised me to > ask for your help. I've encountered following problem: > > #v+ > $ amixer -c 0 -q set Mic 0% mute > amixer: Invalid command! > #v- > > This command is invoked from /etc/init.d/alsa-utils script during > system shutdown. Do you have any ideas how to fix that? > > Here is some basic info about my hardware: > > #v+ > $ lspci -v | sed -ne '/Audio/,/^$/p' > 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High > Definition Audio Controller (rev 01) > Subsystem: Hewlett-Packard Company Device 30a2 > Flags: bus master, fast devsel, latency 0, IRQ 16 > Memory at f4580000 (64-bit, non-prefetchable) [size=16K] > Capabilities: <access denied> > Kernel driver in use: HDA Intel > > $ cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.18a. 1. It would be an option to insert: 'echo option "snd-hda-intel enable_msi=1"' >> /etc/modprobe.d/local.conf 2. It is a good idea to update to alsa-source 1.0.21-2 and build biuld your driver up from there. Elimar -- Never make anything simple and efficient when a way can be found to make it complex and wonderful ;-)

1 0

[alsa-devel] [PATCH] [PATCH] [ASOC] pxa-ssp increase max_channels to 8
by Graeme Gregory 25 Sep '09

25 Sep '09

When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. Signed-off-by: Graeme Gregory <dp(a)xora.org.uk> --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e8a5629 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -767,13 +767,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -787,13 +787,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -808,13 +808,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -829,13 +829,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- 1.6.3.3

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Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 25 Sep '09

25 Sep '09

Hi Peter, Thanks very much for the help. > I think the problem is here. > The MCBSP_CLKS pin is connected to CLK256FS pin of TWL4030 (or TPA). > Since you are not using the TWL codec, there will be no clock on the MCBSP_CLKS > pin, thus no functional clock for McBSP, thus no bits are shifted in or out. > You can try to use the internal functional clock: > > ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, 0, > SND_SOC_CLOCK_IN); I have checked the CLKOUT (WM8731) -> McBSP_CLKS (OMAP3) pin and it has a 12288000 clock on it. I've tried using the internal clock as you show above, just to see if I get anything out on DX but I get the same thing: DX just goes low for the "aplay" duration. Any idea why I'm not getting DMA interrupts? This would indeed cause no data at DX. omap2_dma_irq_handler() never gets called but omap_init_dma() and omap_start_dma() get called. Do you think there may be any issues with my DMA code (specifically arch/arm/plat-omap/dma.c). As I mentioned, I'm at "2.6.29-rc3-omap1" Thanks, Rick

1 0

[alsa-devel] [PATCH] [ASOC] pxa-ssp increase max_channels to 8
by Graeme Gregory 25 Sep '09

25 Sep '09

When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e8a5629 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -767,13 +767,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -787,13 +787,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -808,13 +808,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -829,13 +829,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- 1.6.3.3

1 0

[alsa-devel] Mic stops working after installation (ALC861)
by remrot＠gmail.com 25 Sep '09

25 Sep '09

/* Sorry for posting this to the devel list, but the problem seems to be trickier than I first thought * * and I don't have much time left to fix it (not my machine)! Thanks... */ Hi! I've got a very strange problem with a ALC861 soundchip on a friend's Toshiba A100-507 Laptop. Sound output and input work perfectly within a live session using the Ubuntu Jaunty and Karmic (Alpha6) CDs but immediately stops working after installing the system. Currently I've installed the latest Karmic containing kernel 2.6.31 and ALSA 1.0.20. I've attached the output of the alsa-info script as well as the output of 'pacmd list' for both systems (live and installed). Right after the installation the sound stopped working and I couldn't record anything. Alsamixer showed only output devices, no capture devices at all (everything has been correct within the live session). In 'dmesg' I found the following error messages: [ 279.924019] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x301f000a [ 280.928019] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x301f000a After doing some research I found this guide (thx Mr. Iwai): http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… And indeed, adding 'probe_mask=8' fixed the sound output and brought back the capture devices (remember that they were there within the live session without that option). I've also tried the 'model=toshiba' option (combined with 'probe_mask' and standalone) but that didn't help at all (no capture devices and no sound). The current situation: I've already posted to the PulseAudio mailing ist and got some help from Mr. Colin Guthrie but we haven't found a solution (everything seems OK regarding PA). I still can't record anything within the installed system. I tried audacity, alsarec and the gnome-sound-recorder without luck. When recording in audacity, the timeline cursor doesn't move, it only flickers. In gnome-recorder, recording stops right after it has started and the application hangs on exit. PA's volume meter doesn't show anything either, it is empty. I compared the configurations on both systems but I see no differences that are not related to the 'probe_mask' option... I really believe that there is a solution because everything works in a live session. Can anybody help me with this? Regards JK

1 1

[alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 25 Sep '09

25 Sep '09

Hi, I'm working on getting a WM8731 (in SLAVE) working using I2S on omap3 McBSP2 (in MASTER). Here is the hardware: http://www.efn.org/~rick/pub/wm8731.jpg Here are the PADCONF's and the pins they hook to on the WM8731: /*Audio Interface for csb740 */\ MUX_VAL(CP(McBSP2_FSX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_FSX -> I2S_FRM -> LRCLK -> DACLRC and ADCLRC*/\ MUX_VAL(CP(McBSP2_CLKX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_CLKX -> I2S_CLK -> I2S_BCLK -> BCLK */\ MUX_VAL(CP(McBSP2_DR), (OFF_IN_PD | IEN | PTD | DIS | M0)) /* McBSP2_DR <- I2S_RXD <- ACDDAT */\ MUX_VAL(CP(McBSP2_DX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_DX -> I2S_TXD -> DACDAT */\ MUX_VAL(CP(McBSP_CLKS), (OFF_IN_PD | IEN | PTD | DIS | M0)) /* McBSP_CLKS <- I2S_MCLK <- CLKOUT */\ My kernel is "2.6.29-rc3-omap1", here's what I see at bootup: 1 WM8731 Audio Codec 0.13<6>asoc: WM8731 <-> omap-mcbsp-dai-0 mapping ok 2 ALSA device list: 3 #0: omap3csb740 (WM8731) ... 4 beagle:~# aplay test48000.wav 5 Playing WAVE 'test48000.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo 6 aplay: pcm_write:1269: write error: Input/output error I get no audio output, other than a "click" when the 8731 is initialized upon boot. When I look at the signals the McBSP2_DX line just goes low for the duration of the "aplay", it does not toggle. The other lines look fine. The file is at http://www.efn.org/~rick/pub/omap3csb740.c but roughly, here's what I'm doing: static struct snd_soc_dai_link omap3csb740_dai = { ... .cpu_dai = &omap_mcbsp_dai[0], /* use 1st one */ static int __init omap3csb740_soc_init(void) ... *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ *(unsigned int *)omap3csb740_dai.cpu_dai->private_data = 1; /* McBSP2 debug only? */ static int omap3csb740_hw_params(struct snd_pcm_substream *substream, ... ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, SND_SOC_CLOCK_IN); ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, SND_SOC_CLOCK_IN); ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 256); Any help is greatly appreciated. Rick Bronson

3 7

[alsa-devel] Exposure of the input devices of Conexant CX20561 amplifier
by Matthias Bläsing 24 Sep '09

24 Sep '09

Hello, it would be nice if I could control the input channels of my lenovo laptop (Lenovo R500). These are configured for microphone input, but the amplifier can be deactivated, so that it can work with the higher line-in level. The resulting quality is not good, but at least usable. The volume control is joint, but the amplifier control are seperated. With the HDAAnalyser I could find the Nodes (0x17 (rear) and 0x18 (front)) can be used to control the amplification. The setting can be changed between 0 and 4, where 0 makes it possible to use the input for line-in, while 4 is better suited for microphone input. I'm not sure whether it would be better to expose this as a stepped control or just a switch (amplifier on/off). These are the loaded modules (from kernel 2.6.31): snd_usb_audio 74152 3 snd_usb_lib 13904 1 snd_usb_audio snd_hda_codec_conexant 17684 1 snd_seq_dummy 2748 0 snd_seq_oss 24620 0 snd_hda_intel 24064 9 snd_hda_codec 65724 2 snd_hda_codec_conexant,snd_hda_intel snd_seq_midi 6232 0 snd_hwdep 6772 3 snd_usb_audio,snd_hda_codec snd_rawmidi 19600 2 snd_usb_lib,snd_seq_midi snd_seq_midi_event 6580 2 snd_seq_oss,snd_seq_midi snd_pcm_oss 33516 0 snd_mixer_oss 13416 1 snd_pcm_oss snd_seq 44744 7 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_pcm 64872 6 snd_usb_audio,snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 18448 2 snd_seq,snd_pcm snd_seq_device 6424 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd_page_alloc 8556 2 snd_hda_intel,snd_pcm snd 52452 35 snd_usb_audio,snd_hda_codec_conexant,snd_seq_oss,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_rawmidi,snd_pcm_oss,snd_mixer_oss,snd_seq,snd_pcm,snd_timer,snd_seq_device soundcore 6876 1 snd I attached the /proc/asound/card0/codec#0 info to this email. If you need further information - please ask. Matthias

2 2

[alsa-devel] [PATCH 1/2] ASoC: DaVinci: Fix divide by zero error during 1st execution
by Troy Kisky 24 Sep '09

24 Sep '09

When both playback and capture stream were open davinci_i2s_hw_params was setting parameters for the wrong stream. The fix for davinci_i2s_hw_params is sufficient, but it looks like a race still happens in davici_pcm_open. This patch also makes the race smaller but the next patch provides a better fix. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 3 ++- sound/soc/davinci/davinci-pcm.c | 12 +++++------- 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c54..d32e197 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -353,8 +353,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct davinci_pcm_dma_params *dma_params = dai->dma_data; struct davinci_mcbsp_dev *dev = dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 091dacb..002808b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; struct edmacc_param p_ram; int ret; - if (!dma_data) - return -ENODEV; - - prtd->params = dma_data; - /* Request master DMA channel */ ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, @@ -244,6 +237,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *params = rtd->dai->cpu_dai->dma_data; + if (!params) + return -ENODEV; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ @@ -257,6 +254,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; spin_lock_init(&prtd->lock); + prtd->params = params; runtime->private_data = prtd; -- 1.5.6.3

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[alsa-devel] AIC23 playback volume control not working
by Aggarwal, Anuj 24 Sep '09

24 Sep '09

Hi, I am trying to change the playback volume using amixer commands but am not able to see any effect. Register writes are happening correctly but no change in volume. Here is a short log: root@omap3evm:~# amixer cset numid=1 0 tlv320aic23_write: reg=0x2, value=0x0 data[0]=0x4, data[1]=0x0 tlv320aic23_write: reg=0x3, value=0x0 data[0]=0x6, data[1]=0x0 numid=1,iface=MIXER,name='Digital Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=127,step=0 : values=0,0 | dBscale-min=-121.00dB,step=1.00dB,mute=0 root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# amixer cset numid=1 127 tlv320aic23_write: reg=0x2, value=0x7f data[0]=0x4, data[1]=0x7f tlv320aic23_write: reg=0x3, value=0x7f data[0]=0x6, data[1]=0x7f numid=1,iface=MIXER,name='Digital Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=127,step=0 : values=127,127 | dBscale-min=-121.00dB,step=1.00dB,mute=0 root@omap3evm:~# root@omap3evm:~# Any suggestions are appreciated. Regards, Anuj Aggarwal

1 0

[alsa-devel] Capture not working in MMAP mode with '-v'
by Aggarwal, Anuj 24 Sep '09

24 Sep '09

Hi, When I try to capture audio in MMAP mode using the following command, it works properly: arecord -f cd -d 10 -M rec01.wav However, if I specify '-v' along with the above command, it fails to capture anything and prints log and returns immediately: Recording WAVE 'rec03.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'omap3evm' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 1 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 appl_ptr : 0 hw_ptr : 0 mmap_area[0] = 0x4030e000,0,32 (16) mmap_area[1] = 0x4030e000,16,32 (16) I have not observed this problem while playing audio, only while capture this issue crops up. Any pointers? Thanks & Regards, Anuj Aggarwal

1 0

Re: [alsa-devel] [PATCH - hda-intel conexant benq r55 laptop 1/1] [ALSA] CD-audio sound for hda-intel conexant benq laptop
by Takashi Iwai 24 Sep '09

24 Sep '09

At Tue, 22 Sep 2009 21:42:40 +0200, nowymarluk(a)wp.pl wrote: > > From: edoiks <edoiks(a)edoiks-laptop.(none)> > > After puting a cd-audio inside my laptop there was no sound out here, so I decided to install alsa-driver with debug level and setup a model=test, it didn't help, but then I look at source code and added this few lines, now cd-audio is working both when playback/recording. > > Signed-off-by: Lukasz Marcinowski <nowymarluk(a)wp.pl> > Signed-off-by: edoiks <edoiks(a)edoiks-laptop.(none)> Judging from your mail address, I stripped this edoiks line. Also, I fixed some strings for mixer elements and capture source items to follow the standard rule. E.g. "CD Audio xxx" is "CD xxx", and remove trailing spaces from the string. Now applied to sound git tree, and will be eventually merged to the upstream, too. thanks, Takashi > diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c > index 9d899ed..9afacb6 100644 > --- a/pci/hda/patch_conexant.c > +++ b/pci/hda/patch_conexant.c > @@ -682,11 +682,13 @@ static struct hda_input_mux cxt5045_capture_source = { > }; > > static struct hda_input_mux cxt5045_capture_source_benq = { > - .num_items = 3, > + .num_items = 5, > .items = { > { "IntMic", 0x1 }, > { "ExtMic", 0x2 }, > { "LineIn", 0x3 }, > + { "CD Pin", 0x4 }, > + { "Mixer ", 0x0 }, > } > }; > > @@ -811,11 +813,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = { > }; > > static struct snd_kcontrol_new cxt5045_benq_mixers[] = { > + HDA_CODEC_VOLUME("CD Audio Capture Volume", 0x1a, 0x04, HDA_INPUT), > + HDA_CODEC_MUTE("CD Audio Capture Switch", 0x1a, 0x04, HDA_INPUT), > + HDA_CODEC_VOLUME("CD Audio Playback Volume", 0x17, 0x4, HDA_INPUT), > + HDA_CODEC_MUTE("CD Audio Playback Switch", 0x17, 0x4, HDA_INPUT), > + > HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), > HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), > HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), > HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), > > + HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), > + HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), > + > {} > }; > > -- > 1.6.0.4 >

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 24 Sep '09

24 Sep '09

The following changes since commit 877ae70763fa0b19ffbe25e7e119fc96f52fbbd3: Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo 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 Chaithrika U S (1): ASoC: DaVinci: Correct McASP FIFO initialization Cliff Cai (1): ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G Troy Kisky (2): ASoC: DaVinci: Fix divide by zero error during 1st execution ASoC: Davinci: Fix race with cpu_dai->dma_data sound/soc/blackfin/bf5xx-i2s.c | 8 ++-- sound/soc/blackfin/bf5xx-tdm.c | 8 ++-- sound/soc/davinci/davinci-i2s.c | 37 ++++++----------- sound/soc/davinci/davinci-mcasp.c | 80 +++++++++++++------------------------ sound/soc/davinci/davinci-mcasp.h | 7 +++- sound/soc/davinci/davinci-pcm.c | 13 +++--- sound/soc/davinci/davinci-pcm.h | 1 - 7 files changed, 60 insertions(+), 94 deletions(-)

2 1

[alsa-devel] Audio problems with AD1981A (Subsystem: 0e11:00ad)
by Simos Xenitellis 24 Sep '09

24 Sep '09

Hi All, I am helping a user with the following hardware, http://www.alsa-project.org/db/?f=7f926200ee31b5b22222336b9e25bb0e60671a90 (Desktop computer Compaq Evo D510 SFF, with onboard sound card) lspci -vvc shows: 00:1f.5 0401: 8086:24c5 (rev 01) Subsystem: 0e11:00ad Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 17 Region 0: I/O ports at 2000 [size=256] Region 1: I/O ports at 2400 [size=64] Region 2: Memory at fc480400 (32-bit, non-prefetchable) [size=512] Region 3: Memory at fc480600 (32-bit, non-prefetchable) [size=256] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: Intel ICH Kernel modules: snd-intel8x0 What does not work: 1. Jack sense. Audio plays on onboard speakers (the computer has in-built speakers) but does not work with headset. 2. Recording. Only a 'Capture' control is shown in the recording tab. No other controls for recording are available. When trying to record with gnome-sound-recorder, the application hangs, which then kills pulseaudio. Cheers, Simos

2 1

[alsa-devel] Is there a way to initialize mixer elements?
by Ozan Çağlayan 24 Sep '09

24 Sep '09

Hi, Sometimes specific hardwares needs special tweaks (e.g. a mixer element is muted by default which avoids getting sound from HP, or the master volume's gain is very low, etc.). Do we have some kind of quirks (kernel-space/user-space) for that situations? As a real-world example, I have a user with a VIA VT1708B codec which is not able to get sound from HP unless he mutes the 'Independent HP' control: http://www.alsa-project.org/db/?f=4e7672aee15a64f67286a951e75e4168c6cd43d4 Thanks, Ozan Caglayan

2 1

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 23 Sep '09

23 Sep '09

The following changes since commit 9f072b7b2260a588bfb5fb43418c3358104e7c13: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Barry Song (2): ASoC: some minor changes for AD1836 and AD1938 codec drivers ASoC: Blackfin I2S: use dai state rather than local counter Chaithrika U S (1): ASoC: DaVinci: Correct McASP FIFO initialization Cliff Cai (1): ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G Mark Brown (9): ASoC: Update WM8711 to driver model registration method Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into upstream/wm8711 ASoC: Factor out WM8711 cache I/O ASoC: Add SPI support to WM8711 ASoC: WM8711 minor cleanups ASoC: Add TLV information to WM8711 ASoC: Convert soc-cache to use C99 style initialisers for the table Merge branch 'upstream/wm8711' into for-2.6.33 Merge branch 'for-2.6.32' into for-2.6.33 Mike Arthur (1): ASoC: Add WM8711 CODEC driver Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo Troy Kisky (2): ASoC: DaVinci: Fix divide by zero error during 1st execution ASoC: Davinci: Fix race with cpu_dai->dma_data jassi brar (1): ASoC: Support WM8580 based audio subsystem on SMDK64xx machines sound/soc/blackfin/bf5xx-i2s.c | 23 +- sound/soc/blackfin/bf5xx-tdm.c | 8 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 3 +- sound/soc/codecs/ad1938.c | 2 +- sound/soc/codecs/wm8711.c | 658 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8711.h | 42 +++ sound/soc/codecs/wm8753.c | 1 + sound/soc/davinci/davinci-i2s.c | 37 +-- sound/soc/davinci/davinci-mcasp.c | 80 ++--- sound/soc/davinci/davinci-mcasp.h | 7 +- sound/soc/davinci/davinci-pcm.c | 13 +- sound/soc/davinci/davinci-pcm.h | 1 - sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 273 +++++++++++++++ sound/soc/soc-cache.c | 18 +- 18 files changed, 1068 insertions(+), 115 deletions(-) create mode 100644 sound/soc/codecs/wm8711.c create mode 100644 sound/soc/codecs/wm8711.h create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c

1 0

[alsa-devel] [PATCH] ASoC: DaVinci: Correct McASP FIFO initialization
by Chaithrika U S 23 Sep '09

23 Sep '09

McASP write FIFO registers should be modified for playback and read FIFO registers for capture. Check the PCM mode before manipulating the FIFO registers. Currently, irrespective of playback/capture both the FIFOs are enabled or disbaled. This resulted in errors in audio loopback mode. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 36 ++++++++++++++++++------------------ 1 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 7a06c0a..613fd98 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -386,17 +386,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_start_tx(dev); - else + } else { + if (dev->rxnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_start_rx(dev); - - /* enable FIFO */ - if (dev->txnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -413,17 +413,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_stop_tx(dev); - else + } else { + if (dev->rxnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_stop_rx(dev); - - /* disable FIFO */ - if (dev->txnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, -- 1.5.6

2 1

Re: [alsa-devel] sound issues on an HP Pavilion DV3-1075 (solved)
by Liviu Andronic 23 Sep '09

23 Sep '09

On 9/23/09, Daniel Chen <seven.steps(a)gmail.com> wrote: > Yes, both of these options are necessary. What's your lspci -nv|grep -A1 0403? > debian-liv:/home/liviu# lspci -nv|grep -A1 0403 00:14.2 0403: 1002:4383 Subsystem: 103c:1506 -- 01:05.1 0403: 1002:960f Subsystem: 1002:960f Best Liviu

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[alsa-devel] [PATCH 1/2] ASoC: Blackfin I2S: use dai state rather than local counter
by Mike Frysinger 23 Sep '09

23 Sep '09

From: Barry Song <barry.song(a)analog.com> Since the active field of the dai already tells us the stream activity, the local counter variable is redundant and can be replaced. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-i2s.c | 15 +-------------- 1 files changed, 1 insertions(+), 14 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 876abad..66e7333 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -49,7 +49,6 @@ struct bf5xx_i2s_port { u16 rcr1; u16 tcr2; u16 rcr2; - int counter; int configured; }; @@ -133,16 +132,6 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - pr_debug("%s enter\n", __func__); - - /*this counter is used for counting how many pcm streams are opened*/ - bf5xx_i2s.counter++; - return 0; -} - static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -201,9 +190,8 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); - bf5xx_i2s.counter--; /* No active stream, SPORT is allowed to be configured again. */ - if (!bf5xx_i2s.counter) + if (!dai->active) bf5xx_i2s.configured = 0; } @@ -288,7 +276,6 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, -- 1.6.5.rc1

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[alsa-devel] sound issues on an HP Pavilion DV3-1075 (solved)
by Liviu Andronic 23 Sep '09

23 Sep '09

Dear ALSA devels I am writing on an ALSA-related issue that I managed to solve, on Debian testing with kernel 2.6.30 and ALSA 1.0.21. For background and exact symptoms please see these two threads. [1] [2] The first issue was that there was no sound. I solved it by manually configuring it as follows: debian-liv:/home/liviu# cat /etc/modprobe.d/sound ## ALSA portion alias char-major-116 snd alias snd-card-0 snd-hda-intel alias snd-card-1 snd-usb-audio ## module options should go here options snd-hda-intel index=0 model=hp-dv5 enable_msi=1 options snd-usb-audio index=1 The second issue concerned the external speakers/headphones: when such devices were plugged in, the internal speakers were not muted and the audio stream was not redirected to the external headset (that is, there was no sound in the headphones); although I didn't check, the microphone was probably not working, too. The issue was solved by adding position-fix=1. Thus, the following line was used options snd-hda-intel index=0 model=hp-dv5 enable-msi=1 position-fix=1 I have similar issues on Ubuntu Jackalope, and it seems that on Ubuntu it is quite often that audio cards on HP Pavilions misbehave by default (no sound, no sound in headphones, etc.). Best regards Liviu [1] http://www.mail-archive.com/debian-user@lists.debian.org/msg555984.html [2] http://www.pubbs.net/debian/200909/102780/ ### Debug messages ### liviu@debian-liv:~$ cat /proc/asound/card*/codec* | grep Codec Codec: IDT 92HD71B7X Codec: ATI RS690/780 HDMI liviu@debian-liv:~$ cat /proc/asound/cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xd2400000 irq 16 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xd2310000 irq 29 liviu@debian-liv:~$ cat /proc/asound/devices 0: [ 0] : control 1: : sequencer 4: [ 0- 0]: hardware dependent 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 32: [ 1] : control 33: : timer 36: [ 1- 0]: hardware dependent 51: [ 1- 3]: digital audio playback liviu@debian-liv:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 liviu@debian-liv:~$ aplay -L default:CARD=SB HDA ATI SB, STAC92xx Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) hdmi:CARD=HDMI HDA ATI HDMI, ATI HDMI HDMI Audio Output -Computer- Processor : 2x AMD Turion(tm) X2 Dual-Core Mobile RM-72 Memory : 3802MB (907MB used) Operating System : Debian GNU/Linux squeeze/sid User Name : liviu (Liviu) Date/Time : Tue 22 Sep 2009 12:58:23 BST -Display- Resolution : 1280x800 pixels OpenGL Renderer : ATI Radeon HD 3200 Graphics X11 Vendor : The X.Org Foundation -Multimedia- Audio Adapter : HDA-Intel - HDA ATI SB Audio Adapter : HDA-Intel - HDA ATI HDMI -Input Devices- Macintosh mouse button emulation AT Translated Set 2 keyboard Video Bus Power Button Power Button Lid Switch PC Speaker ST LIS3LV02DL Accelerometer HP Webcam SynPS/2 Synaptics TouchPad HDA Digital PCBeep HDA ATI SB Mic at Ext Front Jack HDA ATI SB HP Out at Ext Front Jack -Printers (CUPS)- psc_1200_first -Version- Kernel : Linux 2.6.30-1-amd64 (x86_64) Compiled : #1 SMP Sat Aug 15 18:09:19 UTC 2009 C Library : GNU C Library version 2.9 (stable) Default C Compiler : GNU C Compiler version 4.3.4 (Debian 4.3.4-2) Distribution : Debian GNU/Linux squeeze/sid -Current Session- Computer Name : debian-liv User Name : liviu (Liviu) Home Directory : /home/liviu Desktop Environment : XFCE 4 -Misc- Uptime : 2 hours, 19 minutes Load Average : 0.57, 0.67, 0.73 liviu@debian-liv:~$ lspci -vv 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) Subsystem: Hewlett-Packard Company Device 1506 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Interrupt: pin ? routed to IRQ 16 Region 0: Memory at d2400000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel [..] 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller Subsystem: ATI Technologies Inc RS780 Azalia controller Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 19 Region 0: Memory at d2310000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel liviu@debian-liv:~$ lsmod | grep -i snd snd_hda_codec_atihdmi 3952 1 snd_hda_codec_idt 62480 1 snd_hda_intel 26680 9 snd_hda_codec 75248 3 snd_hda_codec_atihdmi,snd_hda_codec_idt,snd_hda_intel snd_hwdep 8152 1 snd_hda_codec snd_pcm_oss 37200 0 snd_mixer_oss 15072 2 snd_pcm_oss snd_pcm 78504 4 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_midi 6976 0 snd_rawmidi 23008 1 snd_seq_midi snd_seq_midi_event 7712 1 snd_seq_midi snd_seq 51392 2 snd_seq_midi,snd_seq_midi_event snd_timer 21824 3 snd_pcm,snd_seq snd_seq_device 7476 3 snd_seq_midi,snd_rawmidi,snd_seq snd 63912 25 snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 7984 2 snd snd_page_alloc 10000 2 snd_hda_intel,snd_pcm -- Do you know how to read? http://www.alienetworks.com/srtest.cfm Do you know how to write? http://garbl.home.comcast.net/~garbl/stylemanual/e.htm#e-mail -- Do you know how to read? http://www.alienetworks.com/srtest.cfm Do you know how to write? http://garbl.home.comcast.net/~garbl/stylemanual/e.htm#e-mail

