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[alsa-devel] proper use of snd_mixer_selem_set_capture_switch ?
by Aaron J. Grier 05 Sep '08

05 Sep '08

does anyone have a good example or walkthrough how to change capture sources? I'm having a heck of a time. my calls go like this: /* setup */ snd_mixer_selem_id_alloca snd_mixer_open snd_mixer_attach snd_mixer_selem_register snd_mixer_load /* set up PCM element, unused in this example */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* set up Line element (linecapture) */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* set up Mic element (miccapture) */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* here's that part that throws me... I'm trying to sit in a loop and * trade capture back and forth between Line and Mic. code reproduced * here verbatim... */ do { snd_mixer_selem_channel_id_t chn; for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++) { int junk; if (snd_mixer_selem_has_capture_channel(linecapture, chn)) { status = snd_mixer_selem_get_capture_switch(linecapture, chn, &junk); assert(0 == status); printf("line, chn %d, junk %d, ", chn, junk); status = snd_mixer_selem_set_capture_switch(linecapture, chn, lineflag); printf("set %d\n", lineflag); assert(0 == status); } if (snd_mixer_selem_has_capture_channel(miccapture, chn)) { status = snd_mixer_selem_get_capture_switch(miccapture, chn, &junk); printf("mic, chn %d, junk %d, ", chn, junk); assert(0 == status); status = snd_mixer_selem_set_capture_switch(miccapture, chn, micflag); printf("set %d\n", micflag); assert(0 == status); } } for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++) { if (snd_mixer_selem_has_capture_channel(linecapture, chn)) { status = snd_mixer_selem_get_capture_switch(linecapture, chn, &lineflag); printf("read line, chn %d: %d\n", chn, lineflag); assert(0 == status); } if (snd_mixer_selem_has_capture_channel(miccapture, chn)) { status = snd_mixer_selem_get_capture_switch(miccapture, chn, &micflag); printf("read mic, chn %d: %d\n", chn, micflag); assert(0 == status); } } if (lineflag) { lineflag = 0; micflag = 1; } else { lineflag = 1; micflag = 0; } getchar(); } while (1); I get output like this: line, chn 0, junk 0, set 1 mic, chn 0, junk 1, set 0 line, chn 1, junk 1, set 1 mic, chn 1, junk 0, set 0 read line, chn 0: 1 read mic, chn 0: 0 read line, chn 1: 1 read mic, chn 1: 0 [all is well, alsamixer shows LR capture on Line input, smash return and go for another loop] line, chn 0, junk 1, set 0 mic, chn 0, junk 0, set 1 line, chn 1, junk 1, set 0 mic, chn 1, junk 0, set 1 read line, chn 0: 1 read mic, chn 0: 1 read line, chn 1: 1 read mic, chn 1: 1 [here's where it goes funny. the "junk" is what I set last time, as expected, but the readbacks after the set_capture_switch indicate both line and mic are being captured? that's not right. alsamixer shows LR capture on Mic input. smack return again to see what happens.] line, chn 0, junk 1, set 0 mic, chn 0, junk 1, set 1 line, chn 1, junk 1, set 0 mic, chn 1, junk 1, set 1 read line, chn 0: 0 read mic, chn 0: 1 read line, chn 1: 1 read mic, chn 1: 1 [this gets really weird. we make the same write calls we did last time, and now we get back something different? alsamixer shows R Line and L Capture. screwey.] I feel I must be missing something obvious, like a flush operation, or something very simple. who knew changing a mux could be such an involved procedure? (= -- Aaron J. Grier | "Not your ordinary poofy goof." | agrier(a)poofygoof.com

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[alsa-devel] [PATCH 0/4] ARM: OMAP2: Add support for Gumstix Overo (resubmission)
by sakoman＠gmail.com 05 Sep '08

05 Sep '08

This patch series adds support for the Gumstix Overo board. Since it includes an ASoC driver I am copying both linux-omap and alsa-devel This version incorporates feedback from both linux-omap and alsa-devel

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[alsa-devel] [PATCH 1/1] ASoC codec: SSM2602 audio codec driver (v3)
by Bryan Wu 05 Sep '08

