November 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Conexant CX20561 - Microphone problems
by Shawn Starr 19 Apr '11

19 Apr '11

Hello ALSA developers, I am having problems with the microphone on the following hardware this is on a Lenovo W500 laptop. Card: HDA Intel Chip: Conexant CX20561 (Hermosa) Here is the problems im noticing: 1) If I use ALSA 'default' with the microphone I get noise interference on output [high/low/high/low]. If I switch this to use the "ALSA Capture on HDA Intel [CONEXANT Analog] I get interference except it is the standard 60Hz electrical noise [steady noise] 2) Using the internal/external microphones gives an inconsistent volume. It also does not seem to work with the microphone cutting out so much. I need to adjust Digital + another microphone but it still drops capturing. Alsamixer shows: Digital/Docking Microphone/External Microphone/Internal Microphone Here is the module options I give to the sound driver: options snd-intel-hda enable_msi=1 power_save=10 power_save_controller=true model=laptop index=0 Assuming power saving/controller causes the microphone to cut out or not. 3) I'm now aware that this card has no pinouts for PCM capture https://bugzilla.redhat.com/show_bug.cgi?id=502293 Thanks for any help, Shawn.

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[alsa-devel] CA0110 taints kernel
by Vedran Miletić 13 Apr '11

13 Apr '11

Hi, I experience frequent kernel crashes with 2.6.31 and CA0110. I know this card doesn't work, and there is no problem with that; however, it crashing non-stop is not very nice. Also, card doesn't get any controls (and I believe it does get some controls when it's the only card using HDA driver, that is when I disable onboard one). dmesg is here: http://pastebin.com/m7c53562f lspci: 02:00.0 PCI bridge: Creative Labs Device 7006 (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=02, secondary=03, subordinate=03, sec-latency=64 Memory behind bridge: fe900000-fe9fffff Capabilities: [50] Power Management version 3 Capabilities: [60] MSI: Enable- Count=1/16 Maskable- 64bit+ Capabilities: [80] Subsystem: Creative Labs Device 0010 Capabilities: [90] Express PCI/PCI-X Bridge, MSI 00 Capabilities: [100] Advanced Error Reporting Kernel modules: shpchp 03:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Flags: bus master, medium devsel, latency 64, IRQ 16 Memory at fe9fc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel Any ideas? -- Vedran Miletić P.S. Please CC me, as I'm not subscribed to mailing list at the moment.