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[alsa-devel] [PATCH 1/4] ASoC: ALSA ASoC CQ0093 Voice Codec Driver
by miguel.aguilar＠ridgerun.com 23 Sep '09

23 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds support for the Voice Codec CQ0093. This driver was tested on a DM355 EVM rev C. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 377 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 23 +++ 4 files changed, 406 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93..55d1287 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_CQ0093VC if ARCH_DAVINCI_DM365 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_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28..a3e713c 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-cq93vc-objs := cq93vc.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-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(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/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..e64af25 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,377 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * 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/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +#define AUDIO_NAME "cq0093" +#define CQ93VC_VERSION "0.1" + +/* + * Voice codec registers + * + * The base address for the registers used by this driver is shifted + * 0x80 from the original Voice Codec base adress since the first region + * of the Voice Codec registers is used by the Voice Codec Interface driver. + */ +#define VC_REG05 0x14 /* PGA and MIC Gain Control */ +#define VC_REG09 0x24 /* Digital Soft Mute/Attenuation Control */ +#define VC_REG12 0x30 /* Voice Codec Up/Down */ + +/* Registers bits */ +#define PGA_GAIN 0x07 +#define DIG_ATTEN 0x3F +#define MUTE 0x40 +#define POWER_ALL_ON 0xFD + +/* Codec private data */ +struct cq93vc_priv { + void __iomem *base; + unsigned int sysclk; + resource_size_t pbase; + size_t base_size; +}; + +/* + * Read to the cq0093 register space + */ +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct cq93vc_priv *cq93vc = codec->private_data; + + return __raw_readb(cq93vc->base + reg); +} + +/* + * Write to the cq0093 register space + */ +static int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct cq93vc_priv *cq93vc = codec->private_data; + + __raw_writeb(value, cq93vc->base + reg); + return 0; +} + +static int cq93vc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, VC_REG09) & ~MUTE; + + if (mute) + cq93vc_write(codec, VC_REG09, reg | MUTE); + else + cq93vc_write(codec, VC_REG09, reg); + + return 0; +} + +static int cq93vc_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(cq93vc_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&cq93vc_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cq93vc_priv *cq93vc = codec->private_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + cq93vc->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* all power is driven by DAPM system */ + cq93vc_write(codec, VC_REG12, POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* + * all power is driven by DAPM system, + * so output power is safe if bypass was set + */ + cq93vc_write(codec, VC_REG12, 0x0); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, VC_REG12, 0x0); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .hw_params = cq93vc_hw_params, + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "cq93vc", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_device *cq93vc_socdev; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct cq93vc_priv *cq93vc; + struct snd_soc_codec *codec; + struct resource *res, *mem; + int ret; + + dev_info(dev, "CQ0093 Voice Codec %s\n", CQ93VC_VERSION); + + cq93vc = kzalloc(sizeof(struct cq93vc_priv), GFP_KERNEL); + if (cq93vc == NULL) { + dev_dbg(dev, "%s: could not allocate memory for private data\n", + pdev->name); + return -ENOMEM; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(dev, "%s: could not allocate memory for codec data\n", + pdev->name); + ret = -ENOMEM; + goto fail1; + } + + res = cq93vc_resources; + + cq93vc->pbase = res->start; + cq93vc->base_size = resource_size(res); + + mem = request_mem_region(cq93vc->pbase, + cq93vc->base_size, "cq93vc"); + if (!mem) { + dev_dbg(dev,"%s: voice codec registers at %08x are not free\n", + pdev->name, cq93vc->pbase); + ret = -EBUSY; + goto fail2; + } + + cq93vc->base = ioremap(cq93vc->pbase, cq93vc->base_size); + if (cq93vc->base == NULL) { + dev_dbg(dev,"%s: can't ioremap mem resource.\n", + pdev->name); + ret = -ENOMEM; + goto fail3; + } + + mutex_init(&codec->mutex); + codec->private_data = cq93vc; + codec->name = "cq93vc"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + socdev->card->codec = codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + goto fail4; + } + + /* Set the PGA Gain to 18 dB */ + cq93vc_write(codec, VC_REG05, PGA_GAIN); + + /* Set the DAC digital attenuation to 0 dB */ + cq93vc_write(codec, VC_REG09, DIG_ATTEN); + + /* Set controls */ + cq93vc_add_controls(codec); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Register sound card */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(dev, "%s: failed to register card\n", pdev->name); + goto fail5; + } + + cq93vc_socdev = socdev; + + return 0; +fail5: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +fail4: + iounmap(cq93vc->base); +fail3: + release_mem_region(cq93vc->pbase, cq93vc->base_size); +fail2: + kfree(codec); +fail1: + kfree(cq93vc); + + return ret; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct cq93vc_priv *cq93vc = codec->private_data; + + /* Power down chip */ + if (codec->control_data) + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + iounmap(cq93vc->base); + release_mem_region(cq93vc->pbase, cq93vc->base_size); + + kfree(codec); + kfree(cq93vc); + + cq93vc_socdev = NULL; + + snd_soc_unregister_dai(&cq93vc_dai); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .suspend = cq93vc_suspend, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_init(void) +{ + return snd_soc_register_dai(&cq93vc_dai); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + snd_soc_unregister_dai(&cq93vc_dai); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..8d37763 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,23 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * 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 _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; +extern struct resource cq93vc_resources[]; + +#endif -- 1.6.0.4

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[alsa-devel] [PATCH 4/4] DaVinci: DM365: Add voice codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 23 Sep '09

23 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch enables the voice codec support for the DM365 EVM. For the DM365 EVM only one codec is able to be enabled at time, external (AIC3101) or internal (Voice Codec CQ0093). Either internal or external codec can be selected in the configuration menu. This patch was created based on the temp/asoc branch of the DaVinci git and it assumes that the patch "[PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM" is applied. This patch was tested on the DM365 EVM rev C. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 +++ sound/soc/davinci/davinci-evm.c | 42 +++++++++++++++++++++++++++++- 2 files changed, 44 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 77f815c..083b8d7 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -482,7 +482,11 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) dm365_init_asp(&dm365_evm_snd_data); +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + dm365_init_vcif(&dm365_evm_snd_data); +#endif } static __init void dm365_evm_irq_init(void) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..e826a80 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,9 +28,11 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/cq93vc.h" #include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-vcif.h" #include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -151,6 +153,15 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +#ifdef CONFIG_SND_DM365_INTERNAL_CODEC +static struct snd_soc_dai_link cq93vc_evm_dai = { + .name = "CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, +}; +#endif + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -177,7 +188,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 or dm365 evm audio machine driver */ +/* davinci dm6446, dm355 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -185,6 +196,19 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver external or internal */ +static struct snd_soc_card dm365_snd_soc_card_evm = { +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) + .name = "DaVinci DM365 EVM", + .dai_link = &evm_dai, +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + .name = "On-chip Voice Codec", + .dai_link = &cq93vc_evm_dai, +#endif + .platform = &davinci_soc_platform, + .num_links = 1, +}; + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +241,17 @@ 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, +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + .codec_dev = &soc_codec_dev_cq93vc, +#endif +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -244,12 +279,15 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { + if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } 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] [PATCH 3/4] DaVinci: DM365: Add resources for voice codec and platform device for VCIF
by miguel.aguilar＠ridgerun.com 23 Sep '09

23 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Add the resources needed by the CQ0093 voice codec. 2) Add platform device for the voice codec interface. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/dm365.c | 46 ++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/dm365.h | 5 +++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 3 files changed, 53 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index 2674438..80dc35b 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -605,6 +605,8 @@ 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_VC_TX, 0, 1, 1, false) +EVT_CFG(DM365, EVT3_VC_RX, 1, 1, 1, false) EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif @@ -835,6 +837,42 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vcif_resources[] = { + /* The first are of Voice Codec registers before 0x80 belongs to VCIF */ + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + (SZ_1K>>4) - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_VC_TX, + .end = DAVINCI_DMA_VC_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_VC_RX, + .end = DAVINCI_DMA_VC_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_vcif_device = { + .name = "voice_codec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vcif_resources), + .resource = dm365_vcif_resources, +}; + +struct resource cq93vc_resources[] = { + /* The base address for the Voice Codec registers is shifted by 0x80 */ + { + .start = DAVINCI_DM365_VC_BASE + 0x80, + .end = DAVINCI_DM365_VC_BASE + 0x80 + (SZ_1K>>3) - 1, + .flags = IORESOURCE_MEM, + }, +}; +EXPORT_SYMBOL_GPL(cq93vc_resources); + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -950,6 +988,14 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) platform_device_register(&dm365_asp_device); } +void __init dm365_init_vcif(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_EVT2_VC_TX); + davinci_cfg_reg(DM365_EVT3_VC_RX); + dm365_vcif_device.dev.platform_data = pdata; + platform_device_register(&dm365_vcif_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 2291c0d..b78ec55 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -25,7 +25,12 @@ #define DM365_EMAC_MDIO_OFFSET (0x4000) #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) +#define DAVINCI_DM365_VC_BASE (0x01D0C000) +#define DAVINCI_DMA_VC_TX 2 +#define DAVINCI_DMA_VC_RX 3 + void __init dm365_init(void); void __init dm365_init_asp(struct snd_platform_data *pdata); +void __init dm365_init_vcif(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index 7732832..b01b5c4 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -311,6 +311,8 @@ enum davinci_dm365_index { DM365_INT_NSF_DISABLE, /* EDMA event muxing */ + DM365_EVT2_VC_TX, + DM365_EVT3_VC_RX, DM365_EVT2_ASP_TX, DM365_EVT3_ASP_RX, DM365_EVT26_MMC0_RX, -- 1.6.0.4

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 22 Sep '09

22 Sep '09

The following changes since commit 0c31cf3e4af79ea18bbd365b07ef0de207673894: Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration 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: some minor changes for AD1836 and AD1938 codec drivers Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo sound/soc/codecs/ad1836.c | 3 +-- sound/soc/codecs/ad1938.c | 2 +- sound/soc/codecs/wm8753.c | 1 + 3 files changed, 3 insertions(+), 3 deletions(-)

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[alsa-devel] [PATCH 0/4] Add Voice Codec support for DM365
by Miguel Aguilar 22 Sep '09

22 Sep '09

Hi, The following series of patches adds the following support: 1) CQ0093 Voice Codec driver. 2) Voice Codec Interface. 3) Voice Codec resources and Voice Codec Interface platform device. 4) Voice Codec on the DM365 EVM. Notes: The patch series was created based on the current temp/asoc branch of the DaVinci git. It assumes that the patch "[PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM" is applied. This patch series was tested on a DM365 EVM rev C. Regards, -- Miguel Angel Aguilar Ulloa Embedded Software Engineer RidgeRun Embedded Solutions miguel.aguilar(a)ridgerun.com

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[alsa-devel] (AT91RM2000 - ARM) Driver that puts data over SSC to FPGA
by Michel Müller 22 Sep '09

22 Sep '09

Hello, I am trying to write an ALSA-driver that puts the audio data over the SSC channel to the FPGA. This FPGA routes the Data to the AD/DA chips. I tried to figure out how I can do this with the ASoC Interfaces but I am currently stuck. Currently there is an Atmel SSC and PCM Interface that I can probably use. If I want to use the ASoC API I need to have a Codec Driver. But I don't have direct access to the codec. How should I proceed? I have 2 possibilities I think: Writing a Codec Driver / Writing an Alsa Driver as described in the guide of Takashi Iwai. If I write a character device that transfers data to the FPGA over the SSC I can hear sound, but I want to use ALSA because the use of audio applications will be easy and flexible. Thank you in advance! Regards, Michel

2 1

Re: [alsa-devel] Application and hw mixing for VoIP (modified)
by Takashi Iwai 22 Sep '09

22 Sep '09

At Mon, 21 Sep 2009 15:48:39 +0200, Fabien COMTE wrote: > > Hi, > I need some help please. > I have an application that does VoIP and alsa interface. > > I want to : > - send through VoIP application a mixed sound from mic and an another > application (aplay should be ok for testing). > > mic ---------- soundcard 0 -------------> > mix ---- alsa interface -----> > voip application > application 1 --- alsa interface ------> > (aplay) > > - use sound received from VoIP with HP output and an another application > (arecord should be ok for testing). > > ----- alsa interface > ------> application 2 (arecord) > voip application ----- alsa interface ----> share > ------ soundcard 0 > --------> HP output > > > And a little question : > How to define an application interface in asoundrc ? Well, this isn't so trivial as it sounds. If the application 1 (aplay) needs to be output actually, it means to split the signal to two interfaces, one to voip program, and another to the output channel. If app1 doesn't have to be heard, then you can use a loopback device (snd-aloop driver included in alsa-driver tree) and bind with the mic device using multi plugin. The latter case, split from voip to app2 and HP output, can be also done via multi plugin + aloop device. But, this kind of infrastructure could be better done with a proper daemon, such as JACK, I guess... Takashi

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[alsa-devel] alsa-lib 1.0.21a release
by Jaroslav Kysela 22 Sep '09

22 Sep '09

Hello, I made a quick release 1.0.21a of the alsa-lib package to fix regression for the configuration parsing (makes some device names unavailable). These patches were included: 19abc37 PCM - Change the hw_params determination order d1b70af Change dmix.conf to accept user configuration from defaults.dmix.<driver_id>.xxx eb9ae58 Revert "Fix driver conf parsing in snd_config_hook_load_for_all_cards()" f0a32a6 mixer: Add Speaker and Beep names to the weight list Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

5 7

[alsa-devel] (no subject)
by Delusion's Master 22 Sep '09

22 Sep '09

Hi, I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Windows Live™: Keep your life in sync. Check it out! http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009

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[alsa-devel] [PATCH] wm8753: fix mapping when MONOMIX is set to Stereo
by Phil Vandry 22 Sep '09

22 Sep '09

When MONOMIX is set to Stereo, Left PGA was not powered on but should be. Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue. Signed-off-by: Phil Vandry <vandry(a)TZoNE.ORG> --- sound/soc/codecs/wm8753.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d80d414..5ad677c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -595,6 +595,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Mono Capture mixer-mux */ {"Capture Right Mixer", "Stereo", "Capture Right Mux"}, + {"Capture Left Mixer", "Stereo", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"}, {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},

2 1

[alsa-devel] S/PDIF on NVidia CK804 not working
by Mitko Haralanov 21 Sep '09

21 Sep '09

Hi all, I am new to the list and was hoping that someone here could give me some help and advise as I am stuck and none of my research has turned anything up. I am having trouble getting any sound out of the S/PDIF port on my built-in sound card. The card is recognized and 'aplay -L' shows the iec958 device (as evident by the attached alsa-info.txt file) but when I try to play any sound through it (using aplay) I get one of two errors: 1. If I use the iec958 device directly: $ aplay -D iec958 /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono aplay: set_params:984: Channels count non available 2. If I try to use the plug:iec958: $ aplay -D plug:iec958 /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono (the sound file seems to play but there is no audio coming out of the receiver) Looking through the ALSA website, I see that my sound device is supported. Configuring /etc/asound.conf and /etc/modprobe.conf as per the ALSA website yields the same results as above. Can anyone please, help me get my S/PDIF working? Thank you very much! -- Mitko Haralanov

2 3

[alsa-devel] Application and hw mixing for VoIP (modified)
by Fabien COMTE 21 Sep '09

21 Sep '09

Hi, I need some help please. I have an application that does VoIP and alsa interface. I want to : - send through VoIP application a mixed sound from mic and an another application (aplay should be ok for testing). mic ---------- soundcard 0 -------------> mix ---- alsa interface -----> voip application application 1 --- alsa interface ------> (aplay) - use sound received from VoIP with HP output and an another application (arecord should be ok for testing). ----- alsa interface ------> application 2 (arecord) voip application ----- alsa interface ----> share ------ soundcard 0 --------> HP output And a little question : How to define an application interface in asoundrc ? I tried a lot of asoundrc but they doesn't work. Could someone help me ? Thank you. Fabien

1 0

[alsa-devel] [PATCH] ALSA: lx6464es - remove unused struct member
by Tim Blechmann 21 Sep '09

21 Sep '09

we cannot set the sampling rate of the device, but can only read it from the board, so we don't need the member for it. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.h | 1 - 1 files changed, 0 insertions(+), 1 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: lx6464es - cleanup of rmh message bus function
by Tim Blechmann 21 Sep '09

21 Sep '09

the rmh bus is not used asynchronously, so it is safe to remove the specific code pieces. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.h | 1 - sound/pci/lx6464es/lx_core.c | 98 +---------------------------------------- 2 files changed, 1 insertions(+), 98 deletions(-)

2 1

[alsa-devel] [PATCH] sb16: request_region failure ignored in snd_sb16_isa_probe1()
by Roel Kluin 21 Sep '09

21 Sep '09

request_region() may fail. clean up if it does, also a release region later on was missing. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with sed: http://kernelnewbies.org/roelkluin build, sparse and checkpatch tested. diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 519c363..f50ed6f 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -502,6 +502,11 @@ static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev) fm_port[dev] = port[dev]; /* block the 0x388 port to avoid PnP conflicts */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); + if (acard->fm_res == NULL) { + snd_card_free(card); + return -ENOMEM; + } + #ifdef SNDRV_SBAWE_EMU8000 /* non-PnP AWE port address is hardwired with base port address */ awe_port[dev] = port[dev] + 0x400; @@ -509,6 +514,7 @@ static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev) snd_card_set_dev(card, pdev); if ((err = snd_sb16_probe(card, dev)) < 0) { + release_region(0x388, 4); snd_card_free(card); return err; }

2 1

[alsa-devel] [PATCH] some minor changes for AD1836 and AD1938 codec drivers
by Barry Song 21 Sep '09

21 Sep '09

1. delete redundant assignment to bus field in spi_driver structure 2. fix lost assignment to set_bias_level entry in ad1938 codec dai 3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836" Signed-off-by: Barry Song <barry.song(a)analog.com> --- sound/soc/codecs/ad1836.c | 3 +-- sound/soc/codecs/ad1938.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 01343dc..c48485f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -251,8 +251,7 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi) static struct spi_driver ad1836_spi_driver = { .driver = { - .name = "ad1836-spi", - .bus = &spi_bus_type, + .name = "ad1836", .owner = THIS_MODULE, }, .probe = ad1836_spi_probe, diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 9a049a1..34b30ef 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -456,7 +456,6 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi) static struct spi_driver ad1938_spi_driver = { .driver = { .name = "ad1938", - .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = ad1938_spi_probe, @@ -515,6 +514,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->num_dai = 1; codec->write = ad1938_write_reg; codec->read = ad1938_read_reg_cache; + codec->set_bias_level = ad1938_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- 1.5.6.3

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[alsa-devel] Multiple, independent streams with HDA
by Stefan Bruens 20 Sep '09

20 Sep '09

Hi as HDA supports up to 16 independent streams (sample rate, resolution), is there a possibility to expose this to user space? Use cases: - Simultaneous playback of music to speaker and connecting HPs for VoIP - Multiseat - pair of I/O for each seat ... I think there is already limited support for this, as e.g. SPDIF and analog expose independent PCMS (a least on STAC9271), although it is just a single codec. Maybe even make this configurable, either at run time, or at module load time. E.g. "option hda-intel pcm_out_nids=0x2:0x3,0x4:0x1e pcm_in_nids=0x7:0x20" creates pcmC0D{0,1,2}p and pcmC0D{0,1}c, where 0p is analog stereo out, 1p is analog 4chan out, 2p digital out, 0c is analog stereo in, and finally 1c is digital stereo in. (This example is for STAC9271: cat /proc/asound/Intel/codec#2 | egrep "Node|Converter" | grep Conv -B1 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x06 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 Node 0x09 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 -- Node 0x1e [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 -- Node 0x20 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 ) Stefan -- Stefan Brüns / Bergstraße 21 / 52062 Aachen phone: +49 241 53809034 mobile: +49 151 50412019

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[alsa-devel] [[ALSA] PATCH - /pci/hda/patch_conexant.c 1/1] patch_conexant.c: cd pin added to mixer for benq r55 laptop,
by nowymarluk＠wp.pl 20 Sep '09

20 Sep '09

From: edoiks <nowymarluk(a)wp.pl> Signed-off-by: edoiks <nowymarluk(a)wp.pl> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 9d899ed..700a430 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -687,6 +687,8 @@ static struct hda_input_mux cxt5045_capture_source_benq = { { "IntMic", 0x1 }, { "ExtMic", 0x2 }, { "LineIn", 0x3 }, + { "CD Pin", 0x4 }, + { "Mixer ", 0x0 }, } }; @@ -816,6 +818,14 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), + {} }; -- 1.6.0.4

2 1

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 20 Sep '09

20 Sep '09

The following changes since commit cdc65fbe18aef15e92d2ebb410a189fbf956fb06: Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 The only really big thing here is the addition of the AK4671 driver, other than that it's minor tweaks. Barry Song (4): ASoC: Provide API for reordering channels ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836 ASoC: fix typos in Blackfin headers ASoC: Blackfin I2S: add lost platform_device parameter to resume function Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration Cliff Cai (2): ASoC: Blackfin AC97: add a few missing multichannel define handling ASoC: Blackfin I2S: fix resuming when device hasn't been used Huang Weiyi (1): ASoC: remove unused #include <linux/version.h> Jassi (3): ASoC: S3C lrsync function made to work with IRQs disabled. ASoC: S3C I2S LRCLK polarity option. ASoC: Return correct codec clock in s3c64xx-i2s Joonyoung Shim (2): ASoC: Factor out I2C 8 bit address 8 bit data I/O ASoC: AK4671: add ak4671 codec driver Julia Lawall (1): ASoC: Clean up error handling in MPC5200 DMA setup Mark Brown (5): ASoC: Add source argument to PLL configuration ASoC: Allow per-route connectedness checks for supplies ASoC: Fix display of stream name in DAPM debugfs ASoC: Add S3C64xx IIS CDCLK source selection Merge branch 'for-2.6.32' into for-2.6.33 Miguel Aguilar (1): ASoC: Davinci: Add audio codec support for DM365 EVM Mike Frysinger (1): ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 2 + include/sound/soc-dai.h | 13 +- include/sound/soc-dapm.h | 7 + sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 8 + sound/soc/blackfin/bf5xx-ac97.h | 2 +- sound/soc/blackfin/bf5xx-ad1836.c | 7 + sound/soc/blackfin/bf5xx-ad1938.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 22 +- sound/soc/blackfin/bf5xx-i2s.h | 2 +- sound/soc/blackfin/bf5xx-sport.c | 2 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 45 +- sound/soc/blackfin/bf5xx-tdm.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/ak4671.c | 825 +++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 ++++ sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 4 +- sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8900.c | 4 +- sound/soc/codecs/wm8940.c | 4 +- sound/soc/codecs/wm8960.c | 4 +- sound/soc/codecs/wm8974.c | 5 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9713.c | 4 +- sound/soc/davinci/Kconfig | 4 +- sound/soc/davinci/davinci-evm.c | 7 +- sound/soc/davinci/davinci-mcasp.c | 24 +- sound/soc/fsl/mpc5200_dma.c | 33 +- sound/soc/imx/mx27vis_wm8974.c | 2 +- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/pxa/zylonite.c | 5 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 46 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 19 +- sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + sound/soc/soc-cache.c | 30 + sound/soc/soc-core.c | 32 +- sound/soc/soc-dapm.c | 26 +- 48 files changed, 1312 insertions(+), 101 deletions(-) create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 20 Sep '09

20 Sep '09

The following changes since commit d4e54e871f4d2ca29df081abf8e0d5209d252979: Huang Weiyi (1): ASoC: remove unused #include <linux/version.h> 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 (2): ASoC: fix typos in Blackfin headers ASoC: Blackfin I2S: add lost platform_device parameter to resume function Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration Cliff Cai (2): ASoC: Blackfin AC97: add a few missing multichannel define handling ASoC: Blackfin I2S: fix resuming when device hasn't been used Mike Frysinger (1): ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long sound/soc/blackfin/bf5xx-ac97.c | 8 ++++++++ sound/soc/blackfin/bf5xx-ac97.h | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 22 +++++++++------------- sound/soc/blackfin/bf5xx-i2s.h | 2 +- sound/soc/blackfin/bf5xx-sport.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 24 ++++++++++++++++++++++-- 6 files changed, 42 insertions(+), 18 deletions(-)

2 1

[alsa-devel] [PATCH] SMDK64XX I2S: Added machine driver for WM8580
by jassi brar 20 Sep '09

20 Sep '09

New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 283 +++++++++++++++++++++++++++++++++++ 3 files changed, 293 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..769697e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,14 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..7790406 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..e4b5e0f --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,283 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(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/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_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; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. + * This criterion can't be met if we request PLL output + * as {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a minimum value that + * results in (params_rate(params) * rfs), and itself, acceptable + * to both - the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Assuming the CODEC driver evaluates it's rfs too from this call */ + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static int smdk64xx_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, 0); +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, + .hw_free = smdk64xx_hw_free, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_enable_pin(codec, "Center/Sub"); + snd_soc_dapm_enable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

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[alsa-devel] snd_device_name_hint problems
by David Henningsson 20 Sep '09

20 Sep '09

Hello, First; I'm not completely sure how much of this is your issue, and how much is my distribution's fault (Ubuntu). If you believe my problems is due to bugs in the distribution, please say so. I'm working with a small recording application and I want to enumerate possible recording sources/devices, just to put in a combobox. In my computer I have two soundcards, one on the motherboard and one PCI soundcard (ESI Juli@). I also have Pulseaudio installed, so the devices I want to show up are those three - and possibly alternatives under these, such as analog/digital inputs. According to Lennart's guide to sound API's, snd_device_name_hint is the way to go, as both snd_card_xxx and snd_config_xxx are deprecated. (Speak up if you don't agree.) So assuming I do about the same as "arecord -L" does, I get the following output (descriptions omitted) : front:CARD=Juli,DEV=0 surround40:CARD=Juli,DEV=0 surround41:CARD=Juli,DEV=0 surround50:CARD=Juli,DEV=0 surround51:CARD=Juli,DEV=0 surround71:CARD=Juli,DEV=0 iec958:CARD=Juli,DEV=0 null front:CARD=CK804,DEV=0 surround40:CARD=CK804,DEV=0 surround41:CARD=CK804,DEV=0 surround50:CARD=CK804,DEV=0 surround51:CARD=CK804,DEV=0 surround71:CARD=CK804,DEV=0 I have two problems with this: first thing is that the "pulse" device is not listed. I think that could be solved either by setting "defaults.namehint.showall" to on, or by setting the name hint of the pulse device to be included in the "basic" devices. Would you recommend doing either one of those, and if so, which one? Second problem is that I don't have surround input on either of my soundcards, so most of the device options above do not make any sense. Even if I try, I have a hard time recording from them (arecord fails to record it with various errors, such as "Sample format non available"). If I instead use "plughw:CARD=name", recording works. To sum it up, the output I would like to have, is at least these three: plughw:CARD=Juli plughw:CARD=CK804 pulse How do I/we get there? // David

2 3

Re: [alsa-devel] Application and hw mixing for VoIP
by Mark Brown 19 Sep '09

19 Sep '09

On Thu, Sep 17, 2009 at 11:51:33AM +0200, Fabien COMTE wrote: > I have an application that does VoIP and use OSS emulated interface. I suspect part of your problem here is the in-kernel OSS emulation - that bypasses all the ALSA user space mixing support. If you can't change the OSS using application to use ALSA then I believe there are some other options based on preloading a library to intercept the attempts to open the OSS device and emulate it but I'm not really familiar with them.