05 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> v2-v3: - add the codec to SND_SOC_ALL_CODECS - coding style fixing - rename registers' name - fix an issue with DAPM and the bias configuration. v1-v2: - coding style fixing - use pr_xxx macros to replace printk(KERN_XXX...) - use new-style i2c interface - update to use latest ASoC API Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2602.c | 775 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ssm2602.h | 130 ++++++++ 4 files changed, 911 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5d77dc3..2223993 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X + select SND_SOC_SSM2602 help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -92,3 +93,6 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate depends on I2C + +config SND_SOC_SSM2602 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 35daaa9..0cd55ee 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-ssm2602-objs := ssm2602.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o @@ -33,3 +34,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c new file mode 100644 index 0000000..d43459f --- /dev/null +++ b/sound/soc/codecs/ssm2602.c @@ -0,0 +1,775 @@ +/* + * File: sound/soc/codecs/ssm2602.c + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Tue June 06 2008 + * Description: Driver for ssm2602 sound chip + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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, see the file COPYING, or 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 <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 "ssm2602.h" + +#define AUDIO_NAME "ssm2602" +#define SSM2602_VERSION "0.1" + +struct snd_soc_codec_device soc_codec_dev_ssm2602; + +/* codec private data */ +struct ssm2602_priv { + unsigned int sysclk; + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* + * ssm2602 register cache + * We can't read the ssm2602 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { + 0x0017, 0x0017, 0x0079, 0x0079, + 0x0000, 0x0000, 0x0000, 0x000a, + 0x0000, 0x0000 +}; + +/* + * read ssm2602 register cache + */ +static inline unsigned int ssm2602_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == SSM2602_RESET) + return 0; + if (reg >= SSM2602_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write ssm2602 register cache + */ +static inline void ssm2602_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= SSM2602_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the ssm2602 register space + */ +static int ssm2602_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 ssm2602 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + ssm2602_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define ssm2602_reset(c) ssm2602_write(c, SSM2602_RESET, 0) + +/*Appending several "None"s just for OSS mixer use*/ +static const char *ssm2602_input_select[] = { + "Line", "Mic", "None", "None", "None", + "None", "None", "None", +}; + +static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum ssm2602_enum[] = { + SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select), + SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph), +}; + +static const struct snd_kcontrol_new ssm2602_snd_controls[] = { + +SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V, + 0, 127, 0), +SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V, + 7, 1, 0), + +SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), + +SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), +SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), + +SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), + +SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1), +SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0), + +SOC_ENUM("Capture Source", ssm2602_enum[0]), + +SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), +}; + +/* add non dapm controls */ +static int ssm2602_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Output Mixer */ +static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ssm2602_input_mux_controls = +SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]); + +static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, + &ssm2602_output_mixer_controls[0], + ARRAY_SIZE(ssm2602_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), +SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls), +SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1), +SND_SOC_DAPM_INPUT("MICIN"), +SND_SOC_DAPM_INPUT("RLINEIN"), +SND_SOC_DAPM_INPUT("LLINEIN"), +}; + +static const struct snd_soc_dapm_route audio_conn[] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, + + /* outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + + /* input mux */ + {"Input Mux", "Line", "Line Input"}, + {"Input Mux", "Mic", "Mic Bias"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, + {"Mic Bias", NULL, "MICIN"}, +}; + +static int ssm2602_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets, + ARRAY_SIZE(ssm2602_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return i; +} + +static int ssm2602_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + u16 srate; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; + int i = get_coeff(ssm2602->sysclk, params_rate(params)); + + /*no match is found*/ + if (i == ARRAY_SIZE(coeff_div)) + return -EINVAL; + + srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + ssm2602_write(codec, SSM2602_ACTIVE, 0); + ssm2602_write(codec, SSM2602_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + ssm2602_write(codec, SSM2602_IFACE, iface); + ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); + return 