1 0

[alsa-devel] [PATCH] SMDK64XX I2S: Added machine driver for WM8580
by jassi brar 19 Sep '09

19 Sep '09

New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 284 +++++++++++++++++++++++++++++++++++ 3 files changed, 294 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..769697e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,14 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..7790406 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..186e1de --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,284 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(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/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_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; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* Currently, WM8580 driver doesn't support PLL-out rates + * other than those mentioned in Table-52 Page-58 of WM8580A + * manual. Some of these unavailable rates include + * {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a value that results in + * (params_rate(params) * rfs), and itself, acceptable to the both + * the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Assuming the CODEC driver evaluates it's rfs too from this call */ + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static int smdk64xx_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, 0); +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, + .hw_free = smdk64xx_hw_free, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* All enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_enable_pin(codec, "Center/Sub"); + snd_soc_dapm_enable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

2 2

[alsa-devel] [PATCH] S3C64XX I2S: Return correct codec clock
by jassi brar 18 Sep '09

18 Sep '09

From: Jassi <jassi.brar(a)samsung.com> Instead of always returnig pointer to the 'audio-bus' clock, check which clock is used to generate internal clocks and then return it's pointer. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 62a7fce..66f4ded 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -133,8 +133,12 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) { struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - return i2s->iis_cclk; + if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; } EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock); -- 1.6.2.5

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Fixes to McASP configuration
by Chaithrika U S 18 Sep '09

18 Sep '09

McASP register settings are not correct for DSP mode of operation. There is a channel swap initally. This patch provides fixes to the register values for proper working. Tested on DA830/OMAP-L137 EVM, DM6467 EVM. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 24 ++++++++++++++++++++++-- 1 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index eca22d7..7a06c0a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -512,34 +512,49 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int channel_size) { u32 fmt = 0; + u32 mask, rotate; switch (channel_size) { case DAVINCI_AUDIO_WORD_8: fmt = 0x03; + rotate = 6; + mask = 0x000000ff; break; case DAVINCI_AUDIO_WORD_12: fmt = 0x05; + rotate = 5; + mask = 0x00000fff; break; case DAVINCI_AUDIO_WORD_16: fmt = 0x07; + rotate = 4; + mask = 0x0000ffff; break; case DAVINCI_AUDIO_WORD_20: fmt = 0x09; + rotate = 3; + mask = 0x000fffff; break; case DAVINCI_AUDIO_WORD_24: fmt = 0x0B; + rotate = 2; + mask = 0x00ffffff; break; case DAVINCI_AUDIO_WORD_28: fmt = 0x0D; + rotate = 1; + mask = 0x0fffffff; break; case DAVINCI_AUDIO_WORD_32: fmt = 0x0F; + rotate = 0; + mask = 0xffffffff; break; default: @@ -550,6 +565,13 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, RXSSZ(fmt), RXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + return 0; } @@ -638,7 +660,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "playback tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); } else { /* bit stream is MSB first with no delay */ @@ -655,7 +676,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "capture tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); } } -- 1.5.6

2 4

[alsa-devel] Questions regarding workaround for VMware
by Bankim Bhavsar 18 Sep '09

18 Sep '09

Takashi I came across a workaround for VMware in ALSA kernel code http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commit;h… Thanks for the fix. In the comments you mention about inaccurate timer source. Could you elaborate on the problem? As of now we see that sound apps running in Fedora 11 guest OS with kernel 2.6.29.4-167 don't make any progress and can hang window manager like Metacity. VMware virtualizes Ensoniq ES1371 sound device. Is there something specific you would like us to fix in our virtual sound device or timer source? Thanks, Bankim.

4 6

Re: [alsa-devel] Fwd: Dell Studio 14, internal mic
by Alistair Boyle 18 Sep '09

18 Sep '09

Takashi, As mentioned below I have a Dell Studio 14, with a two channel mic built into the screen, speakers, and two headphone jacks and a mic jack on the side. I've attached an updated alsa-info.sh output for 1.0.21. Things are working significantly better now (vs. 1.0.20) but a little messed up in the mixer controls. (When the module loads it says its making a guess for the chip.) For the playback settings, "headphone" controls both headphones (should only set one of the headphone jacks), "headphone as lineout" sets the front-most headphone jack as lineout and disables the speaker off feature when something is plugged into the front-most jack (good, unless this is supposed to be available/work for both headphone jacks?), "headphone 1" does nothing (bad, should be other headphone control?), "front" sets the built-in speaker volume as expected, "pc beep" makes a lot of crackling on the speaker when a "ding" is hit on a terminal and so is muted (maybe not alsa's fault?) Plugging in the headphones (with "headphone as lineout" disabled) now mutes the "front" speakers as expected. For the capture settings, "digital" should be the first setting since its the master capture control, "front mic" controls the stereo built-in mic (good), "line" doesn't do anything(?), "mic" sets the mic jack (good), there are two capture volumes (capture and capture 1) and two "input source"s (line, mic, or front mic). I think there should only be mic or front mic unless there's someway to set the mic jack as "line in" somewhere else? I can now record from the built-in mic ("front mic") vs 1.0.20. So to summarize: - independent headphones don't have separate headphone volumes - "digital" should be the master capture control - there seems to be duplicate capture controls (capture & capture 1, wit corresponding "input sources") - "pc beep" seems to be a bit busted Alistair On Mon, Sep 14, 2009 at 6:29 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sat, 12 Sep 2009 22:48:28 -0400, > Alistair Boyle wrote: >> >> Hi, >> >> I sent this email to the alsa-user mailing list a while back but >> didn't get much of a response. It looks like you're the guy to talk to >> about this from looking at the alsa repository. Hopefully you can help >> me get this working. > > Please post to alsa-devel ML and add me to Cc. > > What I can advise now is to try the very latest alsa-driver, e.g. > 1.0.21 release. > > > HTH, > > Takashi > >> >> Alistair Boyle >> >> >> ---------- Forwarded message ---------- >> From: Alistair Boyle <alistair.js.boyle(a)gmail.com> >> Date: Tue, Aug 4, 2009 at 2:26 AM >> Subject: Dell Studio 14, internal mic >> To: alsa-users(a)alsa-project.org >> >> >> Hello, >> >> I've got a new Dell Studio 14. The laptop has a built-in mic and >> speakers, two head-phone jacks and a microphone jack. Using the >> snd-hda-intel driver (kernel 2.6.30), sound plays fine through the >> built-in speakers and I can record through the mic jack. It doesn't >> look like there's a way to set the 2 headphone jack volumes, nor is >> there a way to set the capture source to the built-in mic. Finally, >> and not surprisingly, when headphones are plugged in the built-in >> speakers don't mute. >> >> (The built-in mic is the current priority.) >> >> It looks like there were tweaks added to handle the (nominally >> similar) Dell Studio 15 and 17 which can be forced on with the >> "model=dell-m6" driver option but adding this option to the module >> removed all the capture devices from the amixer options. A similar >> result occurs for dell-m6-amic and dmic. >> >> I'm at a loss as to where to look next. I'm "code competent" and with >> a pointer in the right direction I can debug. >> >> I've attached the suggested debug info. >> >> So a couple questions, in no particular order: >> >> 1) Has anyone dealt with the Dell Studio 14 laptops so far? >> >> 2) Is there any other debug output that would be helpful? >> >> 3) Where should I start looking in the code? >> >> Alistair >

2 9

Re: [alsa-devel] ASoC updates for 2.6.32
by Mark Brown 17 Sep '09

17 Sep '09

The following changes since commit cdc65fbe18aef15e92d2ebb410a189fbf956fb06: Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes 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 Huang Weiyi (1): ASoC: remove unused #include <linux/version.h> Jassi (1): ASoC: S3C lrsync function made to work with IRQs disabled. Julia Lawall (1): ASoC: Clean up error handling in MPC5200 DMA setup Mark Brown (1): ASoC: Fix display of stream name in DAPM debugfs sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/fsl/mpc5200_dma.c | 33 ++++++++++++++++++++------------- sound/soc/s3c24xx/s3c-i2s-v2.c | 16 ++++++++++------ sound/soc/soc-dapm.c | 7 ++++--- 6 files changed, 34 insertions(+), 25 deletions(-)

2 1

[alsa-devel] [PATCH 1/7] ak4114 - fix errors in output selector bits
by pavel.hofman＠ivitera.com 17 Sep '09

17 Sep '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/include/ak4114.h b/include/ak4114.h index d293d36..3ce69fd 100644 --- a/include/ak4114.h +++ b/include/ak4114.h @@ -95,13 +95,13 @@ /* AK4114_REG_IO0 */ #define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */ -#define AK4114_OPS12 (1<<2) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS11 (1<<1) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS10 (1<<0) /* Output Though Data Selector for TX1 pin */ +#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */ #define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */ -#define AK4114_OPS02 (1<<2) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS01 (1<<1) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS00 (1<<0) /* Output Though Data Selector for TX0 pin */ +#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */ /* AK4114_REG_IO1 */ #define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */ -- 1.5.6.3

3 17

[alsa-devel] [PATCH] S3C64XX I2S: Added machine driver for WM8580
by jassi brar 17 Sep '09

17 Sep '09

New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 284 +++++++++++++++++++++++++++++++++++ 3 files changed, 294 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..769697e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,14 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..7790406 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..186e1de --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,284 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(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/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_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; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* Currently, WM8580 driver doesn't support PLL-out rates + * other than those mentioned in Table-52 Page-58 of WM8580A + * manual. Some of these unavailable rates include + * {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a value that results in + * (params_rate(params) * rfs), and itself, acceptable to the both + * the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + /* Exceptions for this CPU-CODEC combination */ + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Assuming the CODEC driver evaluates it's rfs too from this call */ + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static int smdk64xx_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, 0); +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, + .hw_free = smdk64xx_hw_free, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* LineIn enabled by default */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* Stereo enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_disable_pin(codec, "Center/Sub"); + snd_soc_dapm_disable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

2 7

[alsa-devel] [PATCH 1/5] ASoC: Blackfin AC97: add a few missing multichannel define handling
by Mike Frysinger 17 Sep '09

17 Sep '09

From: Cliff Cai <cliff.cai(a)analog.com> Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was merged, but two places were missed (the probe/resume functions). Restore handling of this option so it gets initialized properly. Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ac97.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 2758b90..e693229 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -277,7 +277,11 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -334,7 +338,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, goto sport_err; } /*SPORT works in TDM mode to simulate AC97 transfers*/ +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; -- 1.6.5.rc1

3 7

[alsa-devel] Application and hw mixing for VoIP
by Fabien COMTE 17 Sep '09

17 Sep '09

Hi, I need some help please. I have an application that does VoIP and use OSS emulated interface. I want to : - send through VoIP application a mixed sound from mic and an another application (aplay should be ok for testing). mic -----------------> mix ---- oss emulation -----> voip application application 1 -------> (aplay) - use sound received from VoIP with HP output and an another application (arecord should be ok for testing). ---------> application 2 (arecord) voip application ----- oss emulation ----> copy ---------> HP output I tried a lot of asoundrc but they doesn't work. Every try I did, Dsnoop cannot mix sound for application 1 and copy is already in use. Could someone help me ? # wm8731 is the name of my hardware codec pcm_slave.wm8731 { type hw card 0 device 0 } pcm.from_voip { ipc_key 1025 type dmix slave.pcm wm8731 } #one called "to_voip" for capturing pcm.to_voip { ipc_key 1027 type dsnoop slave.pcm wm8731 } #and this is the real magic pcm.temp_voip { type asym playback.pcm "from_voip" capture.pcm "to_voip" } #a quick plug plugin for above device to do the converting magic pcm.voip { type plug slave.pcm "temp_voip" } #a ctl device to keep xmms happy ctl.voip { type hw card 0 } #for aoss: pcm.dsp0 { type plug slave.pcm "voip" } ctl.mixer0 { type hw card 0 } pcm.from_voip_copy { type plug slave from_voip route_policy copy } Thank you. Fabien

1 0

[alsa-devel] [PATCH] S3C64XX I2S: Added machine driver for WM8580
by Jassi 17 Sep '09

17 Sep '09

New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. I2S controller-0 is set as default, set S3C64XX_I2S_CNTRLR to 2 inorder to use I2S_v4 controller of S3C6410. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 288 +++++++++++++++++++++++++++++++++++ 3 files changed, 298 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..769697e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -56,6 +56,14 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..7790406 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -23,6 +23,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +34,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..6489e52 --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,288 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(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/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_CNTRLR 0 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ (12000000) + +static int smdk64xx_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; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_rate(params)) { + case 8000: + pll_out = 8192000 / 4; break; + case 11025: + pll_out = 11289600 / 4; break; + case 16000: + pll_out = 8192000 / 2; break; + case 22050: + pll_out = 11289600 / 2; break; + case 32000: + pll_out = 8192000; break; + case 44100: + pll_out = 11289600; break; + case 48000: + pll_out = 12288000; break; + case 64000: + pll_out = 24576000; break; + case 88200: + pll_out = 11289600 * 2; break; + case 96000: + pll_out = 12288000 * 2; break; + default: + return -EINVAL; + } + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* We block CODCLK from MCLK of WM8580 */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CODCLKSRC_EXT, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from it's PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-ADC/DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_ADC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + /* Don't output to the disconnected pin */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_CLKOUTSRC, + WM8580_CLKSRC_NONE); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + rfs = 256; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + rfs = 256; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static int smdk64xx_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, 0); +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, + .hw_free = smdk64xx_hw_free, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* LineIn enabled by default */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + snd_soc_dapm_enable_pin(codec, "LineIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* Stereo enabled by default */ + snd_soc_dapm_enable_pin(codec, "Front-L/R"); + snd_soc_dapm_disable_pin(codec, "Center/Sub"); + snd_soc_dapm_disable_pin(codec, "Rear-L/R"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_CNTRLR], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_CNTRLR], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +static void __exit smdk64xx_audio_exit(void) +{ + platform_device_unregister(smdk64xx_snd_device); +} +module_exit(smdk64xx_audio_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

3 2

[alsa-devel] Can't resume from suspend because firmware is not found ?!?
by Giuliano Pochini 17 Sep '09

17 Sep '09

I tested suspend/resume on a laptop because I failed to make it work on my G4 (grr!!). Suspend work, resume does not. From dmesg (below) it looks like the soundcard driver is asked to resume before the ATA driver, thus is cannot load the firmware. Am I overlooking something ? Suggestions ? ---- Echoaudio Indigo IOx 0000:07:00.0: resume Echoaudio Indigo IOx 0000:07:00.0: PCI INT A -> GSI 18 (level, low) -> IRQ 18 Echoaudio Indigo IOx 0000:07:00.0: enabling bus mastering ALSA echoaudio.c:2177: res res1 Echoaudio Indigo IOx 0000:07:00.0: firmware: requesting ea/indigo_iox_dsp.fw PM: Adding info for No Bus:0000:07:00.0 ata4.00: ACPI cmd ef/03:0c:00:00:00:a0 filtered out ata4.00: ACPI cmd ef/03:42:00:00:00:a0 filtered out ata4.00: configured for UDMA/33 ata2: SATA link down (SStatus 0 SControl 300) ata3: SATA link down (SStatus 0 SControl 300) ata1: SATA link up 1.5 Gbps (SStatus 113 SControl 300) ata1.00: _GTF unexpected object type 0x1 ata1.00: _GTF unexpected object type 0x1 ata1.00: configured for UDMA/100 tg3: eth0: Link is up at 10 Mbps, half duplex. tg3: eth0: Flow control is off for TX and off for RX. PM: Removing info for No Bus:0000:07:00.0 ALSA echoaudio.c:47: get_firmware(): Firmware not available (-2) pm_op(): pci_pm_resume+0x0/0x79 returns -2 PM: Device 0000:07:00.0 failed to resume: error -2 ---- -- Giuliano.

2 4

[alsa-devel] Support for Infrasonic Quartet
by Pavel Hofman 16 Sep '09

16 Sep '09

Hi Takashi, I am sending patches for Infrasonic Quartet. The card has three external clock modes therefore I had to modify ice1724.c to support multiple spdif_master types. While working on the driver, I did some minor fixes to ak4xxx-adda.c too. For proc files I added a conversion function from int to string representing its binary format (quartet.c:get_binary). If you know of an existing utility, please let me know and I will use it instead. The binary output helps a lot with debugging. The quartet patch does not comply with checkpatch.pl in 2.6.27 since the script throws a false positive for [] in #define. Thanks a lot for comments. Best regards, Pavel.

1 1

[alsa-devel] [PATCH 1/9] ASoC: remove unused #include <linux/version.h>
by Huang Weiyi 16 Sep '09

16 Sep '09

Remove unused #include <linux/version.h>('s) in sound/soc/codecs/ad1836.c sound/soc/codecs/ad1938.c sound/soc/codecs/wm8974.c Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> --- sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/wm8974.c | 1 - 3 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 3612bb9..01343dc 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -18,7 +18,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index e62b277..9a049a1 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -28,7 +28,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index d8a013a..98d663a 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -12,7 +12,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.6.1.3

2 1

Re: [alsa-devel] ALSA: hda - add speaker and fix headphone switch for ALC883
by Takashi Iwai 16 Sep '09

16 Sep '09

At Wed, 16 Sep 2009 14:45:12 +0200, okias wrote: > > log is with auto > > so: no channels (even 6ch), only stereo. Speaker switch work > correctly. Headphone doesn't work (nothing happening, probably > incorrect value). Disconnecting and connecting headphone jack > disabling speaker, so this is okey. int. mic works. The auto model doesn't support multi-channel mode for shared jacks, so it's expected. Could you try the patch below with the very latest alsa-driver snapshot tarball? ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Apply it on alsa-driver/alsa-kernel directory with patch -p2 option. thanks, Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7702f15..1296058 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7927,8 +7927,9 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { static struct snd_kcontrol_new alc883_targa_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), @@ -7947,8 +7948,9 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = { static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -7960,6 +7962,15 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9167,7 +9178,8 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc882_targa_automute, }, [ALC883_TARGA_8ch_DIG] = { - .mixers = { alc883_base_mixer, alc883_chmode_mixer }, + .mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer, + alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, alc883_targa_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids),

1 0

[alsa-devel] Ethernet AVB - XMOS reference desing
by errordeveloper＠gmail.com 16 Sep '09

16 Sep '09

Hello. I m just getting my head around the whole lot of ethernet audio protocols avaliable, it's a big deal of brands really .. but now it looks like there will be a new rather standard protocol - ieee802.1 ethernet AV bridge. here is a reference design which includes driver source-code. i haven't quite figured is tha tlicene is comptible with gpl really.. but i suppose you don't need to use that code, just look at it as a piece of documentation .. it could be so great to have ethernet audio in linux! i have seen the audio science interface and draver exists, but they are using cobra-net ..which is probably good, but it's not as standard as ieee AVB. here you can find an article about the current status of the protocol mess: http://prosoundnewseurope.com/pdf/PSNLive/PSNLive_2009.pdf (page 28) best regards, -- ilya .d

1 0

Re: [alsa-devel] ALSA: hda - add speaker and fix headphone switch for ALC883
by Takashi Iwai 16 Sep '09

16 Sep '09

At Wed, 16 Sep 2009 14:16:11 +0200, okias wrote: > > > At Tue, 15 Sep 2009 21:13:00 +0200, > > okias wrote: > >> > >> ALSA: hda - add speaker and fix headphone switch for ALC883 > >> > >> This patch changing "Headphone Playback Switch" from 0x1b to 0x14 > >> (documentation says "Port-D", 3,5 jack) and adding "Speaker Playback > >> Switch" 0x1b (Port-E, LINE2). > >> > >> Signed-off-by: David Heidelberger <d.okias(a)gmail.com> > > > > Thanks for the patch. > > For which machine is this change necessary? Could you give > > alsa-info.sh output (run with --no-upload option)? > > > > yes, attaching to message Thanks. Just wondering whether model=auto works properly for your device. If it does, we can get rid of the quirk entry but let driver parsing the BIOS pin-configs instead. So far, the quirk for 8ch mode is used although BIOS tells that there is no 8-channel outputs. Takashi > > > Apart from that your patch is for older codes, it's not good to > > change alc88*_base_mixer[]. This definition is for the fixed pin > > design. If it doesn't match with your machine, it means that the > > quirk is wrong (or the auto-parser is buggy). Better to define the > > fixed mixer array, and don't change the common definition just for a > > single model. > > > > > > Takashi > > > >> > >> --- a/sound/pci/hda/patch_realtek.c 2009-09-15 14:54:37.067310114 +0200 > >> +++ b/sound/pci/hda/patch_realtek.c 2009-09-15 14:54:43.444437970 +0200 > >> @@ -7979,6 +7979,8 @@ > >> static struct snd_kcontrol_new alc883_base_mixer[] = { > >> HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), > >> HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), > >> + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), > >> + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), > >> HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), > >> HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), > >> HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, > >> HDA_OUTPUT), > >> @@ -7987,7 +7989,6 @@ > >> HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), > >> HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), > >> HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), > >> - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), > >> HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), > >> HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), > >> HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), > >> [2 alc883_speaker-headphone_fix.patch <application/octet-stream (base64)>] > >> > >> [3 <text/plain; us-ascii (7bit)>] > >> _______________________________________________ > >> Alsa-devel mailing list > >> Alsa-devel(a)alsa-project.org > >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > [2 alsa-info.txt.Y4QfxkuHwO <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH 1/3] ASoC: TWL6030: Add twl6030 codec driver
by Lopez Cruz, Misael 16 Sep '09

16 Sep '09

Initial version of TWL6030 codec driver. The TWL6030 codec uses a propietary digital audio interface called McPDM. TWL6030 codec has two power modes: low-power and high-performance, this initial version only supports high-performance mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc-dai.h | 3 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 670 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 95 +++++++ 5 files changed, 774 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..8b0d04b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -35,6 +35,9 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J #define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J +/* propietary formats */ +#define SND_SOC_DAIFMT_MCPDM 0x10 /* Texas Instruments McPDM */ + /* * DAI Clock gating. * diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..0effb52 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6030 if TWL6030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 @@ -114,6 +115,9 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TWL4030 tristate +config SND_SOC_TWL6030 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..b70c8a1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -13,6 +13,7 @@ snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-twl6030-objs := twl6030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -50,6 +51,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TWL6030) += snd-soc-twl6030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c new file mode 100644 index 0000000..c5e76fa --- /dev/null +++ b/sound/soc/codecs/twl6030.c @@ -0,0 +1,670 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl6030.h" + +#define TWL6030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* + * twl6030 register cache & default register settings + */ +static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6030_ASICID (ro) 0x01 */ + 0x00, /* TWL6030_ASICREV (ro) 0x02 */ + 0x00, /* TWL6030_INTID 0x03 */ + 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_NCPCTRL 0x05 */ + 0x00, /* TWL6030_LDOCTL 0x06 */ + 0x00, /* TWL6030_HPPLLCTL 0x07 */ + 0x00, /* TWL6030_LPPLLCTL 0x08 */ + 0x00, /* TWL6030_LPPLLDIV 0x09 */ + 0x00, /* TWL6030_AMICBCTL 0x0A */ + 0x00, /* TWL6030_DMICBCTL 0x0B */ + 0x18, /* TWL6030_MICLCTL 0x0C */ + 0x18, /* TWL6030_MICRCTL 0x0D */ + 0x00, /* TWL6030_MICGAIN 0x0E */ + 0x1B, /* TWL6030_LINEGAIN 0x0F */ + 0x00, /* TWL6030_HSLCTL 0x10 */ + 0x00, /* TWL6030_HSRCTL 0x11 */ + 0x00, /* TWL6030_HSGAIN 0x12 */ + 0x06, /* TWL6030_EARCTL 0x13 */ + 0x00, /* TWL6030_HFLCTL 0x14 */ + 0x03, /* TWL6030_HFLGAIN 0x15 */ + 0x00, /* TWL6030_HFRCTL 0x16 */ + 0x03, /* TWL6030_HFRGAIN 0x17 */ + 0x00, /* TWL6030_VIBCTLL 0x18 */ + 0x00, /* TWL6030_VIBDATL 0x19 */ + 0x00, /* TWL6030_VIBCTLR 0x1A */ + 0x00, /* TWL6030_VIBDATR 0x1B */ + 0x00, /* TWL6030_HKCTL1 0x1C */ + 0x00, /* TWL6030_HKCTL2 0x1D */ + 0x00, /* TWL6030_GPOCTL 0x1E */ + 0x00, /* TWL6030_ALB 0x1F */ + 0x00, /* TWL6030_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6030_TRIM1 0x28 */ + 0x00, /* TWL6030_TRIM2 0x29 */ + 0x00, /* TWL6030_TRIM3 0x2A */ + 0x00, /* TWL6030_HSOTRIM 0x2B */ + 0x00, /* TWL6030_HFOTRIM 0x2C */ + 0x09, /* TWL6030_ACCCTL 0x2D */ + 0x00, /* TWL6030_STATUS (ro) 0x2E */ +}; + +/* + * read twl6030 register cache + */ +static inline unsigned int twl6030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6030 register cache + */ +static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the twl6030 register space + */ +static int twl6030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if ((reg < TWL6030_REG_INTID) || (reg > TWL6030_REG_ACCCTL)) + /* read-only registers */ + return -EIO; + + twl6030_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL6030_MODULE_AUDIO, value, reg); +} + +static void twl6030_init_chip(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int i; + + /* allow registers to be accessed by i2c */ + twl6030_write(codec, TWL6030_REG_ACCCTL, cache[TWL6030_REG_ACCCTL]); + + /* set all audio section registers to reasonable defaults */ + /* avoid read-only (ASICID, ASICREV, STATUS) and non-used registers */ + for (i = TWL6030_REG_INTMR; i <= TWL6030_REG_DLB; i++) + twl6030_write(codec, i, cache[i]); + for (i = TWL6030_REG_TRIM1; i < TWL6030_REG_ACCCTL; i++) + twl6030_write(codec, i, cache[i]); +} + +static void twl6030_power_up(struct snd_soc_codec *codec) +{ + struct snd_soc_device *socdev = codec->socdev; + struct twl6030_setup_data *setup = socdev->codec_data; + + setup->codec_enable(1); + + /* sync registers updated during power-up sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); +} + +static void twl6030_power_down(struct snd_soc_codec *codec) +{ + struct snd_soc_device *socdev = codec->socdev; + struct twl6030_setup_data *setup = socdev->codec_data; + + setup->codec_enable(0); + + /* sync registers updated during power-down sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6030_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6030_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6030_enum[] = { + SOC_ENUM_SINGLE(TWL6030_REG_MICLCTL, 3, 3, twl6030_amicl_texts), + SOC_ENUM_SINGLE(TWL6030_REG_MICRCTL, 3, 3, twl6030_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6030_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6030_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6030_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier (Attenuator) Volume", + TWL6030_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Amplifier Volume", + TWL6030_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6030_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6030_REG_HFLGAIN, TWL6030_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxers for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6030_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6030_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6030_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6030_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6030_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6030_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6030_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6030_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6030_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6030_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6030_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6030_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6030_dapm_widgets, + ARRAY_SIZE(twl6030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int twl6030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + twl6030_power_up(codec); + break; + case SND_SOC_BIAS_PREPARE: + twl6030_power_up(codec); + break; + case SND_SOC_BIAS_STANDBY: + twl6030_power_up(codec); + break; + case SND_SOC_BIAS_OFF: + twl6030_power_down(codec); + break; + } + codec->bias_level = level; + + return 0; +} + +static int twl6030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int rate, format; + + /* hardware dai (McPDM) requires bit stream of twice + * the actual rate, application processor (audio backend) + * should upsample accordingly + */ + rate = params_rate(params); + switch (rate) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unknown rate %d\n", rate); + return -EINVAL; + } + + /* the sample lenght handled by codec at A-D and D-A + * conversion is 16-bits, but the dai requires 32-bits + */ + format = params_format(params); + if (format != SNDRV_PCM_FORMAT_S32_LE) { + dev_err(codec->dev, "unknown format %d\n", format); + return -EINVAL; + } + + return 0; +} + +static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 hppll, lppll; + + hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + hppll &= TWL6030_HPLLRST; + + switch (freq) { + case 12000000: + /* MCLK input, PLL enabled */ + hppll = TWL6030_MCLK_12000KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLENA; + break; + case 19200000: + /* MCLK input, PLL disabled */ + hppll = TWL6030_MCLK_19200KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLBP; + break; + case 26000000: + /* MCLK input, PLL enabled */ + hppll = TWL6030_MCLK_26000KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer input, PLL disabled */ + hppll = TWL6030_MCLK_38400KHZ + | TWL6030_HPLLSQRENA + | TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + + /* Disable LPPLL and select HPPLL */ + lppll = TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + return 0; +} + +static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + int format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* propietary dai format */ + if (format != SND_SOC_DAIFMT_MCPDM) + return -EINVAL; + + return 0; +} + +static struct snd_soc_dai_ops twl6030_dai_ops = { + .hw_params = twl6030_hw_params, + .set_sysclk = twl6030_set_dai_sysclk, + .set_fmt = twl6030_set_dai_fmt, +}; + +struct snd_soc_dai twl6030_dai = { + .name = "twl6030", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .ops = &twl6030_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6030_dai); + +static int twl6030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6030_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialize the driver + */ +static int twl6030_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + int ret = 0; + + dev_info(codec->dev, "TWL6030 Audio Codec init\n"); + + codec->name = "twl6030"; + codec->owner = THIS_MODULE; + codec->read = twl6030_read_reg_cache; + codec->write = twl6030_write; + codec->set_bias_level = twl6030_set_bias_level; + codec->dai = &twl6030_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6030_reg); + codec->reg_cache = kmemdup(twl6030_reg, sizeof(twl6030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* 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\n"); + goto pcm_err; + } + + /* power on device */ + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + twl6030_init_chip(codec); + + snd_soc_add_controls(codec, twl6030_snd_controls, + ARRAY_SIZE(twl6030_snd_controls)); + twl6030_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *twl6030_socdev; + +static int twl6030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + codec->dev = &pdev->dev; + socdev->card->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + twl6030_socdev = socdev; + twl6030_init(socdev); + + return 0; +} + +static int twl6030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dev_info(codec->dev, "TWL6030 Audio Codec remove\n"); + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6030 = { + .probe = twl6030_probe, + .remove = twl6030_remove, + .suspend = twl6030_suspend, + .resume = twl6030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6030); + +static int __init twl6030_modinit(void) +{ + return snd_soc_register_dai(&twl6030_dai); +} +module_init(twl6030_modinit); + +static void __exit twl6030_exit(void) +{ + snd_soc_unregister_dai(&twl6030_dai); +} +module_exit(twl6030_exit); + +MODULE_DESCRIPTION("ASoC TWL6030 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h new file mode 100644 index 0000000..a83f9e7 --- /dev/null +++ b/sound/soc/codecs/twl6030.h @@ -0,0 +1,95 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL6030_H__ +#define __TWL6030_H__ + +#define TWL6030_REG_ASICID 0x01 +#define TWL6030_REG_ASICREV 0x02 +#define TWL6030_REG_INTID 0x03 +#define TWL6030_REG_INTMR 0x04 +#define TWL6030_REG_NCPCTL 0x05 +#define TWL6030_REG_LDOCTL 0x06 +#define TWL6030_REG_HPPLLCTL 0x07 +#define TWL6030_REG_LPPLLCTL 0x08 +#define TWL6030_REG_LPPLLDIV 0x09 +#define TWL6030_REG_AMICBCTL 0x0A +#define TWL6030_REG_DMICBCTL 0x0B +#define TWL6030_REG_MICLCTL 0x0C +#define TWL6030_REG_MICRCTL 0x0D +#define TWL6030_REG_MICGAIN 0x0E +#define TWL6030_REG_LINEGAIN 0x0F +#define TWL6030_REG_HSLCTL 0x10 +#define TWL6030_REG_HSRCTL 0x11 +#define TWL6030_REG_HSGAIN 0x12 +#define TWL6030_REG_EARCTL 0x13 +#define TWL6030_REG_HFLCTL 0x14 +#define TWL6030_REG_HFLGAIN 0x15 +#define TWL6030_REG_HFRCTL 0x16 +#define TWL6030_REG_HFRGAIN 0x17 +#define TWL6030_REG_VIBCTLL 0x18 +#define TWL6030_REG_VIBDATL 0x19 +#define TWL6030_REG_VIBCTLR 0x1A +#define TWL6030_REG_VIBDATR 0x1B +#define TWL6030_REG_HKCTL1 0x1C +#define TWL6030_REG_HKCTL2 0x1D +#define TWL6030_REG_GPOCTL 0x1E +#define TWL6030_REG_ALB 0x1F +#define TWL6030_REG_DLB 0x20 +#define TWL6030_REG_TRIM1 0x28 +#define TWL6030_REG_TRIM2 0x29 +#define TWL6030_REG_TRIM3 0x2A +#define TWL6030_REG_HSOTRIM 0x2B +#define TWL6030_REG_HFOTRIM 0x2C +#define TWL6030_REG_ACCCTL 0x2D +#define TWL6030_REG_STATUS 0x2E + +#define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) + +/* HPPLLCTL (0x07) fields */ + +#define TWL6030_HPLLENA 0x01 +#define TWL6030_HPLLRST 0x02 +#define TWL6030_HPLLBP 0x04 +#define TWL6030_HPLLSQRENA 0x08 +#define TWL6030_HPLLSQRBP 0x10 +#define TWL6030_MCLK_12000KHZ (0 << 5) +#define TWL6030_MCLK_19200KHZ (1 << 5) +#define TWL6030_MCLK_26000KHZ (2 << 5) +#define TWL6030_MCLK_38400KHZ (3 << 5) +#define TWL6030_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6030_LPLLENA 0x01 +#define TWL6030_LPLLRST 0x02 +#define TWL6030_LPLLSEL 0x04 +#define TWL6030_FIN 0x08 +#define TWL6030_HPLLSEL 0x10 + +extern struct snd_soc_dai twl6030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6030; + +struct twl6030_setup_data { + void (*codec_enable)(int enable); +}; + +#endif /* End of __TWL6030_H__ */ -- 1.5.4.3

2 5

[alsa-devel] alsamixer is not able to set pulseaudio master volume
by Ozan Çağlayan 16 Sep '09

16 Sep '09

Hi, I've just noticed that using alsamixer to set pulseaudio's master volume is not working anymore. I switched back to an earlier alsa-utils snapshot (20090624) still the same. I didn't do any changes to pulseaudio for a long time, it's 0.9.15 with the proposed patchset of Lennart. alsamixer is launched, the master channel is displayed but pressing Up and Down doesn't change anything on the UI. Pressing M changes 'OO' to 'MM'. Here's what happens on PA's side when I click UP/DOWN: I: alsa-source.c: Device suspended... I: module-suspend-on-idle.c: Sink alsa_output.pci_8086_3a3e_sound_card_0 idle for too long, suspending ... I: alsa-sink.c: Device suspended... I: client.c: Created 1 "Native client (UNIX socket client)" D: protocol-native.c: Protocol version: remote 15, local 15 I: protocol-native.c: Got credentials: uid=1000 gid=100 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: module-augment-properties.c: Looking for .desktop file for alsamixer I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. I: module-device-restore.c: Storing volume/mute for device sink:alsa_output.pci_8086_3a3e_sound_card_0. .. .. Synced.

2 2

[alsa-devel] ALSA: hda - add speaker and fix headphone switch for ALC883
by okias 16 Sep '09

16 Sep '09

ALSA: hda - add speaker and fix headphone switch for ALC883 This patch changing "Headphone Playback Switch" from 0x1b to 0x14 (documentation says "Port-D", 3,5 jack) and adding "Speaker Playback Switch" 0x1b (Port-E, LINE2). Signed-off-by: David Heidelberger <d.okias(a)gmail.com> --- --- a/sound/pci/hda/patch_realtek.c 2009-09-15 14:54:37.067310114 +0200 +++ b/sound/pci/hda/patch_realtek.c 2009-09-15 14:54:43.444437970 +0200 @@ -7979,6 +7979,8 @@ static struct snd_kcontrol_new alc883_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), @@ -7987,7 +7989,6 @@ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),

2 2

[alsa-devel] Radeon RV730 (Radeon HD 4670) not playing back
by Anders Kirchenbauer 16 Sep '09

16 Sep '09

I am unable to have sound playback through HDMI, presumably because there is no PCM codec support with the current state of my driver. An alsa-info.sh dump is below, and below that a pastebin.ca of speaker-test http://www.alsa-project.org/db/?f=2e2c6ba9f4052ef0a2defc25688067d597737af9 http://pastebin.ca/1567621 Also, changing the PCI ID to 1002aa38 in the two places in patch_atihdmi.c resulted in no sound or mixer device being available when the drivers were loaded. Went back to distro-default drivers afterwards. Only IEC958 Mute/Unmute control works, rest are permission denied. All permission denied are denied under user and root accounts. alsamixer defines card as "ATI R6xx HDMI"

1 0

[alsa-devel] [PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 15 Sep '09

15 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 | 4 ++-- sound/soc/davinci/davinci-evm.c | 7 ++++--- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 4dfd4ad..047ee39 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -13,9 +13,9 @@ config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..7ccbe66 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 @@ -176,7 +177,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci-evm audio machine driver */ +/* davinci dm6446, dm355 or dm365 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -243,7 +244,7 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm()) { + if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { -- 1.6.0.4

3 3

[alsa-devel] [PATCH 1/6] AK4114 - fix output selector bits, fix spelling
by pavel.hofman＠ivitera.com 15 Sep '09

15 Sep '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/include/ak4114.h b/include/ak4114.h index d293d36..3ce69fd 100644 --- a/include/ak4114.h +++ b/include/ak4114.h @@ -95,13 +95,13 @@ /* AK4114_REG_IO0 */ #define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */ -#define AK4114_OPS12 (1<<2) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS11 (1<<1) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS10 (1<<0) /* Output Though Data Selector for TX1 pin */ +#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */ #define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */ -#define AK4114_OPS02 (1<<2) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS01 (1<<1) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS00 (1<<0) /* Output Though Data Selector for TX0 pin */ +#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */ /* AK4114_REG_IO1 */ #define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */ -- 1.5.6.3

3 17

[alsa-devel] [PATCH v2] use set_channel_map api to reorder channels for AD1938 and AD1836
by Barry Song 15 Sep '09

15 Sep '09

From: Barry Song <barry.song(a)analog.com> Signed-off-by: Barry Song <barry.song(a)analog.com> --- sound/soc/blackfin/bf5xx-ad1836.c | 7 +++++ sound/soc/blackfin/bf5xx-ad1938.c | 9 ++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 ++++-- sound/soc/blackfin/bf5xx-tdm.c | 45 +++++++++++++++++++++++++++++------ sound/soc/blackfin/bf5xx-tdm.h | 11 ++++++++ 5 files changed, 69 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index cd361e3..0f45a3f 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, 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; + unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..2ef1e50 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, 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; + unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return ret; /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e82..a8c73cb 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -43,7 +43,7 @@ #include "bf5xx-tdm.h" #include "bf5xx-sport.h" -#define PCM_BUFFER_MAX 0x10000 +#define PCM_BUFFER_MAX 0x8000 #define FRAGMENT_SIZE_MIN (4*1024) #define FRAGMENTS_MIN 2 #define FRAGMENTS_MAX 32 @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + tdm_port->tx_map[i]) = *src++; dst += 8; } } else { @@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + tdm_port->rx_map[i]); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..600987d 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,40 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + int i; + unsigned int slot; + unsigned int tx_mapped = 0, rx_mapped = 0; + + if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || + (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(tx_mapped & (1 << slot)))) { + bf5xx_tdm.tx_map[i] = slot; + tx_mapped |= 1 << slot; + } else + return -EINVAL; + } + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(rx_mapped & (1 << slot)))) { + bf5xx_tdm.rx_map[i] = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +261,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_channel_map = bf5xx_tdm_set_channel_map, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +327,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d..04189a1 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,17 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +#define BFIN_TDM_DAI_MAX_SLOTS 8 +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS]; + unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS]; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- 1.5.6.3

2 1

[alsa-devel] [PATCH] alsa-lib: mixer/simple_none: fix softfloat compilation
by Manuel Lauss 15 Sep '09

15 Sep '09

mixer/simple_none.c uses HAVE_SOFT_FLOAT tests to work around sqrt() uses but the definition in config.h is never picked up, leading to link errors in libasound.so when built with --with-softfloat. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Tested on MIPS. The pcm_ladspa plugin still calls 'sqrt()' and should be excluded from the build if --with-softfloat is specified. src/mixer/mixer_simple.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/src/mixer/mixer_simple.h b/src/mixer/mixer_simple.h index e88b007..bc9d20c 100644 --- a/src/mixer/mixer_simple.h +++ b/src/mixer/mixer_simple.h @@ -19,6 +19,7 @@ * */ +#include <config.h> #include "mixer_abst.h" /* make local functions really local */ -- 1.6.5.rc1

1 0

[alsa-devel] [PATCH 1/2] ak4113 build support
by pavel.hofman＠ivitera.com 15 Sep '09

15 Sep '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/i2c/other/Makefile b/i2c/other/Makefile index 37f1603..a86990e 100644 --- a/i2c/other/Makefile +++ b/i2c/other/Makefile @@ -5,7 +5,7 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf -export-objs += ak4xxx-adda.o ak4114.o ak4117.o pt2258.o tea575x-tuner.o +export-objs += ak4xxx-adda.o ak4113.o ak4114.o ak4117.o pt2258.o tea575x-tuner.o include $(SND_TOPDIR)/alsa-kernel/i2c/other/Makefile diff --git a/i2c/other/ak4113.c b/i2c/other/ak4113.c new file mode 100644 index 0000000..41cb4ce --- /dev/null +++ b/i2c/other/ak4113.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/i2c/other/ak4113.c" diff --git a/utils/insert b/utils/insert index af911d1..bfbbb83 100755 --- a/utils/insert +++ b/utils/insert @@ -395,6 +395,7 @@ case "$1" in #insert snd-i2c$SUFF #insert snd-cs8427$SUFF insert snd-ak4114$SUFF + insert snd-ak4113$SUFF insert snd-ak4xxx-adda$SUFF insert snd-ice17xx-ak4xxx$SUFF insert snd-ice1724$SUFF enable=1,1 -- 1.5.6.3

1 1

[alsa-devel] Support for Infrasonic Quartet
by Pavel Hofman 15 Sep '09

15 Sep '09

Hi Takashi, I am sending patches for Infrasonic Quartet. The card has three external clock modes therefore I had to modify ice1724.c to support multiple spdif_master types. While working on the driver, I did some minor fixes to ak4xxx-adda.c too. For proc files I added a conversion function from int to string representing its binary format (quartet.c:get_binary). If you know of an existing utility, please let me know and I will use it instead. The binary output helps a lot with debugging. The quartet patch does not comply with checkpatch.pl in 2.6.27 since the script throws a false positive for [] in #define. Thanks a lot for comments. Best regards, Pavel.

1 0

[alsa-devel] [PATCH] RFC: ASoC: Davinci: Fix data_type divide by zero
by Troy Kisky 15 Sep '09

15 Sep '09

This is a proposed fix for the data_type divide by 0 error. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- This applies on top of my last patchset to Kevin's temp/asoc branch sound/soc/davinci/davinci-i2s.c | 18 +++++++----------- sound/soc/davinci/davinci-mcasp.h | 7 ++++++- sound/soc/davinci/davinci-pcm.c | 14 ++++++-------- 3 files changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 5f61016..252c3ea 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -123,13 +123,18 @@ static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { }; struct davinci_mcbsp_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to an array of struct davinci_pcm_dma_params ** in + * davinci_pcm_open. + */ + struct davinci_pcm_dma_params *dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 int mode; u32 pcr; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; /* * Combining both channels into 1 element will at least double the * amount of time between servicing the dma channel, increase @@ -255,14 +260,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback) toggle_clock(dev, playback); } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; - cpu_dai->dma_data = dev->dma_params[substream->stream]; - return 0; -} - #define DEFAULT_BITPERSAMPLE 16 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -393,7 +390,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct davinci_pcm_dma_params *dma_params = dai->dma_data; struct davinci_mcbsp_dev *dev = dai->private_data; struct snd_soc_dai *codec_dai = dev->codec_dai; struct snd_interval *i = NULL; @@ -407,6 +403,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int right_first = 0; int convert_mono_stereo = 0; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0; + struct davinci_pcm_dma_params *dma_params = dev->dma_params[substream->stream]; i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); bits_per_sample = snd_interval_value(i); @@ -529,7 +526,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 554354c..c6ea04d 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,10 +39,15 @@ enum { }; struct davinci_audio_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to an array of struct davinci_pcm_dma_params ** in + * davinci_pcm_open. + */ + struct davinci_pcm_dma_params *dma_params[2]; void __iomem *base; int sample_rate; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; unsigned int codec_fmt; /* McASP specific data */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2729a80..0bd3b2e 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -265,8 +265,7 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd = runtime->private_data; struct snd_dma_buffer *iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *dma_data = prtd->params; unsigned int data_type = dma_data->data_type; unsigned int acnt = dma_data->acnt; unsigned int convert_mono_stereo = dma_data->convert_mono_stereo; @@ -482,15 +481,12 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *dma_data = prtd->params; int link; if (!dma_data) return -ENODEV; - prtd->params = dma_data; - /* Request asp master DMA channel */ link = prtd->asp_channel = edma_alloc_channel(dma_data->channel, davinci_pcm_dma_irq, substream, EVENTQ_0); @@ -660,7 +656,8 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) static int davinci_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params **pa = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = pa[substream->stream]; struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; struct snd_pcm_hardware *ppcm; @@ -668,7 +665,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, dma_data->sram_size, ppcm); + allocate_sram(substream, dma_params->sram_size, ppcm); snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, @@ -681,6 +678,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; spin_lock_init(&prtd->lock); + prtd->params = dma_params; prtd->asp_channel = -1; prtd->asp_link[0] = prtd->asp_link[1] = -1; prtd->ram_channel = -1; -- 1.5.6.3

3 4

[alsa-devel] SPDIF with X-Fi
by Daniel L. Miller 15 Sep '09

15 Sep '09

Perhaps I'm going about this the wrong way (if so, please tell me), but... My X-Fi card is now working with the new driver. I am connected to my amplifier via a fiber optic cable. Sending standard two-channel audio to the "front" alsa plug gets directed to the spdif output and sounds great. But I want to have my full surround active - particularly for movies already encoded with dts and/or ac-3. Thus far, I have not been able to get mplayer to send digital audio - I either get mplayer errors or silence. Once, and only once, I got xbmc to work with digital settings - never since. So I know it SHOULD be possible - I'm just missing something. Is anyone getting this to work? -- Daniel

1 0

[alsa-devel] iecset with X-Fi
by Daniel L. Miller 15 Sep '09

15 Sep '09

Is the iecset utility supposed to do anything with X-Fi cards and the new driver? For me, it reports status - but seems to ignore any parameters to change the settings. -- Daniel

1 0

[alsa-devel] [PATCH] WM8580 DAI: Debugged
by jassisinghbrar＠gmail.com 14 Sep '09

14 Sep '09

From: Jassi <jassi.brar(a)samsung.com> Debugged improper definition of 'struct snd_soc_dai wm8580_dai' Also, implemented various clock sourcing options for WM8580 blocks. Signed-Off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/codecs/wm8580.c | 116 ++++++++++++++++++++++++++++++++------------- sound/soc/codecs/wm8580.h | 17 ++++--- 2 files changed, 93 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..105d452 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -106,9 +106,16 @@ /* CLKSEL (register 8h) */ #define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_MCLK 0x00 #define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 #define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_ADC_CLKSEL_MASK 0x0c +#define WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK 0x00 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLA 0x04 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLB 0x08 +#define WM8580_CLKSEL_ADC_CLKSEL_MCLK 0x0c + /* AIF control 1 (registers 9h-bh) */ #define WM8580_AIF_RATE_MASK 0x7 #define WM8580_AIF_RATE_128 0x0 @@ -121,15 +128,16 @@ #define WM8580_AIF_BCLKSEL_MASK 0x18 #define WM8580_AIF_BCLKSEL_64 0x00 -#define WM8580_AIF_BCLKSEL_128 0x08 -#define WM8580_AIF_BCLKSEL_256 0x10 +#define WM8580_AIF_BCLKSEL_32 0x08 +#define WM8580_AIF_BCLKSEL_16 0x10 #define WM8580_AIF_BCLKSEL_SYSCLK 0x18 #define WM8580_AIF_MS 0x20 #define WM8580_AIF_CLKSRC_MASK 0xc0 +#define WM8580_AIF_CLKSRC_ADCMCLK 0x00 #define WM8580_AIF_CLKSRC_PLLA 0x40 -#define WM8580_AIF_CLKSRC_PLLB 0x40 +#define WM8580_AIF_CLKSRC_PLLB 0x80 #define WM8580_AIF_CLKSRC_MCLK 0xc0 /* AIF control 2 (registers ch-eh) */ @@ -630,6 +638,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div) { case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_DAC_CLKSEL_MCLK; break; case WM8580_CLKSRC_PLLA: @@ -646,6 +655,33 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_CLKSEL, reg); break; + case WM8580_ADC_CLKSEL: + reg = snd_soc_read(codec, WM8580_CLKSEL); + reg &= ~WM8580_CLKSEL_ADC_CLKSEL_MASK; + + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_MCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLB; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_CLKSEL, reg); + break; + case WM8580_CLKOUTSRC: reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; @@ -672,6 +708,32 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_PLLB4, reg); break; + case WM8580_PAIF_CLKSEL: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_CLKSRC_MASK; + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_AIF_CLKSRC_ADCMCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_AIF_CLKSRC_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_AIF_CLKSRC_PLLB; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_AIF_CLKSRC_MCLK; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + default: return -EINVAL; } @@ -731,7 +793,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, #define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops wm8580_dai_ops_playback = { +static struct snd_soc_dai_ops wm8580_dai_ops = { .hw_params = wm8580_paif_hw_params, .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, @@ -739,38 +801,26 @@ static struct snd_soc_dai_ops wm8580_dai_ops_playback = { .digital_mute = wm8580_digital_mute, }; -static struct snd_soc_dai_ops wm8580_dai_ops_capture = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, -}; - struct snd_soc_dai wm8580_dai[] = { - { - .name = "WM8580 PAIFRX", - .id = 0, - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 6, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = WM8580_FORMATS, - }, - .ops = &wm8580_dai_ops_playback, + { + .name = "WM8580 PAIF", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = WM8580_FORMATS, }, - { - .name = "WM8580 PAIFTX", - .id = 1, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = WM8580_FORMATS, - }, - .ops = &wm8580_dai_ops_capture, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = WM8580_FORMATS, }, + .ops = &wm8580_dai_ops, + }, }; EXPORT_SYMBOL_GPL(wm8580_dai); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 0dfb5dd..65f9225 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -20,13 +20,16 @@ #define WM8580_MCLK 1 #define WM8580_DAC_CLKSEL 2 -#define WM8580_CLKOUTSRC 3 - -#define WM8580_CLKSRC_MCLK 1 -#define WM8580_CLKSRC_PLLA 2 -#define WM8580_CLKSRC_PLLB 3 -#define WM8580_CLKSRC_OSC 4 -#define WM8580_CLKSRC_NONE 5 +#define WM8580_ADC_CLKSEL 3 +#define WM8580_CLKOUTSRC 4 +#define WM8580_PAIF_CLKSEL 5 + +#define WM8580_CLKSRC_ADCMCLK 0 +#define WM8580_CLKSRC_MCLK 1 +#define WM8580_CLKSRC_PLLA 2 +#define WM8580_CLKSRC_PLLB 3 +#define WM8580_CLKSRC_OSC 4 +#define WM8580_CLKSRC_NONE 5 #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 -- 1.6.2.5

3 3

[alsa-devel] [PATCH v4 1/2] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 14 Sep '09