0; +} + +static int ssm2602_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + /* The DAI has shared clocks so if we already have a playback or + * capture going then constrain this substream to match it. + */ + if (ssm2602->master_substream) { + master_runtime = ssm2602->master_substream->runtime; + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + ssm2602->slave_substream = substream; + } else + ssm2602->master_substream = substream; + + return 0; +} + +static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + /* set active */ + ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); + + return 0; +} + +static void ssm2602_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + /* deactivate */ + if (!codec->active) + ssm2602_write(codec, SSM2602_ACTIVE, 0); +} + +static int ssm2602_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ssm2602_read_reg_cache(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; + if (mute) + ssm2602_write(codec, SSM2602_APDIGI, + mute_reg | APDIGI_ENABLE_DAC_MUTE); + else + ssm2602_write(codec, SSM2602_APDIGI, mute_reg); + return 0; +} + +static int ssm2602_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 ssm2602_priv *ssm2602 = codec->private_data; + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + ssm2602->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + ssm2602_write(codec, SSM2602_IFACE, iface); + return 0; +} + +static int ssm2602_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = ssm2602_read_reg_cache(codec, SSM2602_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + ssm2602_write(codec, SSM2602_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + ssm2602_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + ssm2602_write(codec, SSM2602_ACTIVE, 0); + ssm2602_write(codec, SSM2602_PWR, 0xffff); + break; + + } + codec->bias_level = level; + return 0; +} + +#define SSM2602_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 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + +struct snd_soc_dai ssm2602_dai = { + .name = "SSM2602", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2602_RATES, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2602_RATES, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .ops = { + .startup = ssm2602_startup, + .prepare = ssm2602_pcm_prepare, + .hw_params = ssm2602_hw_params, + .shutdown = ssm2602_shutdown, + }, + .dai_ops = { + .digital_mute = ssm2602_mute, + .set_sysclk = ssm2602_set_dai_sysclk, + .set_fmt = ssm2602_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(ssm2602_dai); + +static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2602_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(ssm2602_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ssm2602_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the ssm2602 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ssm2602_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int reg, ret = 0; + + codec->name = "SSM2602"; + codec->owner = THIS_MODULE; + codec->read = ssm2602_read_reg_cache; + codec->write = ssm2602_write; + codec->set_bias_level = ssm2602_set_bias_level; + codec->dai = &ssm2602_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(ssm2602_reg); + codec->reg_cache = kmemdup(ssm2602_reg, sizeof(ssm2602_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + ssm2602_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + pr_err("ssm2602: failed to create pcms\n"); + goto pcm_err; + } + /*power on device*/ + ssm2602_write(codec, SSM2602_ACTIVE, 0); + /* set the update bits */ + reg = ssm2602_read_reg_cache(codec, SSM2602_LINVOL); + ssm2602_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_RINVOL); + ssm2602_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_LOUT1V); + ssm2602_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V); + ssm2602_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); + /*select Line in as default input*/ + ssm2602_write(codec, SSM2602_APANA, + APANA_ENABLE_MIC_BOOST2 | APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); + ssm2602_write(codec, SSM2602_PWR, 0); + + ssm2602_add_controls(codec); + ssm2602_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + pr_err("ssm2602: 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 *ssm2602_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * ssm2602 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int ssm2602_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = ssm2602_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = ssm2602_init(socdev); + if (ret < 0) + pr_err("failed to initialise SSM2602\n"); + + return ret; +} + +static int ssm2602_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id ssm2602_i2c_id[] = { + { "ssm2602", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); +/* corgi i2c codec control layer */ +static struct i2c_driver ssm2602_i2c_driver = { + .driver = { + .name = "SSM2602 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ssm2602_i2c_probe, + .remove = ssm2602_i2c_remove, + .id_table = ssm2602_i2c_id, +}; + +static int ssm2602_add_i2c_device(struct platform_device *pdev, + const struct ssm2602_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&ssm2602_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ssm2602", I2C_NAME_SIZE); + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + return 0; +err_driver: + i2c_del_driver(&ssm2602_i2c_driver); + return -ENODEV; +} + +#endif + +static int ssm2602_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ssm2602_setup_data *setup; + struct snd_soc_codec *codec; + struct ssm2602_priv *ssm2602; + int ret = 0; + + pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL); + if (ssm2602 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ssm2602; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ssm2602_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + ret = ssm2602_add_i2c_device(pdev, setup); + } +#else + /* other interfaces */ +#endif + return ret; +} + +/* remove everything here */ +static int ssm2602_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&ssm2602_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ssm2602 = { + .probe = ssm2602_probe, + .remove = ssm2602_remove, + .suspend = ssm2602_suspend, + .resume = ssm2602_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); + +MODULE_DESCRIPTION("ASoC ssm2602 driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h new file mode 100644 index 0000000..f344e6d --- /dev/null +++ b/sound/soc/codecs/ssm2602.h @@ -0,0 +1,130 @@ +/* + * File: sound/soc/codecs/ssm2602.h + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Tue June 06 2008 + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef _SSM2602_H +#define _SSM2602_H + +/* SSM2602 Codec Register definitions */ + +#define SSM2602_LINVOL 0x00 +#define SSM2602_RINVOL 0x01 +#define SSM2602_LOUT1V 0x02 +#define SSM2602_ROUT1V 0x03 +#define SSM2602_APANA 0x04 +#define SSM2602_APDIGI 0x05 +#define SSM2602_PWR 0x06 +#define SSM2602_IFACE 0x07 +#define SSM2602_SRATE 0x08 +#define SSM2602_ACTIVE 0x09 +#define SSM2602_RESET 0x0f + +/*SSM2602 Codec Register Field definitions + *(Mask value to extract the corresponding Register field) + */ + +/*Left ADC Volume Control (SSM2602_REG_LEFT_ADC_VOL)*/ +#define LINVOL_LIN_VOL 0x01F /* Left Channel PGA Volume control */ +#define LINVOL_LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */ +#define LINVOL_LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */ + +/*Right ADC Volume Control (SSM2602_REG_RIGHT_ADC_VOL)*/ +#define RINVOL_RIN_VOL 0x01F /* Right Channel PGA Volume control */ +#define RINVOL_RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */ +#define RINVOL_RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */ + +/*Left DAC Volume Control (SSM2602_REG_LEFT_DAC_VOL)*/ +#define LOUT1V_LHP_VOL 0x07F /* Left Channel Headphone volume control */ +#define LOUT1V_ENABLE_LZC 0x080 /* Left Channel Zero cross detect enable */ +#define LOUT1V_LRHP_BOTH 0x100 /* Left Channel Headphone volume update */ + +/*Right DAC Volume Control (SSM2602_REG_RIGHT_DAC_VOL)*/ +#define ROUT1V_RHP_VOL 0x07F /* Right Channel Headphone volume control */ +#define ROUT1V_ENABLE_RZC 0x080 /* Right Channel Zero cross detect enable */ +#define ROUT1V_RLHP_BOTH 0x100 /* Right Channel Headphone volume update */ + +/*Analogue Audio Path Control (SSM2602_REG_ANALOGUE_PATH)*/ +#define APANA_ENABLE_MIC_BOOST 0x001 /* Primary Microphone Amplifier gain booster control */ +#define APANA_ENABLE_MIC_MUTE 0x002 /* Microphone Mute Control */ +#define APANA_ADC_IN_SELECT 0x004 /* Microphone/Line IN select to ADC (1=MIC, 0=Line In) */ +#define APANA_ENABLE_BYPASS 0x008 /* Line input bypass to line output */ +#define APANA_SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */ +#define APANA_ENABLE_SIDETONE 0x020 /* Enable/Disable Side Tone */ +#define APANA_SIDETONE_ATTN 0x0C0 /* Side Tone Attenuation */ +#define APANA_ENABLE_MIC_BOOST2 0x100 /* Secondary Microphone Amplifier gain booster control */ + +/*Digital Audio Path Control (SSM2602_REG_DIGITAL_PATH)*/ +#define APDIGI_ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */ +#define APDIGI_DE_EMPHASIS 0x006 /* De-Emphasis Control */ +#define APDIGI_ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */ +#define APDIGI_STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */ + +/*Power Down Control (SSM2602_REG_POWER) + *(1=Enable PowerDown, 0=Disable PowerDown) + */ +#define PWR_LINE_IN_PDN 0x001 /* Line Input Power Down */ +#define PWR_MIC_PDN 0x002 /* Microphone Input & Bias Power Down */ +#define PWR_ADC_PDN 0x004 /* ADC Power Down */ +#define PWR_DAC_PDN 0x008 /* DAC Power Down */ +#define PWR_OUT_PDN 0x010 /* Outputs Power Down */ +#define PWR_OSC_PDN 0x020 /* Oscillator Power Down */ +#define PWR_CLK_OUT_PDN 0x040 /* CLKOUT Power Down */ +#define PWR_POWER_OFF 0x080 /* POWEROFF Mode */ + +/*Digital Audio Interface Format (SSM2602_REG_DIGITAL_IFACE)*/ +#define IFACE_IFACE_FORMAT 0x003 /* Digital Audio input format control */ +#define IFACE_AUDIO_DATA_LEN 0x00C /* Audio Data word length control */ +#define IFACE_DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */ +#define IFACE_DAC_LR_SWAP 0x020 /* Swap DAC data control */ +#define IFACE_ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */ +#define IFACE_BCLK_INVERT 0x080 /* Bit Clock Inversion control */ + +/*Sampling Control (SSM2602_REG_SAMPLING_CTRL)*/ +#define SRATE_ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */ +#define SRATE_BOS_RATE 0x002 /* Base Over-Sampling rate */ +#define SRATE_SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */ +#define SRATE_CORECLK_DIV2 0x040 /* Core Clock divider select */ +#define SRATE_CLKOUT_DIV2 0x080 /* Clock Out divider select */ + +/*Active Control (SSM2602_REG_ACTIVE_CTRL)*/ +#define ACTIVE_ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */ + +/*********************************************************************/ + +#define SSM2602_CACHEREGNUM 10 + +#define SSM2602_SYSCLK 0 +#define SSM2602_DAI 0 + +struct ssm2602_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ssm2602_dai; +extern struct snd_soc_codec_device soc_codec_dev_ssm2602; + +#endif -- 1.5.6