14 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

2 1

[PATCH] ASoC: Make WM8580 DAI clocking more standard
by Mark Brown 14 Sep '09

14 Sep '09

Move the WM8580 driver over to more of a standard clocking setup method, specifying the master clock rate for the DAI using the set_sysclk() API and configuring the LRCLK automatically using that. This is an initial update only to present external APIs, further work that needs doing includes enforcing the constraints flowing from the SYSCLK selection, automatic tie in with PLL configuration and BCLK configuration (the current BCLK configuration is compatible with all sample rates). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 121 +++++++++++++++++++++++++++++++-------------- sound/soc/codecs/wm8580.h | 4 +- 2 files changed, 85 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..dffc093 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -105,19 +105,18 @@ #define WM8580_PLLB4_CLKOUTSRC_OSCCLK 0x180 /* CLKSEL (register 8h) */ -#define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 -#define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 -#define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_SHIFT 0 +#define WM8580_CLKSEL_ADC_CLKSEL_MASK 0x0c +#define WM8580_CLKSEL_ADC_CLKSEL_SHIFT 2 + +#define WM8580_CLKSEL_PLLA 0x01 +#define WM8580_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_MCLK 0x03 /* AIF control 1 (registers 9h-bh) */ #define WM8580_AIF_RATE_MASK 0x7 -#define WM8580_AIF_RATE_128 0x0 -#define WM8580_AIF_RATE_192 0x1 -#define WM8580_AIF_RATE_256 0x2 -#define WM8580_AIF_RATE_384 0x3 -#define WM8580_AIF_RATE_512 0x4 -#define WM8580_AIF_RATE_768 0x5 -#define WM8580_AIF_RATE_1152 0x6 +#define WM8580_AIF_RATE_SHIFT 0x2 #define WM8580_AIF_BCLKSEL_MASK 0x18 #define WM8580_AIF_BCLKSEL_64 0x00 @@ -203,6 +202,7 @@ struct wm8580_priv { u16 reg_cache[WM8580_MAX_REGISTER + 1]; struct pll_state a; struct pll_state b; + int sysclk[WM8580_NUM_DAI]; }; static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); @@ -476,6 +476,10 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; } +static int lrclk_ratios[] = { + 128, 192, 256, 384, 512, 768, 1152 +}; + /* * Set PCM DAI bit size and sample rate. */ @@ -483,12 +487,31 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_codec *codec = dai->codec; + struct wm8580_priv *priv = codec->private_data; u16 paifb = snd_soc_read(codec, WM8580_PAIF3 + dai->id); + int sysclk, i; + + /* SYSCLK needs to be configured prior to this being called. */ + if (!priv->sysclk[dai->id]) { + dev_err(dai->dev, "SYSCLK not configured\n"); + return -EINVAL; + } + sysclk = priv->sysclk[dai->id]; + dev_dbg(dai->dev, "SYSCLK is %dHz\n", sysclk); + + paifb &= ~(WM8580_AIF_LENGTH_MASK | WM8580_AIF_RATE_MASK); + + /* Look for an exact match; the steps are too big otherwise */ + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + if (params_rate(params) == sysclk / lrclk_ratios[i]) + break; + } + if (i == ARRAY_SIZE(lrclk_ratios)) + return -EINVAL; + dev_dbg(dai->dev, "LRCLK ratio %d\n", lrclk_ratios[i]); + paifb |= i << WM8580_AIF_RATE_SHIFT; - paifb &= ~WM8580_AIF_LENGTH_MASK; /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -624,28 +647,6 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_PLLB4, reg); break; - case WM8580_DAC_CLKSEL: - reg = snd_soc_read(codec, WM8580_CLKSEL); - reg &= ~WM8580_CLKSEL_DAC_CLKSEL_MASK; - - switch (div) { - case WM8580_CLKSRC_MCLK: - break; - - case WM8580_CLKSRC_PLLA: - reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLA; - break; - - case WM8580_CLKSRC_PLLB: - reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLB; - break; - - default: - return -EINVAL; - } - snd_soc_write(codec, WM8580_CLKSEL, reg); - break; - case WM8580_CLKOUTSRC: reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; @@ -679,6 +680,48 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, return 0; } +static int wm8580_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8580_priv *priv = codec->private_data; + int mask, val, ret; + + switch (clk_id) { + case WM8580_CLKSRC_MCLK: + val = WM8580_CLKSEL_MCLK; + break; + case WM8580_CLKSRC_PLLA: + val = WM8580_CLKSEL_PLLA; + break; + case WM8580_CLKSRC_PLLB: + val = WM8580_CLKSEL_PLLB; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case WM8580_DAI_PAIFRX: + mask = WM8580_CLKSEL_DAC_CLKSEL_MASK; + break; + + case WM8580_DAI_PAIFTX: + mask = WM8580_CLKSEL_ADC_CLKSEL_MASK; + val <<= WM8580_CLKSEL_ADC_CLKSEL_SHIFT; + break; + + default: + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, WM8580_CLKSEL, mask, val); + if (ret == 0) + priv->sysclk[dai->id] = freq; + + return ret; +} + static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -736,6 +779,7 @@ static struct snd_soc_dai_ops wm8580_dai_ops_playback = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, + .set_sysclk = wm8580_set_dai_sysclk, .digital_mute = wm8580_digital_mute, }; @@ -743,13 +787,14 @@ static struct snd_soc_dai_ops wm8580_dai_ops_capture = { .hw_params = wm8580_paif_hw_params, .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, + .set_sysclk = wm8580_set_dai_sysclk, .set_pll = wm8580_set_dai_pll, }; struct snd_soc_dai wm8580_dai[] = { { .name = "WM8580 PAIFRX", - .id = 0, + .id = WM8580_DAI_PAIFRX, .playback = { .stream_name = "Playback", .channels_min = 1, @@ -761,7 +806,7 @@ struct snd_soc_dai wm8580_dai[] = { }, { .name = "WM8580 PAIFTX", - .id = 1, + .id = WM8580_DAI_PAIFTX, .capture = { .stream_name = "Capture", .channels_min = 2, diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 0dfb5dd..f03ae9e 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -19,8 +19,7 @@ #define WM8580_PLLB 2 #define WM8580_MCLK 1 -#define WM8580_DAC_CLKSEL 2 -#define WM8580_CLKOUTSRC 3 +#define WM8580_CLKOUTSRC 2 #define WM8580_CLKSRC_MCLK 1 #define WM8580_CLKSRC_PLLA 2 @@ -30,6 +29,7 @@ #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 +#define WM8580_NUM_DAI 2 extern struct snd_soc_dai wm8580_dai[]; extern struct snd_soc_codec_device soc_codec_dev_wm8580; -- 1.6.3.3

1 0

[alsa-devel] [PATCH 3/3] ASoC: TWL6030: Handle power-up seq completion thru audio interrupt
by Lopez Cruz, Misael 14 Sep '09

14 Sep '09

NAUDINT interrupt line is provided by the TWL6030 codec to signal externally events like headset plug/unplug, hook, power-up sequence completion, etc. When the codec is powered-up through external AUDPWRON line it will start its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. CODEC driver starts a wait_for_completion just after calling external power-up callback. It's signaled as complete when servicing READYINT interrupt. MACHINE drivers should request IRQ line used in corresponding hardware platform and initialize workqueue and completion structs of twl6030_setup_data as well. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 103 +++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl6030.h | 17 +++++++ 2 files changed, 113 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 0f2269e..dc85399 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -43,6 +43,7 @@ /* codec private data */ struct twl6030_priv_data { + int codec_powered; unsigned int sysclk; }; @@ -54,7 +55,7 @@ static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { 0x4B, /* TWL6030_ASICID (ro) 0x01 */ 0x00, /* TWL6030_ASICREV (ro) 0x02 */ 0x00, /* TWL6030_INTID 0x03 */ - 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_INTMR 0x04 */ 0x00, /* TWL6030_NCPCTRL 0x05 */ 0x00, /* TWL6030_LDOCTL 0x06 */ 0x00, /* TWL6030_HPPLLCTL 0x07 */ @@ -127,6 +128,23 @@ static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, } /* + * read from twl6030 hardware register + */ +static int twl6030_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg > TWL6030_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &value, reg); + twl6030_write_reg_cache(codec, reg, value); + + return value; +} + +/* * write to the twl6030 register space */ static int twl6030_write(struct snd_soc_codec *codec, @@ -159,27 +177,89 @@ static void twl6030_init_chip(struct snd_soc_codec *codec) static void twl6030_power_up(struct snd_soc_codec *codec) { struct snd_soc_device *socdev = codec->socdev; + struct twl6030_priv_data *priv = codec->private_data; struct twl6030_setup_data *setup = socdev->codec_data; + if (priv->codec_powered) + return; + setup->codec_enable(1); + /* wait for ready interrupt */ + wait_for_completion(&setup->ready_completion); + /* sync registers updated during power-up sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); } static void twl6030_power_down(struct snd_soc_codec *codec) { struct snd_soc_device *socdev = codec->socdev; + struct twl6030_priv_data *priv = codec->private_data; struct twl6030_setup_data *setup = socdev->codec_data; setup->codec_enable(0); + udelay(500); /* sync registers updated during power-down sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + + priv->codec_powered = 0; +} + +/* audio interrupt handler */ +irqreturn_t twl6030_naudint_handler(int irq, void *data) +{ + struct twl6030_setup_data *setup = data; + + schedule_work(&setup->audint_work); + + /* disable audint irq to let workqueue to execute */ + disable_irq_nosync(irq); + + return IRQ_HANDLED; +} + +void twl6030_naudint_work(struct work_struct *work) +{ + struct snd_soc_codec *codec; + struct twl6030_setup_data *setup; + struct twl6030_priv_data *priv; + u8 intid; + + setup = container_of(work, struct twl6030_setup_data, audint_work); + codec = setup->codec; + priv = codec->private_data; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &intid, TWL6030_REG_INTID); + + switch (intid) { + case TWL6030_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6030_PLUGINT: + case TWL6030_UNPLUGINT: + case TWL6030_HOOKINT: + break; + case TWL6030_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6030_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6030_READYINT: + priv->codec_powered = 1; + complete(&setup->ready_completion); + break; + default: + dev_err(codec->dev, "unknown twl6030 audio interrupt\n"); + } + + enable_irq(setup->irq); } /* set headset dac and driver power mode */ @@ -679,6 +759,7 @@ static int twl6030_resume(struct platform_device *pdev) static int twl6030_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_setup_data *setup = socdev->codec_data; int ret = 0; dev_info(codec->dev, "TWL6030 Audio Codec init\n"); @@ -703,6 +784,14 @@ static int twl6030_init(struct snd_soc_device *socdev) goto pcm_err; } + /* platform setup data is required for power-off/off the device */ + if (setup == NULL) { + printk(KERN_ERR "twl6030: platform setup data missing\n"); + goto pcm_err; + } + + setup->codec = codec; + /* power on device */ twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 9e49c85..1e1fb3f 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -64,6 +64,16 @@ #define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) +/* INTID (0x03) fields */ + +#define TWL6030_THINT 0x01 +#define TWL6030_PLUGINT 0x02 +#define TWL6030_UNPLUGINT 0x04 +#define TWL6030_HOOKINT 0x08 +#define TWL6030_HFINT 0x10 +#define TWL6030_VIBINT 0x20 +#define TWL6030_READYINT 0x40 + /* HPPLLCTL (0x07) fields */ #define TWL6030_HPLLENA 0x01 @@ -105,8 +115,15 @@ extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; +irqreturn_t twl6030_naudint_handler(int irq, void *data); +void twl6030_naudint_work(struct work_struct *work); + struct twl6030_setup_data { void (*codec_enable)(int enable); + int irq; + struct work_struct audint_work; + struct completion ready_completion; + struct snd_soc_codec *codec; }; #endif /* End of __TWL6030_H__ */ -- 1.5.4.3

2 1

[alsa-devel] [PATCH 2/3] ASoC: TWL6030: Add support for low-power mode
by Lopez Cruz, Misael 14 Sep '09

14 Sep '09

TWL6030 codec supports two power modes: low-power and high-performance. In low-power mode, headset downlink must be the only path enabled and components in that path (headset DAC and driver) should be in that mode too. In this mode, codec can stream audio at 44.1 and 48 kHz if sys clock is configured to 17.64 and 19.2 MHz from CLK32K using Low-Power PLL, respectively. In high-performance mode, codec can only work at 19.2 MHz from High-Performance PLL. All paths supported in the codec can be used but the audio can be streamed only at 48 kHz. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 192 +++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/twl6030.h | 19 ++++- 2 files changed, 181 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index c5e76fa..0f2269e 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -41,6 +41,11 @@ #define TWL6030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +/* codec private data */ +struct twl6030_priv_data { + unsigned int sysclk; +}; + /* * twl6030 register cache & default register settings */ @@ -177,6 +182,29 @@ static void twl6030_power_down(struct snd_soc_codec *codec) twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); } +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6030_HSDRVMODEL | TWL6030_HSDACMODEL; + + hslctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSLCTL); + hsrctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6030_write(codec, TWL6030_REG_HSLCTL, hslctl); + twl6030_write(codec, TWL6030_REG_HSRCTL, hsrctl); + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -419,6 +447,10 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_priv_data *twl6030_priv = codec->private_data; int rate, format; /* hardware dai (McPDM) requires bit stream of twice @@ -428,7 +460,20 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { case 44100: + if (twl6030_priv->sysclk != 17640000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, twl6030_priv->sysclk); + return -EINVAL; + } + break; case 48000: + if (twl6030_priv->sysclk != 19200000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, twl6030_priv->sysclk); + return -EINVAL; + } break; default: dev_err(codec->dev, "unknown rate %d\n", rate); @@ -451,46 +496,120 @@ static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_priv_data *twl6030_priv = codec->private_data; u8 hppll, lppll; + lppll = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); - hppll &= TWL6030_HPLLRST; - - switch (freq) { - case 12000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_12000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; - break; - case 19200000: - /* MCLK input, PLL disabled */ - hppll = TWL6030_MCLK_19200KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLBP; + + switch (clk_id) { + case TWL6030_SYSCLK_SEL_LPPLL: + if (freq != 32768) { + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* CLK32K input requires low-power PLL */ + lppll |= TWL6030_LPLLENA | TWL6030_LPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + mdelay(5); + lppll &= ~TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + hppll &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); break; - case 26000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_26000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; + case TWL6030_SYSCLK_SEL_HPPLL: + switch (freq) { + case 12000000: + hppll = TWL6030_MCLK_12000KHZ; + break; + case 26000000: + hppll = TWL6030_MCLK_26000KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 12 and 26 MHz freqs require high-performance PLL */ + hppll |= TWL6030_HPLLSQRBP | TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by HPPLL */ + twl6030_priv->sysclk = 19200000; break; - case 38400000: - /* clk slicer input, PLL disabled */ - hppll = TWL6030_MCLK_38400KHZ - | TWL6030_HPLLSQRENA - | TWL6030_HPLLBP; + case TWL6030_SYSCLK_SEL_MCLK: + switch (freq) { + case 19200000: + hppll = TWL6030_MCLK_19200KHZ | TWL6030_HPLLSQRBP; + break; + case 38400000: + hppll = TWL6030_MCLK_38400KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 19.2 and 38.4 MHz freqs don't require PLL */ + hppll |= TWL6030_HPLLBP; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by MCLK */ + twl6030_priv->sysclk = 19200000; break; default: - dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); return -EINVAL; } - twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + return 0; +} + +static int twl6030_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_priv_data *twl6030_priv = codec->private_data; + int div; + + if (!freq_in || !freq_out) + return -EINVAL; + + /* div = round(freq_out / freq_in) */ + div = (freq_out + (freq_in >> 2)) / freq_in; - /* Disable LPPLL and select HPPLL */ - lppll = TWL6030_HPLLSEL; - twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + if (pll_id == TWL6030_LPPLL_ID) { + if (div < 512) { + dev_err(codec->dev, "invalid pll divider %d\n", div); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_LPPLLDIV, div - 512); + twl6030_priv->sysclk = freq_out; + } return 0; } @@ -510,6 +629,7 @@ static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6030_dai_ops = { .hw_params = twl6030_hw_params, .set_sysclk = twl6030_set_dai_sysclk, + .set_pll = twl6030_set_dai_pll, .set_fmt = twl6030_set_dai_fmt, }; @@ -614,12 +734,21 @@ static int twl6030_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; + struct twl6030_priv_data *twl6030_priv; + int ret; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; + twl6030_priv = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (twl6030_priv == NULL) { + ret = -ENOMEM; + goto priv_err; + } + codec->dev = &pdev->dev; + codec->private_data = twl6030_priv; socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -629,6 +758,10 @@ static int twl6030_probe(struct platform_device *pdev) twl6030_init(socdev); return 0; + +priv_err: + kfree(codec); + return ret; } static int twl6030_remove(struct platform_device *pdev) @@ -640,6 +773,7 @@ static int twl6030_remove(struct platform_device *pdev) twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); + kfree(codec->private_data); kfree(codec); return 0; diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index a83f9e7..9e49c85 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -77,14 +77,31 @@ #define TWL6030_MCLK_38400KHZ (3 << 5) #define TWL6030_MCLK_MSK 0x60 +#define TWL6030_SYSCLK_SEL_LPPLL 1 +#define TWL6030_SYSCLK_SEL_HPPLL 2 +#define TWL6030_SYSCLK_SEL_MCLK 3 + /* LPPLLCTL (0x08) fields */ #define TWL6030_LPLLENA 0x01 #define TWL6030_LPLLRST 0x02 #define TWL6030_LPLLSEL 0x04 -#define TWL6030_FIN 0x08 +#define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +#define TWL6030_HPPLL_ID 1 +#define TWL6030_LPPLL_ID 2 + +/* HSLCTL (0x10) fields */ + +#define TWL6030_HSDACMODEL 0x02 +#define TWL6030_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6030_HSDACMODER 0x02 +#define TWL6030_HSDRVMODER 0x08 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.5.4.3

2 1

[PATCH] ALSA: hda - Add support for HP dv6
by Takashi Iwai 14 Sep '09

14 Sep '09

Add the quirk entry for HP dv6. Also add a workaround for the headphone detection by setting hp_detect=1 beforehand. Without this, the driver won't do auto-muting because BIOS doesn't give any HP pin but only a line-out pin. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index fe2ba76..cc0e26c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1690,6 +1690,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP mini 1000", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, "HP HDX", STAC_HP_HDX), /* HDX16 */ + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, + "HP dv6", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, @@ -5383,6 +5385,11 @@ again: case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); + /* HP dv6 gives the headphone pin as a line-out. Thus we + * need to set hp_detect flag here to force to enable HP + * detection. + */ + spec->hp_detect = 1; break; case STAC_HP_HDX: spec->num_dmics = 1; -- 1.6.4.2

1 0

[alsa-devel] ice1724: dmesg filled with IRQ errors
by Vedran Miletić 14 Sep '09

14 Sep '09

I ran winecfg, and it probed around sound stuff a bit, and then dmesg got filled with: [ 6623.654899] ice1724: Too long irq loop, status = 0x30 [ 6623.654901] ice1724: Disabling MPU_TX [ 6623.654982] ice1724: Too long irq loop, status = 0x30 [ 6623.654983] ice1724: Disabling MPU_TX [ 6623.655064] ice1724: Too long irq loop, status = 0x30 [ 6623.655065] ice1724: Disabling MPU_TX [ 6623.655146] ice1724: Too long irq loop, status = 0x30 [ 6623.655147] ice1724: Disabling MPU_TX [ 6623.655228] ice1724: Too long irq loop, status = 0x30 [ 6623.655229] ice1724: Disabling MPU_TX [ 6623.655310] ice1724: Too long irq loop, status = 0x30 [ 6623.655311] ice1724: Disabling MPU_TX [ 6623.655391] ice1724: Too long irq loop, status = 0x30 [ 6623.655393] ice1724: Disabling MPU_TX [ 6623.655473] ice1724: Too long irq loop, status = 0x30 [ 6623.655475] ice1724: Disabling MPU_TX [ 6623.655555] ice1724: Too long irq loop, status = 0x30 [ 6623.655557] ice1724: Disabling MPU_TX [ 6623.655637] ice1724: Too long irq loop, status = 0x30 [ 6623.655639] ice1724: Disabling MPU_TX [ 6623.655719] ice1724: Too long irq loop, status = 0x30 [ 6623.655721] ice1724: Disabling MPU_TX [ 6623.655801] ice1724: Too long irq loop, status = 0x30 [ 6623.655803] ice1724: Disabling MPU_TX [ 6623.655883] ice1724: Too long irq loop, status = 0x30 [ 6623.655885] ice1724: Disabling MPU_TX [ 6623.655966] ice1724: Too long irq loop, status = 0x30 [ 6623.655967] ice1724: Disabling MPU_TX And so on. I had to reset, since it was impossible to shutdown or unload ice1724 module. Is this a known bug? Can I help debugging it? Vedran Miletić

2 1

[alsa-devel] [PATCH] use set_channel_map api to reorder channels for AD1938 and AD1836
by Barry Song 14 Sep '09

14 Sep '09

From: Barry Song <Barry.Song(a)analog.com> --- sound/soc/blackfin/bf5xx-ad1836.c | 7 +++++++ sound/soc/blackfin/bf5xx-ad1938.c | 9 ++++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 7 +++++-- sound/soc/blackfin/bf5xx-tdm.c | 28 ++++++++++++++++++++-------- sound/soc/blackfin/bf5xx-tdm.h | 11 +++++++++++ 5 files changed, 51 insertions(+), 11 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index cd361e3..d3b41b7 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, 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; + unsigned int channel_map[8] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, 8, channel_map, + 8, channel_map); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..686e84c 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, 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; + unsigned int channel_map[8] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return ret; /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, 8, channel_map, + 8, channel_map); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e82..881a1d9 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + tdm_port->tx_map[i]) = *src++; dst += 8; } } else { @@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + tdm_port->rx_map[i]); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..6995d5b 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,23 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + int i; + + if ((tx_num > MAX_TDM_SLOTS) || (rx_num > MAX_TDM_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) + bf5xx_tdm.tx_map[i] = tx_slot[i]; + for (i = 0; i < rx_num; i++) + bf5xx_tdm.rx_map[i] = rx_slot[i]; + + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +244,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_channel_map = bf5xx_tdm_set_channel_map, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +310,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d..f85790a 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,17 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +#define MAX_TDM_SLOTS 8 +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + unsigned int tx_map[MAX_TDM_SLOTS]; + unsigned int rx_map[MAX_TDM_SLOTS]; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- 1.5.6.3

2 1

[alsa-devel] [RFC][PATCH] ALSA: pcm_native.c initialize drec and num_drecs before it get used
by Jaswinder Singh Rajput 14 Sep '09

14 Sep '09

[This is untested] drec and num_drecs is used uninitialized for nonblock(goto lock) This also fixed following compilation warnings : CC [M] sound/core/pcm_native.o sound/core/pcm_native.c: In function ‘snd_pcm_drain’: sound/core/pcm_native.c:1413: warning: ‘drec’ may be used uninitialized in this function sound/core/pcm_native.c:1411: warning: ‘num_drecs’ may be used uninitialized in this function Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput(a)gmail.com> --- sound/core/pcm_native.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 59e5fbe..8284dc1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1433,6 +1433,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } else if (substream->f_flags & O_NONBLOCK) nonblock = 1; + drec = &drec_tmp; + num_drecs = 0; + if (nonblock) goto lock; /* no need to allocate waitqueues */ @@ -1445,11 +1448,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_power_unlock(card); return -ENOMEM; } - } else - drec = &drec_tmp; + } /* count only playback streams */ - num_drecs = 0; snd_pcm_group_for_each_entry(s, substream) { runtime = s->runtime; if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 1.6.4.2

2 1

[alsa-devel] still have popping every few seconds, like metronome, with xfi drivers
by Ted T. Logan 14 Sep '09

14 Sep '09

I just tried latest snapshot with my SB0670/XFI, which works fine with the drivers: XFiDrv_Linux_Public_US_1.00.tar.gz and the same exact problems remain for all the snapshots I've tried... clicking like a metronome every 1 second or so on ALL playback. Can anything be done about this? It's completely unusable with these drivers. Thanks

1 0

[alsa-devel] [PATCH v2] add channel order map interface in asoc core
by Barry Song 13 Sep '09

13 Sep '09

The patch adds an interface to set the relationship between audio channel number and slot number. The interface should be really useful because audio channel n doesn't always use slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- include/sound/soc-dai.h | 7 +++++++ sound/soc/soc-core.c | 24 ++++++++++++++++++++++++ 2 files changed, 31 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 97ca9af..5c84566 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -114,6 +114,10 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot); + int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ @@ -148,6 +152,9 @@ struct snd_soc_dai_ops { int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); + int (*set_channel_map)(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad..ba1f8e5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2251,6 +2251,30 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** + * snd_soc_dai_set_channel_map - configure DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + if (dai->ops && dai->ops->set_tdm_slot) + return dai->ops->set_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable -- 1.5.6.3

2 3

[alsa-devel] How to test suspend ?
by Giuliano Pochini 12 Sep '09

12 Sep '09

I couldn't make s2ram work on my machine and /sys/pci/.../power/state virtual files were removed one year ago. How can I test suspend/resume ? -- Giuliano.

3 4

[alsa-devel] SPDIF issues
by Ben 12 Sep '09

12 Sep '09

All, I have been happily using a small setup for the purpose of MythTV. I recently decided to upgrade the graphics so I could enjoy VPDAU which has been integrated into the 0.22 branch of MythTV. There is just one snag, I don't have any sound. I am using an Asus P5E-VM HDMI board, it has onboard sound & graphics. The HDMI interface outputs to my TV and it works fine. I then came across an nvidia 9500GT (gainwood) card which has a VGA/DVI/HDMI out and also takes in sound via SPDIF. I connected the cable to my spdif headers and I can't seem to get any sound. I'm running Ubuntu Karmic so I thought I would upgrade the Alsa drivers/lib/utils to the latest, this has not helped matters. I have checked Alsa mixer and nothing is muted. I have tried: speaker-test -Dplughw:0,1 - (I think this is correct). I'm sure it's something pretty trivial I could be doing wrong but someone mentioned this kind of setup can be quite problematic and suggested I posted the details to this list. Details here: http://www.alsa-project.org/db/?f=ffb9e4b76f37a87f21fec2a4db18295479608d24 I'd appreciate your thoughts! Thanks, Ben

1 0

[alsa-devel] [PATCH 8/8, retry] sound/soc: introduce missing kfree
by Julia Lawall 12 Sep '09

12 Sep '09

From: Julia Lawall <julia(a)diku.dk> Error handling code following a kzalloc should free the allocated data. Error handling code following an ioremap should iounmap the allocated data. The semantic match that finds the first problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @r exists@ local idexpression x; statement S; expression E; identifier f,f1,l; position p1,p2; expression *ptr != NULL; @@ x@p1 = $kmalloc\|kzalloc\|kcalloc$(...); ... if (x == NULL) S <... when != x when != if (...) { <+...x...+> } ( x->f1 = E | (x->f1 == NULL || ...) | f(...,x->f1,...) ) ...> ( return $0\|<+...x...+>\|ptr$; | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/mpc5200_dma.c | 33 ++++++++++++-------- 1 files changed, 20 insertions(+), 13 deletions(-) diff --git a/var/linuxes/linux-next/sound/soc/fsl/mpc5200_dma.c b/var/julia/linuxcopy/sound/soc/fsl/mpc5200_dma.c index 9ff62e3..6096d22 100644 --- a/var/linuxes/linux-next/sound/soc/fsl/mpc5200_dma.c +++ b/var/julia/linuxcopy/sound/soc/fsl/mpc5200_dma.c @@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op) int size, irq, rc; const __be32 *prop; void __iomem *regs; + int ret; /* Fetch the registers and IRQ of the PSC */ irq = irq_of_parse_and_map(op->node, 0); @@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Allocate and initialize the driver private data */ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); if (!psc_dma) { - iounmap(regs); - return -ENOMEM; + ret = -ENOMEM; + goto out_unmap; } /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) - return -ENODEV; + if (!prop || size < sizeof *prop) { + ret = -ENODEV; + goto out_free; + } spin_lock_init(&psc_dma->lock); mutex_init(&psc_dma->mutex); @@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op) if (!psc_dma->capture.bcom_task || !psc_dma->playback.bcom_task) { dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); - iounmap(regs); - kfree(psc_dma); - return -ENODEV; + ret = -ENODEV; + goto out_free; } /* Disable all interrupts and reset the PSC */ @@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op) &psc_dma_bcom_irq_tx, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, - &psc_dma->capture); - free_irq(psc_dma->playback.irq, - &psc_dma->playback); - return -ENODEV; + ret = -ENODEV; + goto out_irq; } /* Save what we've done so it can be found again later */ @@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Tell the ASoC OF helpers about it */ return snd_soc_register_platform(&mpc5200_audio_dma_platform); +out_irq: + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, &psc_dma->capture); + free_irq(psc_dma->playback.irq, &psc_dma->playback); +out_free: + kfree(psc_dma); +out_unmap: + iounmap(regs); + return ret; } EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);

2 1

[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 12 Sep '09

12 Sep '09

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

1 0

[alsa-devel] [PATCH 8/8] sound/soc: introduce missing kfree
by Julia Lawall 12 Sep '09

12 Sep '09

From: Julia Lawall <julia(a)diku.dk> Error handling code following a kzalloc should free the allocated data. Error handling code following an ioremap should iounmap the allocated data. The semantic match that finds the first problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @r exists@ local idexpression x; statement S; expression E; identifier f,f1,l; position p1,p2; expression *ptr != NULL; @@ x@p1 = $kmalloc\|kzalloc\|kcalloc$(...); ... if (x == NULL) S <... when != x when != if (...) { <+...x...+> } ( x->f1 = E | (x->f1 == NULL || ...) | f(...,x->f1,...) ) ...> ( return $0\|<+...x...+>\|ptr$; | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/mpc5200_dma.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9ff62e3..41ffd57 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -469,8 +469,11 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) + if (!prop || size < sizeof *prop) { + iounmap(regs); + kfree(psc_dma); return -ENODEV; + } spin_lock_init(&psc_dma->lock); mutex_init(&psc_dma->mutex);

2 2

[alsa-devel] WM8580 DAI: Debugged
by jassi brar 12 Sep '09

12 Sep '09

Hi, I have been making SMDK6410 I2S work with WM8580 codec. In the process i debugged the wm8580 driver. The patch is:- Last patch(Minor WM8580 enhancements) _minus_ the MCLKRATIO/BCLKRATIO stuff _plus_ the snd_soc_dai definition debugged. The snd_soc_dai debugging is necessary if the Audio Capture is to work with the CODEC driver. Next comes the patch.....