2 1

[alsa-devel] [PATCH] ALSA: ASoC: Fix memory leaks for the CS4270 driver
by Timur Tabi 05 Sep '08

05 Sep '08

The CS4270 ASoC driver was not cleaning up resources completely during an I2C unbind. The same problem could occur if the bind failed. Rearranged a couple functions to eliminate an unnecessary function declaration. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is a fix for 2.6.27. It applies only to the ASoC V1 version of the driver. A similar fix for the V2 version will be posted later. sound/soc/codecs/cs4270.c | 94 ++++++++++++++++++++++++++++++-------------- 1 files changed, 64 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 82d94f0..eb38e92 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -490,29 +490,6 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) #endif -static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *); - -/* A list of non-DAPM controls that the CS4270 supports */ -static const struct snd_kcontrol_new cs4270_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", - CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) -}; - -static const struct i2c_device_id cs4270_id[] = { - {"cs4270", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs4270_id); - -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "CS4270 I2C", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, -}; - /* * Global variable to store socdev for i2c probe function. * @@ -530,6 +507,12 @@ static struct i2c_driver cs4270_i2c_driver = { */ static struct snd_soc_device *cs4270_socdev; +/* A list of non-DAPM controls that the CS4270 supports */ +static const struct snd_kcontrol_new cs4270_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) +}; + /* * Initialize the I2C interface of the CS4270 * @@ -559,8 +542,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); if (!codec->reg_cache) { printk(KERN_ERR "cs4270: could not allocate register cache\n"); - ret = -ENOMEM; - goto error; + return -ENOMEM; } /* Verify that we have a CS4270 */ @@ -610,20 +592,72 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, return 0; error: - if (codec->control_data) { - i2c_detach_client(i2c_client); - codec->control_data = NULL; + for (i = 1; i <= ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl = + snd_ctl_find_numid(codec->card, i); + + if (kctrl) + snd_ctl_remove(codec->card, kctrl); } + codec->control_data = NULL; + kfree(codec->reg_cache); codec->reg_cache = NULL; codec->reg_cache_size = 0; - kfree(i2c_client); - return ret; } +/* + * Remove I2C interface of the CS4270 + * + * Since this driver cannot be compiled as a module, the only way this + * function can be called is if the I2C subsystem unbinds it. This can be + * triggered, for instance, by writing the device ID to this drivers + * 'unbind' file in /sys. + */ +static int cs4270_i2c_remove(struct i2c_client *i2c_client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(i2c_client); + + if (codec) { + unsigned int i; + + for (i = 1; i <= ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl = + snd_ctl_find_numid(codec->card, i); + + if (kctrl) + snd_ctl_remove(codec->card, kctrl); + } + + codec->control_data = NULL; + + kfree(codec->reg_cache); + codec->reg_cache = NULL; + codec->reg_cache_size = 0; + } + + return 0; +} + +static const struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, +}; + #endif /* USE_I2C*/ struct snd_soc_dai cs4270_dai = { -- 1.5.5