1 0

[alsa-devel] ALSA 1.0.21 release
by Jaroslav Kysela 11 Sep '09

11 Sep '09

Hello all, all ALSA packages except alsa-oss and alsa-firmware (no changes from 1.0.20) are available for download at http://www.alsa-project.org . Changelog is available here: http://www.alsa-project.org/main/index.php/Changes_v1.0.20_v1.0.21 Jaroslav Kysela ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

3 5

[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 11 Sep '09

11 Sep '09

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, who is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

1 0

[alsa-devel] [PATCH] add channel order setting interface in asoc core
by Barry Song 11 Sep '09

11 Sep '09

Hi Mark, As discussed, I am adding a patch to add an interface to set the relationship between audio channel No. and slot No. The interface should be really useful because nobody can ensure audio channel n always uses slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1. After the patch is applied, I can change bfin stuff based on it. Thanks, Barry Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- include/sound/soc-dai.h | 5 +++++ sound/soc/soc-core.c | 18 ++++++++++++++++++ 2 files changed, 23 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 97ca9af..7439425 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -114,6 +114,9 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); +int snd_soc_dai_set_channel_order(struct snd_soc_dai *dai, + unsigned int num, unsigned int *slot); + int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ @@ -148,6 +151,8 @@ struct snd_soc_dai_ops { int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); + int (*set_channel_order)(struct snd_soc_dai *dai, + unsigned int num, unsigned int *slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad..30d0143 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2251,6 +2251,24 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** + * snd_soc_dai_set_channel_order - configure DAI audio channel order + * @dai: DAI + * @num: how many channels + * @slot: pointer to an array which imply the slot number channel 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_order(struct snd_soc_dai *dai, + unsigned int num, unsigned int *slot) +{ + if (dai->ops && dai->ops->set_tdm_slot) + return dai->ops->set_channel_order(dai, num, slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_order); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable -- 1.5.6.3

2 1

[alsa-devel] [PATCH] [RFC 8/13] Intel SST sound card driver
by Vinod Koul 11 Sep '09

11 Sep '09

This adds support for Moorestown ALSA Sound card driver. This is an ALSA driver for supporting PCM playback/capture in traditional ALSA way. Anyone who chooses not to use DSP for decoding/encoding can use ALSA path to play/capture (in non low power mode). This driver registers the control interface and PCM interface with the LPE driver which finally sends it to the hardware. This driver allows any subsystem in OS which wants to use the audio-subsystems to be routed through the ALSA Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intelmid.c --- sound/pci/sst/intelmid.c | 1761 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 1761 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intelmid.c diff --git a/sound/pci/sst/intelmid.c b/sound/pci/sst/intelmid.c new file mode 100644 index 0000000..28a6dcc --- /dev/null +++ b/sound/pci/sst/intelmid.c @@ -0,0 +1,1761 @@ +/* + * intelmid.c - Intel Sound card driver for MID + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Harsha Priya <priya.harsha(a)intel.com> + * Vinod Koul <vinod.koul(a)intel.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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * ALSA driver for Intel MID sound card chipset + */ +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/info.h> +#include <sound/initval.h> +#include <sound/pcm-indirect.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intelmid_snd_control.h" +#include "intelmid.h" +#include "intelmid_pvt.h" +#include <linux/spi/spi.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/sched.h> +#include <linux/gpe.h> +#include <net/netlink.h> +#include <net/genetlink.h> + +MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_DESCRIPTION("Intel MAD Sound card driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{Intel,Intel_MAD}"); + +static int card_index = SNDRV_DEFAULT_IDX1;/* Index 0-MAX */ +static char *card_id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int vendor_id; +static unsigned int audio_event_seqnum; +static int cum_bytes; + + +#ifdef FULL_CTRL +static char *out_names[] = {"Headphones", + "Internal speakers", + "EarPiece"}; +static char *in_names[] = { "DMIC", + "MIC1", + "MIC2", + "Line_in"}; +#endif +static struct genl_family audio_event_genl_family = { + .id = GENL_ID_GENERATE, + .name = "audio events", + .version = 0x01, + .maxattr = 0, +}; + +static struct genl_multicast_group audio_event_mcgrp = { + .name = "audio_group", +}; + +/*control path functionalities*/ +/** +* snd_intelmad_mute_info - provides information about the mute controls +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + WARN_ON(!uinfo); + WARN_ON(!kcontrol); + + /*set up the mute as a boolean mono control with min-max values*/ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = MONO_CNTL; + uinfo->value.integer.min = MIN_MUTE; + uinfo->value.integer.max = MAX_MUTE; + return 0; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_volume_line_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_line_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].linein_vol_max, + ctrl_val[vendor_id].linein_vol_min); + return 0; +} + +/** +* snd_intelmad_mic1_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mic1_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].mic1_vol_max, + ctrl_val[vendor_id].mic1_vol_min); + return 0; +} + +/** +* snd_intelmad_mic2_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mic2_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].mic2_vol_max, + ctrl_val[vendor_id].mic2_vol_min); + return 0; +} + +/** +* snd_intelmad_dmic_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_dmic_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].dmic_vol_max, + ctrl_val[vendor_id].dmic_vol_min); + return 0; +} +#endif +/** +* snd_intelmad_hp_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_hp_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].hp_vol_max, + ctrl_val[vendor_id].hp_vol_min); + return 0; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_speaker_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].speaker_vol_max, + ctrl_val[vendor_id].speaker_vol_min); + return 0; +} + +/** +* snd_intelmad_earpiece_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_earpiece_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, MONO_CNTL, + ctrl_val[vendor_id].earpiece_vol_max, + ctrl_val[vendor_id].earpiece_vol_min); + return 0; +} + +/** +* snd_intelmad_device_info - provides information about the devices available +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the devices's info need +* to be filled +* This function is called when a mixer application requests for device's info +*/ +static int snd_intelmad_device_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + WARN_ON(!kcontrol); + WARN_ON(!uinfo); + /*setup device select as drop down controls with different values*/ + if (OUTPUT_SEL == kcontrol->id.numid) + uinfo->value.enumerated.items = ARRAY_SIZE(out_names); + else + uinfo->value.enumerated.items = ARRAY_SIZE(in_names); + uinfo->count = MONO_CNTL; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = 1; + if (OUTPUT_SEL == kcontrol->id.numid) + strncpy(uinfo->value.enumerated.name, + out_names[uinfo->value.enumerated.item], + strlen(out_names[uinfo->value.enumerated.item])); + else + strncpy(uinfo->value.enumerated.name, + in_names[uinfo->value.enumerated.item], + strlen(in_names[uinfo->value.enumerated.item])); + return 0; +} +#endif + +/** +* snd_intelmad_volume_get - gets the current volume for the control +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info need +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + int ret_val = 0, cntl_list[2] = {0,}, value = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_VOL: + cntl_list[0] = PMIC_SND_RIGHT_HP_VOL; + cntl_list[1] = PMIC_SND_LEFT_HP_VOL; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_VOL: + cntl_list[0] = PMIC_SND_RIGHT_SPEAKER_VOL; + cntl_list[1] = PMIC_SND_LEFT_SPEAKER_VOL; + break; + case LINE_IN_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_RIGHT_LINE_IN; + cntl_list[1] = PMIC_SND_INPUT_VOL_LEFT_LINE_IN; + break; + case EARPIECE_VOL: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_VOL; + break; + case MIC1_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC1; + break; + case MIC2_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC2; + break; + case DMIC_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_DMIC; + break; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->get_vol(cntl_list[0], &value); + uval->value.integer.value[0] = value; + + if (0 != ret_val) + return ret_val; +#ifdef FULL_CTRL + if (HEADPHONES_VOL == kcontrol->id.numid + || INTERNAL_SPEAKERS_VOL == kcontrol->id.numid || + LINE_IN_VOL == kcontrol->id.numid) { +#else + if (1) { +#endif + + ret_val = scard_ops->get_vol(cntl_list[1], &value); + uval->value.integer.value[1] = value; + + if (0 != ret_val) + return ret_val; + } + return ret_val; +} + +/** +* snd_intelmad_mute_get - gets the current mute status for the control +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info need +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + + int cntl_list = 0, ret_val = 0, value = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_MUTE: + cntl_list = PMIC_SND_LEFT_HP_MUTE; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_MUTE: + cntl_list = PMIC_SND_LEFT_SPEAKER_MUTE; + break; + case EARPIECE_MUTE: + cntl_list = PMIC_SND_MONO_EARPIECE_MUTE; + break; + case LINE_IN_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_LINE_IN; + break; + case MIC1_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_MIC1; + break; + case MIC2_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_MIC2; + break; + case DMIC_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_DMIC; + break; + case MASTER_MUTE: + uval->value.integer.value[0] = kcontrol->private_value; + return 0; +#endif + + default: + return -EINVAL; + } + + ret_val = scard_ops->get_mute(cntl_list, &value); + uval->value.integer.value[0] = value; + return ret_val; +} + +/** +* snd_intelmad_volume_set - sets the volume control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + + int ret_val = 0, cntl_list[2] = {0,}; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("volume set called\n:%ld %ld", + uval->value.integer.value[0], + uval->value.integer.value[1]); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_VOL: + cntl_list[0] = PMIC_SND_RIGHT_HP_VOL; + cntl_list[1] = PMIC_SND_LEFT_HP_VOL; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_VOL: + cntl_list[0] = PMIC_SND_RIGHT_SPEAKER_VOL; + cntl_list[1] = PMIC_SND_LEFT_SPEAKER_VOL; + break; + case LINE_IN_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_RIGHT_LINE_IN; + cntl_list[1] = PMIC_SND_INPUT_VOL_LEFT_LINE_IN; + break; + case EARPIECE_VOL: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_VOL; + break; + case MIC1_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC1; + break; + case MIC2_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC2; + break; + case DMIC_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_DMIC; + break; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->set_vol(cntl_list[0], + uval->value.integer.value[0]); + if (0 != ret_val) + return ret_val; +#ifdef FULL_CTRL + if (HEADPHONES_VOL == kcontrol->id.numid + || INTERNAL_SPEAKERS_VOL == kcontrol->id.numid || + LINE_IN_VOL == kcontrol->id.numid) { +#else + if (1) { +#endif + ret_val = scard_ops->set_vol(cntl_list[1], + uval->value.integer.value[1]); + if (0 != ret_val) + return ret_val; + } + kcontrol->private_value = uval->value.integer.value[0]; + return ret_val; +} + + + +/** +* snd_intelmad_mute_set - sets the mute control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_mute_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + int cntl_list[2] = {0,}, ret_val = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + kcontrol->private_value = uval->value.integer.value[0]; + + switch (kcontrol->id.numid) { + case HEADPHONES_MUTE: + cntl_list[0] = PMIC_SND_LEFT_HP_MUTE; + cntl_list[1] = PMIC_SND_RIGHT_HP_MUTE; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_MUTE: + cntl_list[0] = PMIC_SND_LEFT_SPEAKER_MUTE; + cntl_list[1] = PMIC_SND_RIGHT_SPEAKER_MUTE; + break; + case EARPIECE_MUTE: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_MUTE; + break; + case LINE_IN_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_LINE_IN; + break; + case MIC1_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_MIC1; + break; + case MIC2_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_MIC2; + break; + case DMIC_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_DMIC; + break; + case MASTER_MUTE: + mute_all(kcontrol, uval); + kcontrol->private_value = uval->value.integer.value[0]; + return 0; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->set_mute(cntl_list[0], + uval->value.integer.value[0]); + if (0 != ret_val) + return ret_val; + +#ifdef FULL_CTRL + if (HEADPHONES_MUTE == kcontrol->id.numid + || INTERNAL_SPEAKERS_MUTE == kcontrol->id.numid) +#endif + ret_val = scard_ops->set_mute(cntl_list[1], + uval->value.integer.value[0]); + kcontrol->private_value = uval->value.integer.value[0]; + return ret_val; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_device_get - get the device select control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_device_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + uval->value.enumerated.item[0] = kcontrol->private_value; + return 0; +} + +/** +* snd_intelmad_device_set - set the device select control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_device_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + int ret_val = 0; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + /*store value with driver*/ + kcontrol->private_value = uval->value.enumerated.item[0]; + + switch (kcontrol->id.numid) { + case OUTPUT_SEL: + ret_val = scard_ops->set_output_dev( + uval->value.enumerated.item[0]); + break; + case INPUT_SEL: + ret_val = scard_ops->set_input_dev( + uval->value.enumerated.item[0]); + break; + default: + return -EINVAL; + } + kcontrol->private_value = uval->value.enumerated.item[0]; + return ret_val; +} +#endif + +/*structure of all the controls of the sound card that is exposed*/ +static struct snd_kcontrol_new snd_intelmad_controls[] __devinitdata = { +#ifdef FULL_CTRL +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_line_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DMIC Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_dmic_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DMIC Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic1 Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mic1_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic1 Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic2 Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mic2_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic2 Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_device_info, + .get = snd_intelmad_device_get, + .put = snd_intelmad_device_set, + .private_value = 0, +}, +#endif +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_hp_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +#ifdef FULL_CTRL +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Internal Speaker Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_speaker_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Internal Speaker Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mono Earpiece Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_earpiece_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mono Earpiece Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_device_info, + .get = snd_intelmad_device_get, + .put = snd_intelmad_device_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Mute Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +#endif +}; + +/*Data path functionalities*/ +static struct snd_pcm_hardware snd_intelmad_stream = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | + /*SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |*/ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24), + .rates = (SNDRV_PCM_RATE_8000| + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000), + .rate_min = MIN_RATE, + + .rate_max = MAX_RATE, + .channels_min = MIN_CHANNEL, + .channels_max = MAX_CHANNEL, + .buffer_bytes_max = MAX_BUFFER, + .period_bytes_min = MIN_PERIOD_BYTES, + .period_bytes_max = MAX_PERIOD_BYTES, + .periods_min = MIN_PERIODS, + .periods_max = MAX_PERIODS, + .fifo_size = FIFO_SIZE, +}; + +static void send_buffer_to_sst(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_intelmad *intelmaddata = snd_pcm_substream_chip(substream); + struct stream_buffer buffer_to_sst = {0,}; + struct mad_stream_pvt *stream = substream->runtime->private_data; + int ret_val = 0; + /*sends data to SST to be processed*/ + + cum_bytes += bytes; + sst_dbg("bytes = %d \n", bytes); + sst_dbg("cum_bytes = 0x%x, \n", cum_bytes); + buffer_to_sst.length = bytes; + buffer_to_sst.addr = (unsigned long) substream->runtime->dma_area + + rec->sw_data; + /*SST API to actually send the buffer to be played*/ + ret_val = intelmaddata->sstdrv_ops->send_buffer(stream->stream_id, + &buffer_to_sst); + sst_dbg("send_buffer ret_val = 0x%x \n", ret_val); + return; +} + +void start_capture(struct work_struct *work) +{ + int ret_val = 0; + struct mad_capture_wq *stream_work = + container_of(work, struct mad_capture_wq, wq); + struct stream_buffer *buffer_to_sst = &stream_work->buffer_to_sst; + struct snd_pcm_substream *substream = stream_work->substream; + struct mad_stream_pvt *stream = substream->runtime->private_data; + struct snd_intelmad *intelmaddata = snd_pcm_substream_chip(substream); + + ret_val = intelmaddata->sstdrv_ops->send_buffer(stream->stream_id, + buffer_to_sst); + sst_dbg("send_buffer ret_val = 0x%x \n", ret_val); + return; +} + +void snd_card_pcm_wq_function(struct work_struct *work) +{ + struct mad_wq *stream_work = + container_of(work, struct mad_wq, wq); + struct snd_pcm_substream *substream; + struct snd_pcm_indirect *rec = NULL; + struct mad_stream_pvt *stream; + + mutex_lock(&stream_work->wq_lock); + substream = stream_work->substream; + if (substream == NULL || substream->runtime == NULL) { + sst_err("subsream structure is null \n"); + mutex_unlock(&stream_work->wq_lock); + return; + } + stream = substream->runtime->private_data; + sst_dbg("called %d\n", stream->stream_status); + if (stream == NULL) { + sst_err("stream is null in wq function \n"); + mutex_unlock(&stream_work->wq_lock); + return; + } + + if (stream->stream_status != INIT) { + rec = &stream->pcm_indirect; + if (substream->stream == STREAM_OPS_PLAYBACK) + snd_pcm_indirect_playback_transfer(substream, rec, + send_buffer_to_sst); + else if (substream->stream == STREAM_OPS_CAPTURE) { + sst_dbg("calling indirect capture transfer\n"); + snd_pcm_indirect_capture_transfer(substream, rec, + send_buffer_to_sst); + } + stream->stream_status = RUNNING; + } + mutex_unlock(&stream_work->wq_lock); + return; +} + +static int snd_intelmad_pcm_ack(struct snd_pcm_substream *substream) +{ + struct mad_stream_pvt *stream; + + stream = substream->runtime->private_data; + sst_dbg("...called\n"); + stream->mad_msg_wq.substream = substream; + schedule_work(&stream->mad_msg_wq.wq); + return 0; +} + +/** +* snd_intelmad_pcm_trigger - stream activities are handled here +* @substream:substream for which the stream function is called +*@cmd:the stream commamd thats requested from upper layer +* This function is called whenever an a stream activity is invoked +*/ +static int snd_intelmad_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int ret_val = 0, stream_type = 0/*, sample_size = 0*/; + struct snd_intelmad *intelmaddata = NULL; + struct mad_stream_pvt *stream = NULL; + + sst_dbg("called\n"); + + WARN_ON(!substream); + + intelmaddata = snd_pcm_substream_chip(substream); + stream_type = substream->stream; + stream = substream->runtime->private_data; + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + WARN_ON(!intelmaddata->sstdrv_ops); + WARN_ON(!intelmaddata->sstdrv_ops->scard_ops); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + /*invoked on a power managment -resume operation*/ + /*stream parameters and context resuming to be done here*/ + /*TBD*/ + sst_dbg("in resume\n"); + break; + case SNDRV_PCM_TRIGGER_START: + stream->substream = substream; + stream->pcm_size = snd_pcm_lib_buffer_bytes(substream); + sst_dbg("pcm_size=%d\n", stream->pcm_size); +#ifdef TIMER_TS + bps = (substream->runtime->sample_bits/8) + * (substream->runtime->channels) + * (substream->runtime->rate); + stream->pcm_jiffie = ((bps * ELAPSED_PERIOD) / HZ); + stream->pcm_buf_pos = 0; + sst_dbg("pcm_jiffie=%d \n", stream->pcm_jiffie); +#endif + stream->stream_status = STARTED; + if (substream->stream == STREAM_OPS_PLAYBACK) + snd_intelmad_pcm_ack(substream); + else if (substream->stream == STREAM_OPS_CAPTURE) { + stream->mad_capture_wq.substream = substream; + stream->mad_capture_wq.buffer_to_sst.length = + frames_to_bytes(substream->runtime, + substream->runtime->period_size); + stream->mad_capture_wq.buffer_to_sst.addr = + (unsigned long) substream->runtime->dma_area; + schedule_work(&stream->mad_capture_wq.wq); + cum_bytes = stream->mad_capture_wq.buffer_to_sst.length; + stream->stream_status = RUNNING; + } + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + /*invoked on suspend - power management event*/ + /*stream parameters and context to be saved*/ + /*TBD*/ + sst_dbg("in suspend\n"); + break; + case SNDRV_PCM_TRIGGER_STOP: + sst_dbg("in stop\n"); + /*ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_DROP, + &stream->stream_id); + if (0 != ret_val) + return ret_val;*/ + stream->stream_status = DROPPED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_dbg("in pause\n"); + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_PAUSE, + &stream->stream_id); + if (0 != ret_val) + return ret_val; + stream->stream_status = PAUSED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_dbg("in pause release \n"); + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_RESUME, + &stream->stream_id); + if (0 != ret_val) + return ret_val; + stream->stream_status = RUNNING; + break; + default: + return -EINVAL; + } + return ret_val; +} + +/** +* snd_intelmad_pcm_prepare- internal preparation before starting a stream +* @substream: substream for which the function is called +* This function is called when a stream is started for internal preparation. +*/ +static int snd_intelmad_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct mad_stream_pvt *stream = NULL; + int ret_val = 0; + + sst_dbg("called \n"); + + WARN_ON(!substream); + + stream = substream->runtime->private_data; + + sst_dbg("format = %d \n", substream->runtime->format); + if (stream->stream_id == 0) { + ret_val = snd_intelmad_alloc_stream(substream); + cum_bytes = 0; + if (ret_val < 0) + return ret_val; + ret_val = snd_intelmad_init_stream(substream); + if (0 != ret_val) + return ret_val; + sst_dbg("period size = %d \n", + (int)substream->runtime->period_size); + sst_dbg("buf size = %d \n", + (int)substream->runtime->buffer_size); + memset(&stream->pcm_indirect, 0, sizeof(stream->pcm_indirect)); + stream->pcm_indirect.hw_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + stream->pcm_indirect.sw_buffer_size = + stream->pcm_indirect.hw_buffer_size; + /*return back the stream id*/ + snprintf(substream->pcm->id, sizeof(substream->pcm->id), + "%d", stream->stream_id); + sst_dbg("stream id to user = %s\n", substream->pcm->id); + } + return ret_val; +} + +static int snd_intelmad_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret_val = 0; + + sst_dbg("called\n"); + ret_val = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(hw_params)); + return ret_val; +} + +static int snd_intelmad_hw_free(struct snd_pcm_substream *substream) +{ + sst_dbg("called\n"); + return snd_pcm_lib_free_pages(substream); +} + +/** +* snd_intelmad_pcm_pointer- to send the current buffer pointer processed by hw +* @substream: substream for which the function is called +* This function is called by ALSA framework to get the current hw buffer ptr +* when a period is elapsed +*/ +static snd_pcm_uframes_t snd_intelmad_pcm_pointer + (struct snd_pcm_substream *substream) +{ + /*struct snd_pcm_runtime *runtime = substream->runtime;*/ + struct mad_stream_pvt *stream = NULL; + struct snd_intelmad *intelmaddata = NULL; + int ret_val = 0; + unsigned long buf_size = 0; + + WARN_ON(!substream); + + intelmaddata = snd_pcm_substream_chip(substream); + stream = substream->runtime->private_data; + if (stream->stream_status == INIT) + return 0; + + stream->buf_ptr.str_id = stream->stream_id; + +#ifndef TIMER_TS + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_BUFFER_POINTER, + &stream->buf_ptr); + if (0 != ret_val) { + sst_err("error code = 0x%x \n", ret_val); + return ret_val; + } + sst_dbg("samples reported out 0x%lx \n", stream->buf_ptr.buffer_ptr); + buf_size = frames_to_bytes(substream->runtime, + stream->buf_ptr.buffer_ptr); + sst_dbg(" bytes reported out = 0x%lx\n", buf_size); + if (buf_size > cum_bytes) + sst_err("excess reported\n"); + + if (substream->stream == STREAM_OPS_PLAYBACK) + stream->period_elapsed_ptr = snd_pcm_indirect_playback_pointer( + substream, &stream->pcm_indirect, buf_size); + else + stream->period_elapsed_ptr = snd_pcm_indirect_capture_pointer( + substream, &stream->pcm_indirect, buf_size); + + sst_dbg("indirect value = 0x%lx\n", stream->period_elapsed_ptr); + + return stream->period_elapsed_ptr; +#else + stream->pcm_buf_pos += stream->pcm_jiffie; + stream->pcm_buf_pos %= stream->pcm_size; + return snd_pcm_indirect_playback_pointer(substream, + &stream->pcm_indirect, stream->pcm_buf_pos); +#endif + +} + +/** +* snd_intelmad_close- to free parameteres when stream is stopped +* @substream: substream for which the function is called +* This function is called by ALSA framework when stream is stopped +*/ +static int snd_intelmad_close(struct snd_pcm_substream *substream) +{ + struct snd_intelmad *intelmaddata = NULL; + struct mad_stream_pvt *stream = NULL; + int ret_val = 0; + + WARN_ON(!substream); + + stream = substream->runtime->private_data; + + sst_dbg("called \n"); + intelmaddata = snd_pcm_substream_chip(substream); + mutex_lock(&stream->mad_msg_wq.wq_lock); + sst_dbg("str id = %d\n", stream->stream_id); + /*SST API to actually stop/free the stream*/ + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_FREE, + &stream->stream_id); + kfree(substream->runtime->private_data); + mutex_unlock(&stream->mad_msg_wq.wq_lock); + return ret_val; +} + +/** +* snd_intelmad_open- to set runtime parameters during stream start +* @substream: substream for which the function is called +* This function is called by ALSA framework when stream is started +*/ +static int snd_intelmad_open(struct snd_pcm_substream *substream) +{ + struct snd_intelmad *intelmaddata; + struct snd_pcm_runtime *runtime; + struct mad_stream_pvt *stream; + + WARN_ON(!substream); + + sst_dbg("called \n"); + + intelmaddata = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + /*set the runtime hw parameter with our local snd_pcm_hardware struct*/ + runtime->hw = snd_intelmad_stream; + /*setup the internal datastruture stream pointers based on it being + playback or capture stream*/ + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (NULL == stream) + return -ENOMEM; + stream->stream_id = 0; + stream->stream_status = INIT; + INIT_WORK(&stream->mad_msg_wq.wq, snd_card_pcm_wq_function); + INIT_WORK(&stream->mad_capture_wq.wq, start_capture); + mutex_init(&stream->mad_msg_wq.wq_lock); + runtime->private_data = stream; + return 0; +} + +static struct snd_pcm_ops snd_intelmad_playback_ops = { + .open = snd_intelmad_open, + .close = snd_intelmad_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_intelmad_hw_params, + .hw_free = snd_intelmad_hw_free, + .prepare = snd_intelmad_pcm_prepare, + .trigger = snd_intelmad_pcm_trigger, + .pointer = snd_intelmad_pcm_pointer, + .ack = snd_intelmad_pcm_ack, +}; + +static struct snd_pcm_ops snd_intelmad_capture_ops = { + .open = snd_intelmad_open, + .close = snd_intelmad_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_intelmad_hw_params, + .hw_free = snd_intelmad_hw_free, + .prepare = snd_intelmad_pcm_prepare, + .trigger = snd_intelmad_pcm_trigger, + .pointer = snd_intelmad_pcm_pointer, + .ack = snd_intelmad_pcm_ack, +}; + +int snd_intelmad_generate_netlink(u32 orig, enum eaudio_events event) +{ + struct sk_buff *skb = NULL; + struct nlattr *attr = NULL; + struct audio_genl_event *aud_event = NULL; + void *msg_header = NULL; + int size = 0, ret_val = 0; + + /* allocate memory */ + size = nla_total_size(sizeof(struct audio_genl_event)) + \ + nla_total_size(0); + + skb = genlmsg_new(size, GFP_ATOMIC); + if (!skb) + return -ENOMEM; + + /* add the genetlink message header */ + msg_header = genlmsg_put(skb, 0, audio_event_seqnum++, + &audio_event_genl_family, 0, + AUDIO_GENL_CMD_EVENT); + if (!msg_header) { + nlmsg_free(skb); + return -ENOMEM; + } + + /* fill the data */ + attr = nla_reserve(skb, AUDIO_GENL_ATTR_EVENT, \ + sizeof(struct audio_genl_event)); + + if (!attr) { + nlmsg_free(skb); + return -EINVAL; + } + + aud_event = nla_data(attr); + if (!aud_event) { + nlmsg_free(skb); + return -EINVAL; + } + + memset(aud_event, 0, sizeof(struct audio_genl_event)); + + aud_event->orig = orig; + aud_event->event = event; + + /* send multicast genetlink message */ + ret_val = genlmsg_end(skb, msg_header); + if (ret_val < 0) { + nlmsg_free(skb); + return ret_val; + } + + ret_val = genlmsg_multicast(skb, 0, audio_event_mcgrp.id, GFP_ATOMIC); + + if (ret_val) + printk(KERN_INFO "Failed to send a Genetlink message!\n"); + return 0; +} + +#ifdef REG_IRQ +/** +* snd_intelmad_intr_handler- interrupt handler +*@irq : irq number of the interrupt received +*@dev: device context +* This function is called when an interrupt is raised at the sound card +*/ +static irqreturn_t snd_intelmad_intr_handler(int irq, void *dev) +{ + struct snd_intelmad *intelmaddata = + (struct snd_intelmad *)dev; + struct sc_reg_access pmic_reg = {0,}; + int ret_val = 0; + u8 intsts = 0; + memcpy_fromio(&intsts, + ((void *)(intelmaddata->int_base)), + sizeof(u8)); + if (intsts != 0) + sst_dbg("intstatus = 0x%x\n", intsts); + switch (intelmaddata->sstdrv_ops->vendor_id) { + case SND_FS: + if (intsts & 0x1) { + /*send long push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_LONG_PRESS); + } + if (intsts & 0x2) { + /*send short push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_SHORT_PRESS); + } + if (intsts & 0x4) { + /*send headphone detect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HP_DETECT); + } + if (intsts & 0x8) { + /*send headset detect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HS_DETECT); + } + break; + case SND_MX: + if (intsts & 0x2) { + pmic_reg.reg_addr = 0x201; + ret_val = sst_sc_reg_access(&pmic_reg, PMIC_READ, 1); + if (0 != ret_val) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return IRQ_HANDLED; + } + intsts = pmic_reg.value; + if (intsts & 0xc0) + /*send jack detect/undetect*/ + sst_dbg("to handle jack sensing\n"); + } + break; + case SND_NC: + if (intsts & 0x1) { + /*send headset detect/undetect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HS_DETECT); + } + if (intsts & 0x2) { + /*send headphone detect/undetect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HP_DETECT); + } + if (intsts & 0x4) { + /*send short push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_SHORT_PRESS); + } + if (intsts & 0x8) { + /*send long push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_LONG_PRESS); + } + break; + } + return IRQ_HANDLED; +} +static int __devinit snd_intelmad_register_irq( + struct snd_intelmad *intelmaddata) +{ + int ret_val = 0; + u32 regbase = AUDINT_BASE, regsize = 8; + + /* interpret irq field */ + sst_dbg("irq = 0x%x\n", intelmaddata->irq); + + if ((intelmaddata->irq) & PMIC_SOUND_IRQ_TYPE_MASK) { + /* irq -> GPE_ID */ + intelmaddata->irq = intelmaddata->irq & + (~PMIC_SOUND_IRQ_TYPE_MASK); + sst_dbg("gpe = 0x%x\n", intelmaddata->irq); + ret_val = request_gpe(intelmaddata->irq, + (gpio_function_t)snd_intelmad_intr_handler, + intelmaddata, 0); + if (ret_val) { + sst_dbg("cannot register gpe\n"); + return ret_val; + } + + } else { + /* irq -> system irq */ + ret_val = request_irq(intelmaddata->irq, + snd_intelmad_intr_handler, + IRQF_SHARED, DRIVER_NAME, + intelmaddata); + if (0 != ret_val) { + sst_dbg("cannot register IRQ\n"); + return ret_val; + } + } + + intelmaddata->int_base = ioremap_nocache(regbase, regsize); + return ret_val; +} +static int __devinit snd_intelmad_register_netlink(void) +{ + int ret_val = 0; + + ret_val = genl_register_family(&audio_event_genl_family); + if (0 != ret_val) { + sst_dbg("netlink registration failed\n"); + return ret_val; + } + ret_val = genl_register_mc_group(&audio_event_genl_family, + &audio_event_mcgrp); + if (ret_val) { + sst_dbg("netlink group registration failed\n"); + genl_unregister_family(&audio_event_genl_family); + return ret_val; + } + return ret_val; +} +#endif + +static int __devinit snd_intelmad_sst_register( + struct snd_intelmad *intelmaddata) +{ + int ret_val = 0; + struct sc_reg_access pmic_reg = {0,}; + + pmic_reg.reg_addr = 0; + ret_val = sst_sc_reg_access(&pmic_reg, PMIC_READ, 1); + + if (0 != ret_val) + return ret_val; + + vendor_id = pmic_reg.value & (MASK2|MASK1|MASK0); + sst_info("orginal reg n extrated vendor id = 0x%x %d\n", + pmic_reg.value, vendor_id); + if (vendor_id < 0 || vendor_id > 2) { + sst_err("vendor card not supported!!\n"); + return -EIO; + } + intelmaddata->sstdrv_ops->module_name = SST_CARD_NAMES; + intelmaddata->sstdrv_ops->vendor_id = vendor_id; + intelmaddata->sstdrv_ops->scard_ops = &v[vendor_id]; + + /*registering with SST driver to get access to SST APIs to use*/ + ret_val = register_sst_card(intelmaddata->sstdrv_ops); + if (0 != ret_val) { + sst_err("sst card registration failed\n"); + return ret_val; + } + + vendor_id = intelmaddata->sstdrv_ops->vendor_id; + intelmaddata->pmic_status = PMIC_UNINIT; + return ret_val; +} + +/*Driver Init/exit functionalities*/ +/** +* snd_intelmad_pcm- to setup pcm for the card +* @card: pointer to the sound card structure +*@intelmaddata: pointer to internal context +* This function is called from probe function to set up pcm params and functions +*/ +static int __devinit snd_intelmad_pcm(struct snd_card *card, + struct snd_intelmad *intelmaddata) +{ + struct snd_pcm *pcm; + int i, ret_val = 0; + char name[32] = INTEL_MAD; + + WARN_ON(!card); + WARN_ON(!intelmaddata); + + for (i = 0; i < MAX_DEVICES; i++) { + ret_val = snd_pcm_new(card, name, i, PLAYBACK_COUNT, + CAPTURE_COUNT, &pcm); + if (0 != ret_val) + break; + /*setup the ops for playback and capture streams*/ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_intelmad_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_intelmad_capture_ops); + /*setup private data which can be retrieved when required*/ + pcm->private_data = intelmaddata; + pcm->info_flags = 0; + strncpy(pcm->name, card->shortname, strlen(card->shortname)); + /*allocate dma pages for ALSA stream operations*/ + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MIN_BUFFER, MAX_BUFFER); + } + return ret_val; +} + +/** +* snd_intelmad_mixer- to setup mixer settings of the card +*@intelmaddata: pointer to internal context +* This function is called from probe function to set up mixer controls +*/ +static int __devinit snd_intelmad_mixer(struct snd_intelmad *intelmaddata) +{ + struct snd_card *card = NULL; + unsigned int idx; + int ret_val = 0; + char *mixername = "IntelMAD Controls"; + + WARN_ON(!intelmaddata); + + card = intelmaddata->card; + + strncpy(card->mixername, mixername, strlen(mixername)); + /*add all widget controls and expose the same*/ + for (idx = 0; idx < ARRAY_SIZE(snd_intelmad_controls); idx++) { + ret_val = snd_ctl_add(card, + snd_ctl_new1(&snd_intelmad_controls[idx], + intelmaddata)); + sst_dbg("mixer[idx]=%d added \n", idx); + if (0 != ret_val) + break; + } + return ret_val; +} + +/** +* snd_intelmad_dev_free- to free device +*@device: pointer to the device +* This function is called when driver module is removed +*/ +static int snd_intelmad_dev_free(struct snd_device *device) +{ + struct snd_intelmad *intelmaddata; + + WARN_ON(!device); + + intelmaddata = device->device_data; + + sst_dbg("called\n"); + snd_card_free(intelmaddata->card); + genl_unregister_family(&audio_event_genl_family); + unregister_sst_card(intelmaddata->sstdrv_ops); + + /*free allocated memory for internal context*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + return 0; +} + +/** +* snd_intelmad_create- called from probe to create a snd device +*@intelmaddata : pointer to the internal context +*@card : pointer to the sound card +* This function is called when driver module is started +*/ +static int __devinit snd_intelmad_create( + struct snd_intelmad *intelmaddata, + struct snd_card *card) +{ + int ret_val = 0; + static struct snd_device_ops ops = { + .dev_free = snd_intelmad_dev_free, + }; + + WARN_ON(!intelmaddata); + WARN_ON(!card); + /*ALSA api to register for the device*/ + ret_val = snd_device_new(card, SNDRV_DEV_LOWLEVEL, intelmaddata, &ops); + return ret_val; +} + +/********************************************************************* + * SPI Functions + *********************************************************************/ + + +/** +* snd_intelmad_probe- function registred for init +*@spi : pointer to the spi device context +* This function is called when the device is initialized +*/ +int __devinit snd_intelmad_probe(struct spi_device *spi) +{ + struct snd_card *card = NULL; + int ret_val = 0; + struct snd_intelmad *intelmaddata = NULL; + + sst_dbg("called \n"); + + /*allocate memory for saving internal context and working*/ + intelmaddata = kzalloc(sizeof(*intelmaddata), GFP_KERNEL); + if (NULL == intelmaddata) + return -ENOMEM; + + /*allocate memory for LPE API set*/ + intelmaddata->sstdrv_ops = kzalloc(sizeof(struct intel_sst_card_ops), + GFP_KERNEL); + if (NULL == intelmaddata->sstdrv_ops) { + sst_err("mem alloctn fail \n"); + kfree(intelmaddata); + return -ENOMEM; + } + + /*create a card instance with ALSA framework*/ + card = snd_card_new(card_index, card_id, THIS_MODULE, 0); + if (NULL == card) + return -ENOMEM; + + intelmaddata->spi = spi; + intelmaddata->irq = spi->irq; + dev_set_drvdata(&spi->dev, intelmaddata); + intelmaddata->card = card; + intelmaddata->card_id = card_id; + intelmaddata->card_index = card_index; + strncpy(card->driver, INTEL_MAD, strlen(INTEL_MAD)); + strncpy(card->shortname, INTEL_MAD, strlen(INTEL_MAD)); + +#ifdef REG_IRQ + ret_val = snd_intelmad_register_irq(intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_register_irq fail\n"); + goto free_allocs; + } + ret_val = snd_intelmad_register_netlink(); + if (0 == ret_val) { + sst_dbg("...complete\n"); + return ret_val; + } +#endif + + /*internal function call to register device with ALSA*/ + ret_val = snd_intelmad_create(intelmaddata, card); + if (0 != ret_val) { + sst_err("snd_intelmad_create failed\n"); + goto free_allocs; + } + + ret_val = snd_intelmad_pcm(card, intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_pcm failed\n"); + goto free_allocs; + } + + ret_val = snd_intelmad_mixer(intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_mixer failed\n"); + goto free_allocs; + } + + card->private_data = &intelmaddata; + snd_card_set_dev(card, &spi->dev); + ret_val = snd_card_register(card); + if (0 != ret_val) + goto free_allocs; + + + intelmaddata->sstdrv_ops->module_name = SST_CARD_NAMES; + /*registering with LPE driver to get access to LPE APIsito use*/ + ret_val = snd_intelmad_sst_register(intelmaddata); + if (0 == ret_val) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + sst_dbg("...complete\n"); + return ret_val; + } + + sst_err("registering with LPE failed \n"); + +free_allocs: + /*TODO: unregister IRQ*/ + sst_err("probe failed\n"); + /*snd_card_free(card);*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + return ret_val; +} + + +/** +* snd_intelmad_remove- function registred for exit +*@spi : pointer to the spi device context +* This function is called when the device is uninitialized +*/ +static int snd_intelmad_remove(struct spi_device *spi) +{ + struct snd_intelmad *intelmaddata = + dev_get_drvdata(&spi->dev); + /* + * TODO:: de-register interrupt handler + */ + + if (intelmaddata) { + snd_card_free(intelmaddata->card); + genl_unregister_family(&audio_event_genl_family); + unregister_sst_card(intelmaddata->sstdrv_ops); + /*free allocated memory for internal context*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + } + return 0; +} + +/********************************************************************* + * Driver initialization and exit + *********************************************************************/ + +static struct spi_driver snd_intelmad_driver = { + .driver = { + .name = DRIVER_NAME, + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = snd_intelmad_probe, + .remove = __devexit_p(snd_intelmad_remove), +}; + +/* +* alsa_card_intelmad_init- driver init function +* This function is called when driver module is inserted +*/ +static int __init alsa_card_intelmad_init(void) +{ + sst_dbg("called\n"); + return spi_register_driver(&snd_intelmad_driver); +} + +/** +* alsa_card_intelmad_exit- driver exit function +* This function is called when driver module is removed +*/ +static void __exit alsa_card_intelmad_exit(void) +{ + sst_dbg("called\n"); + spi_unregister_driver(&snd_intelmad_driver); +} + +module_init(alsa_card_intelmad_init) +module_exit(alsa_card_intelmad_exit) + -- 1.5.4.5

6 31

[alsa-devel] [PATCH 1/1] pcm_hw: Always use delay ioctl in snd_pcm_delay()
by Kai Vehmanen 11 Sep '09

11 Sep '09

As the result of snd_pcm_delay() is affected not only by hw_ptr and appl_ptr, but also by 'runtime->delay' property, either SNDRV_PCM_IOCTL_DELAY or SNDRV_PCM_IOCTL_STATUS ioctl must be used to get the correct result. Previously 'runtime->delay' was ignored in case 'hw->sync_ptr' was used. Signed-off-by: Kai Vehmanen <kvehmanen(a)eca.cx> --- src/pcm/pcm_hw.c | 22 ---------------------- 1 files changed, 0 insertions(+), 22 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c46d14f..8abb204 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -507,28 +507,6 @@ static int snd_pcm_hw_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) { snd_pcm_hw_t *hw = pcm->private_data; int fd = hw->fd, err; - if (hw->sync_ptr) { - err = sync_ptr1(hw, SNDRV_PCM_SYNC_PTR_HWSYNC); - if (err < 0) - return err; - switch (FAST_PCM_STATE(hw)) { - case SNDRV_PCM_STATE_RUNNING: - case SNDRV_PCM_STATE_DRAINING: - case SNDRV_PCM_STATE_PAUSED: - case SNDRV_PCM_STATE_PREPARED: - case SNDRV_PCM_STATE_SUSPENDED: - break; - case SNDRV_PCM_STATE_XRUN: - return -EPIPE; - default: - return -EBADFD; - } - if (pcm->stream == SND_PCM_STREAM_PLAYBACK) - *delayp = snd_pcm_mmap_playback_hw_avail(pcm); - else - *delayp = snd_pcm_mmap_capture_avail(pcm); - return 0; - } if (ioctl(fd, SNDRV_PCM_IOCTL_DELAY, delayp) < 0) { err = -errno; SYSMSG("SNDRV_PCM_IOCTL_DELAY failed"); -- 1.5.6.5

1 0

[alsa-devel] How to support mutiple audio interface in alsa
by Feng Ye 10 Sep '09

10 Sep '09

Hello there, This question is for ALSA, not ALSA SOC. I am doing an ALSA driver for our chip (soc chip with cpu inside), inside it has I2S and DAC. The I2S can be feed to DAC, or come out of the chip to an external CODEC chip. I wonder how should I model this in the driver, are they multiple ALSA cards, or devices, or subdevices? Since the I2S is shared, I think the most appropriate is 'subdevice', but I do not see how this is used in linux/sound, this is only used in pci, and seems to tied together with substream number. If the answer is 'device', is there some existing driver that are doing this? Thanks, Feng

2 1

[alsa-devel] i.MX31L + AC97
by Geoffrey Wossum 10 Sep '09

10 Sep '09

Hi all, I'm currently looking at a board which uses a Freescale i.MX31L connected to a Wolfson WM9715L over the AC97 bus. The WM9715L will be used for both sound and a touchscreen. It doesn't look like there are any AC97 SoC drivers for the i.MX31. Is this correct? If so, should I try to get the board changed to use a codec on the I2S bus (the board is being layed out right now), or would creating the AC97 driver simple enough that we should continue on with this hardware and write the driver when needed? TIA, --- Geoffrey

2 1

[alsa-devel] Alsa plugin unable to be loaded by Pulseaudio
by Ng Oon-Ee 10 Sep '09

10 Sep '09

Hello, please find below an issue I had with loading plugins into pulseaudio. The pulse dev has sent this mail on his list concerning alsa's interaction with pulse, indicating that there is something buggy with how it works. Details below. I'd be willing to post something to the bugtracker, if I only knew what the below means... -------- Forwarded Message -------- From: Lennart Poettering <lennart(a)poettering.net> Reply-to: General PulseAudio Discussion <pulseaudio-discuss(a)mail.0pointer.de> To: pulseaudio-discuss(a)mail.0pointer.de Subject: Re: [pulseaudio-discuss] Alsa output devices, able to use alsa plugin? Date: Fri, 4 Sep 2009 01:20:15 +0200 On Wed, 26.08.09 12:27, Tanu Kaskinen (tanuk(a)iki.fi) wrote: > D: alsa-util.c: Trying test with SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying test without SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying plug:test with SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open plug:test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying plug:test without SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open plug:test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: Failed to set hardware parameters on plug:test: Tiedostoa tai hakemistoa ei ole > E: module.c: Failed to load module "module-alsa-sink" (argument: "device=test"): initialization failed. > > That "Tiedostoa tai hakemistoa ei ole" error means "No such file or > directory". I tried also with device=hw:0,0 and device=plughw:0,0 and > the first worked while the latter gave the same errors. Since the failed > function calls pertain to buffering, I also tried with tsched=0. It > worked. So, maybe this is a bug in the "plug" alsa plugin, ie. it > doesn't implement the *_near() functions properly. > > Lennart, any insights? I am pretty sure that ALSA is broken here. We nowadays first try to set the number of periods, and then set the buffer size in the hwparams struct. If this fails we print a warning message (the ones you see above with all the vowels regarding snd_pcm_hw_params_set_periods_near() resp. snd_pcm_hw_params_set_buffer_size_near()), but we do not consider this fatal. After doing this we call snd_pcm_hw_params() to activate our settings -- and if this fails then we consider this a fatal problem. Now the brokeness in ALSA is that after refusing our parameters it still fails to configure our settings, i.e. calling the two functions that fail breaks the hwparams setup: if we wouldn't call them, no breakage happens at all. The two functions in question should either succeed or fail, but if they fail they should not modify hwparams in a way that it subsequently becomes unusable for snd_pcm_hw_params(). This issue needs to be fixed in ALSA. Lennart

3 9

[alsa-devel] Problems with safe API and snd-cs46xx
by Sophie Hamilton 10 Sep '09

10 Sep '09

Hi there, I was pointed here by Audacious developers, who believe I've found a bug in the snd-cs46xx driver, which I use for my sound card (a Turtle Beach Santa Cruz, whose PCI ID is 1013:6003). Audacious, as of version 2.1, uses the default period time offered by ALSA and limits itself to the "safe API" as per Lennart Poettering. This already uncovered a bug in the Aureal Vortex driver, and I believe, based on the information that the devs gave me, that I have just uncovered another. (I'm also having problems with recent versions of OpenAL. However, I'll explain this further below.) I first started having the problem after having upgraded to Audacious 2.1, from Audacious 1.5.1. (to be more precise, the Gentoo ebuild 1.5.1-r1.) At the time, I was using the Gentoo-patched 2.6.29-series kernel, which I use as my normal kernel. After upgrading, I could no longer play audio directly using ALSA; instead, Audacious would initialise to 00:00, not play anything, and report the following to the owning console every so often: > ERROR: ALSA: alsa-core.c:226 (alsaplug_write_buffer): (write) snd_pcm_recover: Input/output error The thread which was doing this also seemed to be blocking, as when I tried to do anything else in Audacious, the interface would then freeze until the thread was done. OSS output seems to be okay, although as I don't have OSS compiled at all into the kernel (not even as a module), this would be going via ALSA's own OSS emulation. In addition, ALSA output to another audio device from Audacious works fine. I Googled for this and came across another snd-cs46xx user having the exact same problem in bug AUD-53 in the JIRA implementation they were using: http://jira.atheme.org/browse/AUD-53 . Since that user had been asked to upgrade to tip but hadn't, I did so instead (uninstalling the Gentoo ebuilds frst, of course) and reported back the additional debug info: > DEBUG: ALSA: alsa-core.c:226 (alsaplug_write_buffer): snd_pcm_writei error: Input/output error > ERROR: ALSA: alsa-core.c:230 (alsaplug_write_buffer): snd_pcm_recover error: Input/output error I then didn't know how to continue as I wasn't sure whether the bug would be seen as it had previously been closed as unreproducible, so I hopped onto the IRC channel and asked about what should be done. It was there that the probability of it being a driver bug in the snd-cs46xx sound driver was mentioned, and the reasoning - that the use of ALSA's safe API has been exposing driver bugs that were previously hidden. As mentioned previously, I'm also having issues with OpenAL. The version I'm using now is 1.7.411; previously I was using 1.5.304, which I believe worked fine. (I can't easily test this now as the Gentoo ebuild for it has gone, however.) The symptoms I got were similar - I got no sound, and messages were printed to the console every so often (more often than with Audacious, however): > AL lib: alsa.c:194: Wait timeout... buffer size too low? At first I thought this was an issue with Second Life, but on seeing the exact same symptoms using mplayer with the '-ao openal' switch, and also seeing it occur with 'openal-info', it seems to be an OpenAL thing. (Since I don't have many apps that I use with OpenAL, I hadn't known this before.) It seems that this is a symptom of whatever driver bug is causing the Audacious error. For the purposes of making this post, I decided to compile and install a vanilla 2.6.30.5 kernel to boot into, so that I could make sure there were no problems with the latest stable build. (I'd rather not run unstable versions if I can help it, or I would have done so; this is my primary machine. However, I'm willing to apply patches for this specific problem in order to test them.) The results of this were that I no longer get the messages and the threads don't block, but I get entirely new problems now - instead of useable sound, I get what sounds a little like noise, but more crackly, and the seek bar in Audacious seems to be going along as fast as it can - for each wallclock second, the seek bar goes up by about 114 seconds. The same occurs when I use mplayer with the '-ao openal' switch; I get the noise/crackles, and the time position shown on the console zooms by. The description of my sound card given by 'lspci -vv' (under both kernels) is thus: > 05:04.0 Multimedia audio controller: Cirrus Logic CS 4614/22/24/30 [CrystalClear SoundFusion Audio Accelerator] (rev 01) > Subsystem: Voyetra Technologies Santa Cruz > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: 64 (1000ns min, 6000ns max) > Interrupt: pin A routed to IRQ 22 > Region 0: Memory at fdafe000 (32-bit, non-prefetchable) [size=4K] > Region 1: Memory at fd900000 (32-bit, non-prefetchable) [size=1M] > Capabilities: [40] Power Management version 2 > Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) > Status: D0 PME-Enable- DSel=0 DScale=0 PME- > Kernel driver in use: Sound Fusion CS46xx > Kernel modules: snd-cs46xx Most applications that I have seem to work properly with regards to audio, although I've heard that this is probably because ALSA's safe API isn't very well documented and as such most applications will not be using it, thus not exposing the bug in snd-cs46xx. No information is output to dmesg when this happens, on either the Gentoo-patched 2.6.29 kernel, or the vanilla 2.6.30.5 kernel. If you need any further information, please let me know! As stated above, I'm willing to apply patches to the vanilla sources of the latest stable kernel in order to test them. I don't have experience of downloading the ALSA drivers and compiling them separately but I could also do that if it's needed. Thank you, - Sophie.