2 9

[alsa-devel] [PATCH 0/9] ASoC Blackfin supporting (v2)
by Bryan Wu 05 Sep '08

05 Sep '08

Hi Mark, With Cliff's effort, we update this patch series quickly according to your review. v1-v2: - move ASoC Blackfin Kconfig and Makefile patch to be the last one in the series - fix coding style issues - The SND_SOC_DAFIMT_LEFT_J: ought to be default - fix other minor issues v0-v1: - fix coding style issues - use latest ASoC API - split the whole patch into this 9 patches in a patchset Thanks a lot -Bryan

2 14

[alsa-devel] Problems recording with HP DV2839TX laptop.
by Travis Place 05 Sep '08

05 Sep '08

Hello everyone, Im Travis (wishie) once of the many helpers who man the unofficial #alsa on freenode irc network.. I usually spend my time fixing other peoples sound issues, but now i have my own. I have a HP DV2839TX laptop, and i get ALOT of static when recording via the inbuilt mic. Also, the mic controls are very strange.. I have (in the Capture section of alsamixer).. Digital, Docking Mic, External Mic, Internal Mic.. only the "Digital" control seems to effect my recording volume at all. Strangely, though, the '* Mic' controls in the PLAYBACK section of alsamixer do change the recording properties somewhat (volume being the main factor). In any event, i get a rather large amount of static, no matter what mixer settings i try. This isnt a problem in the 'other' OS. My alsa-info.sh output for alsa-driver 1.0.16 is at : http://www.alsa-project.org/db/?f=0d1df6446ff1c71ad305e7b7b367f0a6b22b80c8 My alsa-info.sh output for alsa-driver 1.0.18rc1 is at : http://www.alsa-project.org/db/?f=9e5cd99f4822e0d671866fa630698bd1b0cd954b

2 2

[alsa-devel] WM8750 mixer controls
by Harald Radke 05 Sep '08

05 Sep '08

Hi there! I noticed that using "alsamixer" several controls that are connected to sound input (ALC) are listed in the "playback" tab as well as in "capture", as well as certain playback controls as "bass" "trebel" also can be found under "capture". However, changing those controls' values only seem to take effect in the "playback" tab. I was wondering if this is intentional or unintentional done by the WM8750 codec driver or an alsamixer "feature". Thx Greez, Harry

2 1

Re: [alsa-devel] [PATCH] add the nvidia HDMI codec driver for MCP77/79
by Takashi Iwai 05 Sep '08