6 32

Re: [alsa-devel] [PATCH] To avoid the divide by zero error during the first execution, initialize the data type.
by Troy Kisky 10 Sep '09

10 Sep '09

avm(a)ti.com wrote: > From: Arun Mani <a0270733(a)gtcx26221.gt.design.ti.com> > > Signed-off-by: Arun Mani <a0270733(a)gtcx26221.gt.design.ti.com> > --- > sound/soc/davinci/davinci-i2s.c | 2 ++ > 1 files changed, 2 insertions(+), 0 deletions(-) > > diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c > index b1ea52f..61b1763 100644 > --- a/sound/soc/davinci/davinci-i2s.c > +++ b/sound/soc/davinci/davinci-i2s.c > @@ -104,10 +104,12 @@ enum { > > static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { > .name = "I2S PCM Stereo out", > + .data_type = 2, //Initialize the data type for playback to avoid divide by zero You should have ".acnt = 2" also for consistency. > }; > > static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { > .name = "I2S PCM Stereo in", > + .data_type = 2, //Initialize the data type for playback to avoid divide by zero > }; > > struct davinci_mcbsp_dev { And if the 1st stream is an 8-bit audio stream, will it initialize it incorrectly ??? Of course, we have formats = SNDRV_PCM_FMTBIT_S16_LE, currently so it's not an issue yet... I don't see how data_type is not being set in davinci_i2s_hw_params before being used in davinci_pcm_prepare. Can prepare be called before hw_params ? There is a "return -EINVAL" in davinci_i2s_hw_params with a "printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n")" before it. Do you see this message in your log ? Thanks Troy

2 2

[alsa-devel] [PATCH v3] ASoC: AK4671: add ak4671 codec driver
by Joonyoung Shim 10 Sep '09

10 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> --- Change log: v2: - use soc-cache - remove unused pll_on variable - add event handler for DAC and ADC - remove unnecessary comments and PM functions - modify error handling in ak4671_register function v3: - use SND_SOC_DAPM_SUPPLY instead of SND_SOC_DAPM_DAC_E and SND_SOC_DAPM_ADC_E sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4671.c | 825 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 +++++++++ 4 files changed, 987 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..b61214d --- /dev/null +++ b/sound/soc/codecs/ak4671.c @@ -0,0 +1,825 @@ +/* + * 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]; +}; + +/* 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) */ +}; + +/* + * 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("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("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 = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg |= AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg &= ~AK4671_MUTEN; + snd_soc_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), + + /* Supply */ + SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DAC Left", "NULL", "PMPLL"}, + {"DAC Right", "NULL", "PMPLL"}, + {"ADC Left", "NULL", "PMPLL"}, + {"ADC Right", "NULL", "PMPLL"}, + + /* 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 = snd_soc_read(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; + } + + snd_soc_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 = snd_soc_read(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; + } + + snd_soc_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; + u8 mode; + u8 format; + + /* set master/slave audio interface */ + mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode |= AK4671_M_S; + break; + case SND_SOC_DAIFMT_CBM_CFS: + mode &= ~(AK4671_M_S); + break; + default: + return -EINVAL; + } + + /* interface format */ + format = snd_soc_read(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 */ + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode); + snd_soc_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) +{ + u8 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, + reg | AK4671_PMVCM); + break; + case SND_SOC_BIAS_OFF: + snd_soc_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, +}; + +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; +} + +struct snd_soc_codec_device soc_codec_dev_ak4671 = { + .probe = ak4671_probe, + .remove = ak4671_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671); + +static int ak4671_register(struct ak4671_priv *ak4671, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &ak4671->codec; + + if (ak4671_codec) { + dev_err(codec->dev, "Another AK4671 is registered\n"); + ret = -EINVAL; + goto err; + } + + 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->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)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + 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); + goto err; + } + + ret = snd_soc_register_dai(&ak4671_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(ak4671); + return ret; +} + +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, SND_SOC_I2C); +} + +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_modinit(void) +{ + return i2c_add_driver(&ak4671_i2c_driver); +} +module_init(ak4671_modinit); + +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

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[alsa-devel] Is there a way to figure out if the clocks of two snd_pcm_t's deviate? (snd_pcm_info_get_sync?)
by Lennart Poettering 09 Sep '09

09 Sep '09

Heya! I am looking for a way to find out whether the clocks of two snd_pcm_t's are perfectly in sync and don't deviate (or only deviate by a constant phase). i.e. for real duplex devices that use the same crystal for input and output this should tell me "Yes" for the playback snd_pcm_t tested against the recording snd_pcm_t; and for all other, unrelated audio devices it should return "No". I found snd_pcm_info_get_sync() in the docs, but its documentation is a bit terse, so I am wondering if this might be what I want. Is it? Does this (untested) code snippet do what I hope it does? <snip> snd_pcm_info(a, info_a); snd_pcm_info(b, info_b); in_sync = memcmp(&snd_pcm_info_get_sync(info_a), &snd_pcm_info_get_sync(info_b)) == 0; </snip> Thanks, Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

4 6

[alsa-devel] pts value using alsa for audio capture
by Nicolas Krieger 09 Sep '09

09 Sep '09

Hi, I have a question about how pts are computed using alsa while grabbing audio. I take an example to explain : I do grab audio during 10 seconds, the first 2 seconds, there is no problem, so the pts are like that (I take examples values) : 1 2 3 4 5 6 .... x <- x is the value of the last pts (2nd second). Then, for any reason, no capture during 3 seconds. And then no problem until the end. Nicolas Krieger

1 1

Re: [alsa-devel] [PATCH 2/3] ASoC: DaVinci: pcm, rename variables in prep for ping/pong
by Nori, Sekhar 09 Sep '09

09 Sep '09

On Wed, Sep 09, 2009 at 18:03:58, Caglar Akyuz wrote: > On Wednesday 09 September 2009 15:08:42 Nori, Sekhar wrote: [...] > > > > However, with or without your patch, I noticed a segmentation fault for the > > first time the 'arecord | aplay' loop is created. There is no fault the > > second time around. I haven't spent time debugging this yet. > > > > This is due to these lines in davinci_pcm_enqueue_dma: > > data_type = prtd->params->data_type; > count = period_size / data_type; > > First time running data_type is set to 0. This is a long time issue, I > remember modifying my kernels to check for zero before this division.. Ah, thanks! Planning to submit a fix for this? We can work on the patch if you are busy otherwise. Regards, Sekhar

3 2

[alsa-devel] Layla 3G does not recover from ACPI Suspend
by Mark Hills 09 Sep '09

09 Sep '09

I have an Echo Layla 3G in my workstation. It works for audio, but does not recover from ACPI suspend to RAM. On recovery the system is fine, and the Layla exists in /proc/asound/cards. But when the Layla is used it prints this message to dmesg, multiple times: wait_handshake(): Timeout waiting for DSP Here are the relevant dmesg lines after awakening: Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0xf (was 0x100, writing 0x104) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x4 (was 0x0, writing 0xe7d00000) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x3 (was 0x0, writing 0xc010) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x1 (was 0x2800000, writing 0x2800112) Echoaudio Echo3G 0000:03:03.0: PCI INT A -> GSI 19 (level, low) -> IRQ 19 I couldn't see anything loading the firmware. I thought unloading and loading snd-echo3g after recovery would help. This shows the firmware being loaded, but then no ALSA device is shown in /proc/asound/cards. On loading snd-echo3g: Echoaudio Echo3G 0000:03:03.0: PCI INT A -> GSI 19 (level, low) -> IRQ 19 Echoaudio Echo3G 0000:03:03.0: firmware: requesting ea/echo3g_dsp.fw Echoaudio Echo3G 0000:03:03.0: firmware: requesting ea/loader_dsp.fw Echoaudio Echo3G 0000:03:03.0: PCI INT A disabled Echoaudio Echo3G: probe of 0000:03:03.0 failed with error -5 Is the firmware not being loaded when it should be? Or is there some extra initialisation not being done (eg. init_hw() in echo3g_dsp.c)? All other devices in the system recover fine, but as far as I'm aware no other PCI devices require firmware. It's a Dell x86 system, kernel 2.6.31-rc7. Thanks for any help, -- Mark

3 3

[alsa-devel] [PATCH v2] ASoC: au1x: PSC-AC97 bugfixes
by Manuel Lauss 08 Sep '09

08 Sep '09

This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- The previous patch missed a git add for psc.h, sorry! sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ sound/soc/au1x/psc.h | 1 + 2 files changed, 97 insertions(+), 33 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss(a)gmail.com>"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a0..3f474e8 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -29,6 +29,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; + struct mutex lock; }; #define PCM_TX 0 -- 1.6.4.2

3 3

[alsa-devel] ASoC update for 2.6.32
by Mark Brown 08 Sep '09

08 Sep '09

The following changes since commit 87831cb660954356d68cebdb1406f3be09e784e9: Mark Brown (1): ASoC: Fix WM835x Out4 capture enumeration 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 Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ sound/soc/au1x/psc.h | 1 + 2 files changed, 97 insertions(+), 33 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: au1x: PSC-AC97 bugfixes
by Manuel Lauss 08 Sep '09

08 Sep '09

This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ 1 files changed, 96 insertions(+), 33 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss(a)gmail.com>"); -- 1.6.4.2

1 0

[alsa-devel] Update to problem with Creative X-fi mixer problems
by Susan Cragin 08 Sep '09

08 Sep '09

I saw that someone was going to try to get all the mixer controls to work properly. I updated and opened alsamixer this morning, and found that both the incoming/recording controls were there! (I should have quit while I was ahead.) Then I re-booted and recompiled wine and opened winecfg and found that oss could find the proper mixer but alsa did not. (The oss mixer is listed as Creative CA0110-IBG) ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm_dsnoop.c:593:(snd_pcm_dsnoop_open) unable to open slave fixme:mixer:ALSA_MixerInit No master control found on HD-Audio Generic, disabling mixer susan@ubuntu:~$ alsamixer No mixer elems found

1 0

Re: [alsa-devel] snd_dummy on Centos - Redhat
by Takashi Iwai 08 Sep '09

08 Sep '09

At Fri, 4 Sep 2009 00:43:09 +0200, Giovanni Maruzzelli wrote: > > 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.... OK, then upgrade your alsa driver stack to 1.0.21. There have been some changes in PCM core. Then, get dummy.c again from the latest alsa-driver-unstable tarball. It already contains my latest fix. If 1.0.21+dummy.c doesn't work, apply the patch below to get position logs. If this still causes the invalid backward jump to position 0, you should check xrun debug, e.g. # echo 9 > /proc/asound/card0/pcm0p/xrun_debug (assuming snd-dummy is card0). Takashi --- diff --git a/dummy.c b/dummy.c index e8e29bf..f7c21d5 100644 --- a/dummy.c +++ b/dummy.c @@ -239,6 +239,10 @@ static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) dpcm->frac_period_rest += dpcm->frac_period_size; } dpcm->frac_period_rest -= delta; + printk(KERN_DEBUG "xxx delta=%ld, pos=%d(%d), rest=%d(%d)\n", + delta / dpcm->rate, + dpcm->frac_pos, dpcm->frac_pos / HZ, + dpcm->frac_period_rest, dpcm->frac_period_rest / HZ); } static int dummy_systimer_start(struct snd_pcm_substream *substream) @@ -248,6 +252,10 @@ static int dummy_systimer_start(struct snd_pcm_substream *substream) dpcm->base_time = jiffies; dummy_systimer_rearm(dpcm); spin_unlock(&dpcm->lock); + printk(KERN_DEBUG "*** START: buffer=%d, period=%d, rate=%d\n", + (int)substream->runtime->buffer_size, + (int)substream->runtime->period_size, + substream->runtime->rate); return 0; } @@ -257,6 +265,7 @@ static int dummy_systimer_stop(struct snd_pcm_substream *substream) spin_lock(&dpcm->lock); del_timer(&dpcm->timer); spin_unlock(&dpcm->lock); + printk(KERN_DEBUG "*** STOP\n"); return 0; }

2 24

[alsa-devel] Audio cutoff when playing using dmix
by ruchi sirauthiya 08 Sep '09

08 Sep '09

Hi, I am facing audio clipping issues when dmix is used for playback. For the test we used a full scale sine wave tone generated using software and encoded as "Signed 16 bit Little Endian, Rate 44100 Hz, Mono". The audio is getting cut out at a 2-3 second interval during playback. This happens only for a fraction of second but it is consistent. This does not happen when audio is played back on the device directly but only when going through dmix. We have observed that there were differences in audio cutting off when the sampling rate was set to 8000Hz and 32000 Hz in asound.conf. The contents of asound.conf is pasted below for your reference: pcm.dmixer { type dmix ipc_key 2048 ipc_key_add_uid true slave { pcm hw:0 rate 8000 channels 2 } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "hw:0,0" } pcm.rateConvert { type plug slave.pcm "asymed" } The configuration given above works fine with dmix in 2.6.29 kernel, I am using kernel 2.6.22.18. I have compared the configuration differences (buffers and periods) with and without using dmix. I tried applying some parameters (like buffer_size, period_time and period_size) which was in use when audio was played directly to the hardware device by aplay using the following command: aplay -D rateConvert --period-time=92879 --buffer-size=65536 --period-size=4096 /mnt/test.wav -vvv -d 5 There was come improvement in the audio cutting off issue but it was not eliminated completely. Given below are the logs for aplay (-vvv) with and without using dmix (with audio cutoff issue on kernel 2.6.22): Without using dmix: [root@OMAP3 tests]# aplay /mnt/test.wav -vvv -d 5 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 65536 period_size : 4096 period_time : 92879 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 65536 stop_threshold : 65536 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% ...Same text repeats 50 times... Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% Using dmix: [root@OMAP3 tests]# aplay -D rateConvert /mnt/test.wav -vvv -d 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Rate conversion PCM (8000, sformat=S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 5512 xfer_align : 5512 start_threshold : 22048 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 1445068800 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 16000 period_size : 1000 period_time : 125000 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 1 stop_threshold : 2097152000 silence_threshold: 0 silence_size : 2097152000 boundary : 2097152000 Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% ...Same text repeats 36 times... Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00005f1a ############### 74% Please help me fix this issue. Does this occur due to non-optimal settings of the parameters (buffers, periods) in the configuration or is it some other issue with the ALSA library or the kernel? Please ask if any more setup/configuration settings are required for reference. Thanks and Regards, Ruchi Sirauthiya

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 08 Sep '09

08 Sep '09

The following changes since commit 367da1527aff2a4f789d03082f9fb191cdf332c4: Robert Schwebel (1): ASoC: fix pxa2xx-ac97.c breakage 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 Mark Brown (2): ASoC: Remove unuused hw_read_t ASoC: Fix WM835x Out4 capture enumeration include/sound/soc.h | 1 - sound/soc/codecs/wm8350.c | 2 +- 2 files changed, 1 insertions(+), 2 deletions(-)

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Re: [alsa-devel] Patch for au88x0 driver. Fix audacious issue and add auto reset in case of hardware ADB ctrl failure.
by Raymond Yau 08 Sep '09

08 Sep '09

2009/9/7 Manuel Jander <manuel.jander(a)gmail.com> > Hi Raymond, > > On Mon, Sep 7, 2009 at 3:47 AM, Raymond Yau<superquad.vortex2(a)gmail.com> > wrote: > > On Thursday 03 September 2009 21:10, Manuel Jander wrote: > >> 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<http://micro.homelinux.net/%7Emjander/au88x0.diff.gz>< > http://micro.homelinux.net/%7Emjander/au88x0.diff.gz> > >> > >> Best Regards, > >> Manuel > > > > Which version of audaicous (1.5.x or 2.1) are you testing ? > > (alsa-gapless or alsa-ng plugin) > > > > au88x0 most likely cannot support 1 sample per period because of the > > FIFO as described in patent 6167465. > > Well, that is clear to me now. Using a DMA transfer that is smaller > than one audio sample is not really a good idea. > I would be better off using PIO instead of DMA. > > > if you look at the source code of alsa-ng and alsa-gapless plugin > > both of them use the own way to calculate the buffer size instead of > > using snd_pcm_hw_params_get_buffer_size() > > > > > > > http://hg.atheme.org/audacious-plugins/file/2c0a9bba77d7/src/alsa-ng/alsa-c… > > > > CHECK_FAIL (snd_pcm_hw_params_set_access (pcm_handle, hwparams, > > SND_PCM_ACCESS_RW_INTERLEAVED), "snd_pcm_hw_params_set_access"); > > CHECK_FAIL (snd_pcm_hw_params_set_format (pcm_handle, hwparams, afmt), > > "snd_pcm_hw_params_set_format"); > > CHECK_FAIL (snd_pcm_hw_params_set_channels (pcm_handle, hwparams, nch), > > "snd_pcm_hw_params_set_channels"); > > CHECK_FAIL (snd_pcm_hw_params_set_rate (pcm_handle, hwparams, rate, 0), > > "snd_pcm_hw_params_set_rate"); > > CHECK_FAIL (snd_pcm_hw_params (pcm_handle, hwparams), > "snd_pcm_hw_params"); > > bitwidth = snd_pcm_format_physical_width(afmt); > > bps = (rate * bitwidth * nch) >> 3; > > buf_size = MAX(aud_cfg->output_buffer_size, 100); > > ringbuf_size = buf_size * bps / 1000; > > I increased the minimum period bytes value to 256 in that patch. > Audacious version 2 (Debian SID) > works for me. > well, audacious.i386 1.5.1-9.fc10 cannot play any music (alsa plugin) on my au8830 with your patch, it only play music with (oss plugin) Use xmms instead of audacious since they have similar UI 256 seem much bigger than the figures mentioned in the patent , may be it is the practical limit on your VIA ITX In one embodiment of the present invention, the FIFO buffer is 32 samples or words in depth, which would withstand a maximum bus latency of about 4 microseconds. By way of example, if 48 DMA channels trigger simultaneously, with a trigger level set at half the buffer size (16 samples), each DMA channel can wait 16 samples (20 microseconds per sample), or approximately 320 microseconds before running out of data. By way of example, CPU 210 may update-sub-buffer tables 380(0)-380(x) routinely every 10 milliseconds. The periodic checks may be initiated by audio processing peripheral 230 by way of interrupts, or may be scheduled independently by the driver > > How do you get the "ADB Ctrl failure" error ? > > I do not know how to reproduce that. Sometimes it happens, and its not > possible > to recover without a hardware reset (unload and load module). Somehow > it semes to > happen when my computer gets really hot (watching an MPEG-4 video, or > compiling the kernel...). > Its a old VIA ITX board, with a 1.3GHz processor (on demand cpufreq). > > The patch tries to reset the audio card if all PCM streams are > released once such a > condition was detected. This is the situation were the windows driver > used to crash the > whole machine. > it work quite well on my two motherbroad using intel chipset > > There are two main problem of the au88x0 driver > > 1) alsa is unable to calculate the correct value for the three > > functions snd_pcm_hw_params_set_buffer_time_near() , > > snd_pcm_hw_params_set_period_time_near() and snd_pcm_set_params() > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1724 > > > > Test case : > > > > alsa-lib/test/pcm_min.c ( change default to hw:0,0 ) > > > > Real world case: > > > > gstreamer and winealsa.drv > > I will try that. I had no idea about that bugs. > A simple workaround is add another constraint snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096 ); however winealsa.drv will be using a larger period bytes than wineoss.drv > > > 2) unable to recover when the number of available src is not enough > > (FATAL: ResManager: resource type 1 exhaust) > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2228 > > > > test case : > > > > running application to use 15 src ( 7 stereo and one mono stream ) and > > open another stereo stream > > Hmm, the good ol' resource manager :) OK, I will try to find a > solution. I have not much > time these days, but maybe it rains again, and I get bored, who knows :D > > Best Regards, > Manuel > simple fix may be adding counts of the resource and check that all resources are enough before allocating them

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[alsa-devel] Duplicate configuration nodes
by Jochem Wichers Hoeth 07 Sep '09

07 Sep '09

Hi, Since I upgraded from alsa-lib 1.0.20 to 1.0.21 my configuration is not parsed correctly anymore: $ aplay test.wav ALSA lib conf.c:4639:(snd_config_expand) Expand error (walk): File exists Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono As a result of this error, ALSA seems to ignore the second half of HDA-Intel.conf, including the iec958 definition. I traced the "File exists" error back to snd_config_add() returning -EEXIST, while evaluating the "strings" node at line 36 of cards/HDA-Intel.conf. I verified that at that point the src node indeed has four children: two duplicates of each child node of the original strings array. I'm not exactly sure what's happening here, but I do see that the HDA-Intel.conf file (and all of its included files) are actually read *twice* by aplay. With alsa-lib 1.0.20 aplay only reads these files once. The problem seems to be related to commit 96da0c842d14b40ce8e37726b259229634b3aa21 ("Fix driver conf parsing in snd_config_hook_load_for_all_cards()"), which subtly changes the logic when the same driver is encountered for the second time. I have two cards using the HDA-Intel config (one analog and one digital). For the first card, the config is loaded without issues. But for the second card, snd_config_search(root, fdriver, &n) succeeds but snd_config_get_string(n, &driver) fails. In 1.0.20 this results in a "goto __err", skipping the actual config load. But in 1.0.21 this results in the else-branch being chosen (driver = fdriver) and next snd_config_hook_load being called. I know too little of the entire config loading process to be sure what exactly is going wrong here, but a assume that the second call to snd_config_hook_load for the same configaration file somehow eventually results in the duplicate node error in snd_config_add(). I've verified that reverting 96da0c842d14b40ce8e37726b259229634b3aa21 fixes the problem, or at least the symptoms. :-) Please let me know if you need extra information, of if you'd like me to do some more testing. Cheers, Jochem Wichers Hoeth

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[alsa-devel] [PATCH] Minor WM8580 enhancements.
by jassi brar 07 Sep '09

07 Sep '09

Add more options to select clock source for various blocks in WM8580. Also, explicitly set defaults for the sake of consistency. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/codecs/wm8580.c | 133 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8580.h | 25 +++++---- 2 files changed, 144 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..276bc8e 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -95,6 +95,7 @@ /* PLLB4 (register 7h) */ #define WM8580_PLLB4_MCLKOUTSRC_MASK 0x60 +#define WM8580_PLLB4_MCLKOUTSRC_MCLK 0x00 #define WM8580_PLLB4_MCLKOUTSRC_PLLA 0x20 #define WM8580_PLLB4_MCLKOUTSRC_PLLB 0x40 #define WM8580_PLLB4_MCLKOUTSRC_OSC 0x60 @@ -106,9 +107,16 @@ /* CLKSEL (register 8h) */ #define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_MCLK 0x00 #define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 #define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_ADC_CLKSEL_MASK 0x0c +#define WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK 0x00 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLA 0x04 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLB 0x08 +#define WM8580_CLKSEL_ADC_CLKSEL_MCLK 0x0c + /* AIF control 1 (registers 9h-bh) */ #define WM8580_AIF_RATE_MASK 0x7 #define WM8580_AIF_RATE_128 0x0 @@ -121,15 +129,16 @@ #define WM8580_AIF_BCLKSEL_MASK 0x18 #define WM8580_AIF_BCLKSEL_64 0x00 -#define WM8580_AIF_BCLKSEL_128 0x08 -#define WM8580_AIF_BCLKSEL_256 0x10 +#define WM8580_AIF_BCLKSEL_32 0x08 +#define WM8580_AIF_BCLKSEL_16 0x10 #define WM8580_AIF_BCLKSEL_SYSCLK 0x18 #define WM8580_AIF_MS 0x20 #define WM8580_AIF_CLKSRC_MASK 0xc0 +#define WM8580_AIF_CLKSRC_ADCMCLK 0x00 #define WM8580_AIF_CLKSRC_PLLA 0x40 -#define WM8580_AIF_CLKSRC_PLLB 0x40 +#define WM8580_AIF_CLKSRC_PLLB 0x80 #define WM8580_AIF_CLKSRC_MCLK 0xc0 /* AIF control 2 (registers ch-eh) */ @@ -604,12 +613,13 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div) { case WM8580_CLKSRC_MCLK: - /* Input */ + reg |= WM8580_PLLB4_MCLKOUTSRC_MCLK; break; case WM8580_CLKSRC_PLLA: reg |= WM8580_PLLB4_MCLKOUTSRC_PLLA; break; + case WM8580_CLKSRC_PLLB: reg |= WM8580_PLLB4_MCLKOUTSRC_PLLB; break; @@ -630,6 +640,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div) { case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_DAC_CLKSEL_MCLK; break; case WM8580_CLKSRC_PLLA: @@ -646,6 +657,59 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_CLKSEL, reg); break; + case WM8580_ADC_CLKSEL: + reg = snd_soc_read(codec, WM8580_CLKSEL); + reg &= ~WM8580_CLKSEL_ADC_CLKSEL_MASK; + + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_MCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLB; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_CLKSEL, reg); + break; + + case WM8580_PAIF_CLKSEL: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_CLKSRC_MASK; + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_AIF_CLKSRC_ADCMCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_AIF_CLKSRC_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_AIF_CLKSRC_PLLB; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_AIF_CLKSRC_MCLK; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + case WM8580_CLKOUTSRC: reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; @@ -672,6 +736,67 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_PLLB4, reg); break; + case WM8580_MCLKRATIO: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_RATE_MASK; + switch(div) { + case 128: + reg |= WM8580_AIF_RATE_128; + break; + + case 192: + reg |= WM8580_AIF_RATE_192; + break; + + case 256: + reg |= WM8580_AIF_RATE_256; + break; + + case 384: + reg |= WM8580_AIF_RATE_384; + break; + + case 512: + reg |= WM8580_AIF_RATE_512; + break; + + case 768: + reg |= WM8580_AIF_RATE_768; + break; + + case 1152: + reg |= WM8580_AIF_RATE_1152; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + + case WM8580_BCLKRATIO: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_BCLKSEL_MASK; + switch(div) { + case 64: + reg |= WM8580_AIF_BCLKSEL_64; + break; + + case 32: + reg |= WM8580_AIF_BCLKSEL_32; + break; + + case 16: + reg |= WM8580_AIF_BCLKSEL_16; + break; + + default: + reg |= WM8580_AIF_BCLKSEL_SYSCLK; + break; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + default: return -EINVAL; } diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 0dfb5dd..0ecdbfa 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -18,16 +18,21 @@ #define WM8580_PLLA 1 #define WM8580_PLLB 2 -#define WM8580_MCLK 1 -#define WM8580_DAC_CLKSEL 2 -#define WM8580_CLKOUTSRC 3 - -#define WM8580_CLKSRC_MCLK 1 -#define WM8580_CLKSRC_PLLA 2 -#define WM8580_CLKSRC_PLLB 3 -#define WM8580_CLKSRC_OSC 4 -#define WM8580_CLKSRC_NONE 5 - +#define WM8580_MCLK 1 +#define WM8580_DAC_CLKSEL 2 +#define WM8580_ADC_CLKSEL 3 +#define WM8580_PAIF_CLKSEL 4 +#define WM8580_CLKOUTSRC 5 +#define WM8580_MCLKRATIO 6 +#define WM8580_BCLKRATIO 7 + +#define WM8580_CLKSRC_ADCMCLK 0 +#define WM8580_CLKSRC_MCLK 1 +#define WM8580_CLKSRC_PLLA 2 +#define WM8580_CLKSRC_PLLB 3 +#define WM8580_CLKSRC_OSC 4 +#define WM8580_CLKSRC_NONE 5 + #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 -- 1.6.2.5

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