05 Sep '08

At Wed, 3 Sep 2008 15:58:17 +0800, peerchen wrote: > > Add the nvidia HDMI codec driver for MCP77/79. > The patch based on kernel 2.6.27-rc5. > > Signed-off-by: Wei Ni <wni(a)nvidia.com> > Signed-off-by: Peer Chen <peerchen(a)gmail.com> Thanks for the patch. Some comments below. > diff -uprN -X linux-2.6.26-Orig/Documentation/dontdiff linux-2.6.26-Orig/sound/pci/hda/hda_intel.c linux-2.6.26-New/sound/pci/hda/hda_intel.c > --- linux-2.6.26-Orig/sound/pci/hda/hda_intel.c 2008-08-28 13:47:20.000000000 +0800 > +++ linux-2.6.26-New/sound/pci/hda/hda_intel.c 2008-08-28 15:33:51.000000000 +0800 > @@ -223,8 +223,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO > #define RIRB_INT_MASK 0x05 > > /* STATESTS int mask: SD2,SD1,SD0 */ > -#define AZX_MAX_CODECS 3 > -#define STATESTS_INT_MASK 0x07 > +#define AZX_MAX_CODECS 4 > +#define STATESTS_INT_MASK 0x0f This may break other platforms. Expanding STATESTS_INT_MASK is OK and would be harmless, but AZX_MAX_CODECS is not. AZX_MAX_CODECS is the default number of max codecs. The chipset specific max number of codecs is defined in azx_max_codecs[]. If Nvidia chipsets supports properly more than 3, change azx_max_codecs[AZX_DRIVER_NVIDIA] to 4. > +static unsigned short convert_from_spdif_status_nv(unsigned int sbits) > +{ > + unsigned short val = 0; > + > + if (sbits & IEC958_AES0_PROFESSIONAL) > + val |= AC_DIG1_PROFESSIONAL; > + if (sbits & IEC958_AES0_NONAUDIO) > + val |= AC_DIG1_NONAUDIO; > + if (sbits & IEC958_AES0_PROFESSIONAL) { > + if ((sbits & IEC958_AES0_PRO_EMPHASIS) == > + IEC958_AES0_PRO_EMPHASIS_5015) > + val |= AC_DIG1_EMPHASIS; > + } else { > + if ((sbits & IEC958_AES0_CON_EMPHASIS) == > + IEC958_AES0_CON_EMPHASIS_5015) > + val |= AC_DIG1_EMPHASIS; > + if (!(sbits & IEC958_AES0_CON_NOT_COPYRIGHT)) > + val |= AC_DIG1_COPYRIGHT; > + if (sbits & (IEC958_AES1_CON_ORIGINAL << 8)) > + val |= AC_DIG1_LEVEL; > + val |= sbits & (IEC958_AES1_CON_CATEGORY << 8); > + } > + return val; > +} > + > +static unsigned int convert_to_spdif_status_nv(unsigned short val) > +{ > + unsigned int sbits = 0; > + > + if (val & AC_DIG1_NONAUDIO) > + sbits |= IEC958_AES0_NONAUDIO; > + if (val & AC_DIG1_PROFESSIONAL) > + sbits |= IEC958_AES0_PROFESSIONAL; > + if (sbits & IEC958_AES0_PROFESSIONAL) { > + if (sbits & AC_DIG1_EMPHASIS) > + sbits |= IEC958_AES0_PRO_EMPHASIS_5015; > + } else { > + if (val & AC_DIG1_EMPHASIS) > + sbits |= IEC958_AES0_CON_EMPHASIS_5015; > + if (!(val & AC_DIG1_COPYRIGHT)) > + sbits |= IEC958_AES0_CON_NOT_COPYRIGHT; > + if (val & AC_DIG1_LEVEL) > + sbits |= (IEC958_AES1_CON_ORIGINAL << 8); > + sbits |= val & (0x7f << 8); > + } > + return sbits; > +} > + > +static int snd_hda_hdmi_cmask_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | > + IEC958_AES0_NONAUDIO | > + IEC958_AES0_CON_EMPHASIS_5015 | > + IEC958_AES0_CON_NOT_COPYRIGHT; > + ucontrol->value.iec958.status[1] = IEC958_AES1_CON_CATEGORY | > + IEC958_AES1_CON_ORIGINAL; > + return 0; > +} > + > +static int snd_hda_hdmi_pmask_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | > + IEC958_AES0_NONAUDIO | > + IEC958_AES0_PRO_EMPHASIS_5015; > + return 0; > +} > + > +static int snd_hda_hdmi_mask_info(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_info *uinfo) > +{ > + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; > + uinfo->count = 1; > + return 0; > +} > + > +static int snd_hda_hdmi_default_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + > + ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff; > + ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff; > + ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff; > + ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff; > + > + return 0; > +} > + > +static int snd_hda_hdmi_default_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + hda_nid_t nid = kcontrol->private_value; > + unsigned short val; > + int change; > + > + mutex_lock(&codec->spdif_mutex); > + codec->spdif_status = ucontrol->value.iec958.status[0] | > + ((unsigned int)ucontrol->value.iec958.status[1] << 8) | > + ((unsigned int)ucontrol->value.iec958.status[2] << 16) | > + ((unsigned int)ucontrol->value.iec958.status[3] << 24); > + val = convert_from_spdif_status_nv(codec->spdif_status); > + val |= codec->spdif_ctls & 1; > + change = codec->spdif_ctls != val; > + codec->spdif_ctls = val; > + > + if (change) { > + snd_hda_codec_write_cache(codec, nid, 0, > + AC_VERB_SET_DIGI_CONVERT_1, > + val & 0xff); > + snd_hda_codec_write_cache(codec, nid, 0, > + AC_VERB_SET_DIGI_CONVERT_2, > + val >> 8); > + } > + > + mutex_unlock(&codec->spdif_mutex); > + return change; > +} > + > +#define snd_hda_hdmi_out_switch_info snd_ctl_boolean_mono_info > + > +static int snd_hda_hdmi_out_switch_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + > + ucontrol->value.integer.value[0] = codec->spdif_ctls & AC_DIG1_ENABLE; > + return 0; > +} > + > +static int snd_hda_hdmi_out_switch_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + hda_nid_t nid = kcontrol->private_value; > + unsigned short val; > + int change; > + > + mutex_lock(&codec->spdif_mutex); > + val = codec->spdif_ctls & ~AC_DIG1_ENABLE; > + if (ucontrol->value.integer.value[0]) > + val |= AC_DIG1_ENABLE; > + change = codec->spdif_ctls != val; > + if (change) { > + codec->spdif_ctls = val; > + snd_hda_codec_write_cache(codec, nid, 0, > + AC_VERB_SET_DIGI_CONVERT_1, > + val & 0xff); > + /* unmute amp switch (if any) */ > + if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && > + (val & AC_DIG1_ENABLE)) > + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, > + HDA_AMP_MUTE, 0); > + } > + mutex_unlock(&codec->spdif_mutex); > + return change; > +} These are almost identical with the functions in hda_codec.c. Is your intention to allow a different set of IEC958 status bits controls for HDMI *in addition* to SPDIF? If so, we should change the codes in hda_codec.c to allow other control names. > +struct snd_kcontrol_new hdmi_mixes[] = { > + { > + .access = SNDRV_CTL_ELEM_ACCESS_READ, > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "IEC958 Playback Con Mask HDMI", > + .info = snd_hda_hdmi_mask_info, > + .get = snd_hda_hdmi_cmask_get, > + }, These have also different names from the standard ones. Any reason? > +static struct hda_pcm_stream nvhdmi_pcm_digital_playback = { > + .substreams = 1, > + .channels_min = 2, > + .channels_max = 2, > + .nid = 0x4, /* NID to query formats and rates and setup streams */ > + .rates = SNDRV_PCM_RATE_48000, > + .maxbps = 16, > + .formats = SNDRV_PCM_FMTBIT_S16_LE, Use tabs please. > +static int nvhdmi_build_pcms(struct hda_codec *codec) > +{ > + struct nvhdmi_spec *spec = codec->spec; > + struct hda_pcm *info = &spec->pcm_rec; > + > + codec->num_pcms = 1; > + codec->pcm_info = info; > + > + info->name = "NVIDIA HDMI"; > + info->pcm_type = HDA_PCM_TYPE_HDMI Ditto. thanks, Takashi

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Re: [alsa-devel] noise and consume pcm faster
by Nie Jun 05 Sep '08

05 Sep '08

It is strange that there is no noise in the first several seconds. It seems like a buffer overflow with too much data accumulated. However, there should not be overflow in playback, right? 2008/9/5 Nie Jun <niej0001(a)gmail.com> > I do not know how these two params will be modified by ALSA lib. It does > not directly accept specified value. > With this cmd: > ./aplay 48k_stero.wav -F5120 -B106400 > I get the nearest setting as following, but still has noise. > > channels: 2 > rate: 48000 (48000/1) > period_size: 248 > buffer_size: 5208 > > > > 2008/9/5 Takashi Iwai <tiwai(a)suse.de> > > At Fri, 5 Sep 2008 14:34:51 +0800, >> Nie Jun wrote: >> > >> > I can not figure out the reason of noise and faster consumption from >> these >> > different. Do you have any idea about these different? Or any other data >> may >> > contain clue? >> >> So, do you mean that the problem persists even if you use the exactly >> same period_size and buffer_size as OSS (i.e. 256 and 5120)? >> >> >> Takashi >> > >

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[alsa-devel] Making envy24control/kenvy24 work on Envy24 HT/HT-S cards
by Vedran Miletić 05 Sep '08

05 Sep '08

Hi, any ideas if it would be possible to make envy24control or kenvy24 work on HT/HT-S cards? I'm referring to VU-meters / mixer / settings / profiles part, not the routing since those chipsets don't support it, according to VIA. -- Vedran Miletić

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