2 2

[alsa-devel] [PATCH] ASoC: add SSM2604 codec driver
by Mike Frysinger 27 Mar '11

27 Mar '11

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2604.c | 643 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ssm2604.h | 92 +++++++ 4 files changed, 741 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ssm2604.c create mode 100644 sound/soc/codecs/ssm2604.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5da30eb..5db82d2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C + select SND_SOC_SSM2604 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -150,6 +151,9 @@ config SND_SOC_SPDIF config SND_SOC_SSM2602 tristate +config SND_SOC_SSM2604 + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 91429ea..7593eeb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -16,6 +16,7 @@ snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-ssm2604-objs := ssm2604.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -81,6 +82,7 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_SSM2604) += snd-soc-ssm2604.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o diff --git a/sound/soc/codecs/ssm2604.c b/sound/soc/codecs/ssm2604.c new file mode 100644 index 0000000..7522586 --- /dev/null +++ b/sound/soc/codecs/ssm2604.c @@ -0,0 +1,643 @@ +/* + * Driver for ssm2604 sound codec + * + * Copyright 2010 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#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/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ssm2604.h" + +#define SSM2604_VERSION "0.1" + +struct snd_soc_codec_device soc_codec_dev_ssm2604; +static struct snd_soc_codec *ssm2604_codec; +/* codec private data */ +struct ssm2604_priv { + unsigned int sysclk; + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; + struct snd_soc_codec codec; +}; + +/* + * ssm2604 register cache + * We can't read the ssm2604 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 ssm2604_reg[SSM2604_CACHEREGNUM] = { + 0x0017, 0x0017, 0x0000, 0x0000, + 0x0000, 0x000a, 0x0000, 0x0000 +}; + +#define ssm2604_reset(c) snd_soc_write(c, SSM2604_RESET, 0) + +static const char *ssm2604_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum ssm2604_enum[] = { + SOC_ENUM_SINGLE(SSM2604_APDIGI, 1, 4, ssm2604_deemph), +}; + +static const struct snd_kcontrol_new ssm2604_snd_controls[] = { + +SOC_DOUBLE_R("Capture Volume", SSM2604_LINVOL, SSM2604_RINVOL, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", SSM2604_LINVOL, SSM2604_RINVOL, 7, 1, 1), + +SOC_SINGLE("ADC High Pass Filter Switch", SSM2604_APDIGI, 0, 1, 1), +SOC_SINGLE("Store DC Offset Switch", SSM2604_APDIGI, 4, 1, 0), + +SOC_ENUM("Capture Source", ssm2604_enum[0]), + +SOC_ENUM("Playback De-emphasis", ssm2604_enum[1]), +}; + +/* Output Mixer */ +static const struct snd_kcontrol_new ssm2604_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", SSM2604_APANA, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget ssm2604_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", SSM2604_PWR, 4, 1, + &ssm2604_output_mixer_controls[0], + ARRAY_SIZE(ssm2604_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2604_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2604_PWR, 2, 1), +SND_SOC_DAPM_PGA("Line Input", SSM2604_PWR, 0, 1, NULL, 0), +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"}, + + /* outputs */ + {"ROUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + + /* input mux */ + {"Input Mux", "Line", "Line Input"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, +}; + +static int ssm2604_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ssm2604_dapm_widgets, + ARRAY_SIZE(ssm2604_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); + + 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 ssm2604_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u16 srate; + 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 ssm2604_priv *ssm2604 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = codec->control_data; + u16 iface = snd_soc_read(codec, SSM2604_IFACE) & 0xfff3; + int i = get_coeff(ssm2604->sysclk, params_rate(params)); + + if (substream == ssm2604->slave_substream) { + dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + return 0; + } + + /*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; + + snd_soc_write(codec, SSM2604_ACTIVE, 0); + snd_soc_write(codec, SSM2604_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; + } + snd_soc_write(codec, SSM2604_IFACE, iface); + snd_soc_write(codec, SSM2604_ACTIVE, ACTIVE_ACTIVATE_CODEC); + return 0; +} + +static int ssm2604_startup(struct snd_pcm_substream *substream, + 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 ssm2604_priv *ssm2604 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = codec->control_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. + * TODO: the ssm2604 allows pairs of non-matching PB/REC rates + */ + if (ssm2604->master_substream) { + master_runtime = ssm2604->master_substream->runtime; + dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + master_runtime->sample_bits, + master_runtime->rate); + + if (master_runtime->rate != 0) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + if (master_runtime->sample_bits != 0) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + ssm2604->slave_substream = substream; + } else + ssm2604->master_substream = substream; + + return 0; +} + +static int ssm2604_pcm_prepare(struct snd_pcm_substream *substream, + 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; + /* set active */ + snd_soc_write(codec, SSM2604_ACTIVE, ACTIVE_ACTIVATE_CODEC); + + return 0; +} + +static void ssm2604_shutdown(struct snd_pcm_substream *substream, + 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 ssm2604_priv *ssm2604 = snd_soc_codec_get_drvdata(codec); + + /* deactivate */ + if (!codec->active) + snd_soc_write(codec, SSM2604_ACTIVE, 0); + + if (ssm2604->master_substream == substream) + ssm2604->master_substream = ssm2604->slave_substream; + + ssm2604->slave_substream = NULL; +} + +static int ssm2604_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, SSM2604_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; + if (mute) + snd_soc_write(codec, SSM2604_APDIGI, + mute_reg | APDIGI_ENABLE_DAC_MUTE); + else + snd_soc_write(codec, SSM2604_APDIGI, mute_reg); + return 0; +} + +static int ssm2604_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 ssm2604_priv *ssm2604 = snd_soc_codec_get_drvdata(codec); + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + ssm2604->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int ssm2604_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 |= 0x0013; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0003; + 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 */ + snd_soc_write(codec, SSM2604_IFACE, iface); + return 0; +} + +static int ssm2604_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = snd_soc_read(codec, SSM2604_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + snd_soc_write(codec, SSM2604_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_write(codec, SSM2604_PWR, reg | PWR_CLK_OUT_PDN); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_write(codec, SSM2604_ACTIVE, 0); + snd_soc_write(codec, SSM2604_PWR, 0xffff); + break; + + } + codec->bias_level = level; + return 0; +} + +#define SSM2604_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define SSM2604_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 ssm2604_dai_ops = { + .startup = ssm2604_startup, + .prepare = ssm2604_pcm_prepare, + .hw_params = ssm2604_hw_params, + .shutdown = ssm2604_shutdown, + .digital_mute = ssm2604_mute, + .set_sysclk = ssm2604_set_dai_sysclk, + .set_fmt = ssm2604_set_dai_fmt, +}; + +struct snd_soc_dai ssm2604_dai = { + .name = "SSM2604", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2604_RATES, + .formats = SSM2604_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2604_RATES, + .formats = SSM2604_FORMATS,}, + .ops = &ssm2604_dai_ops, +}; +EXPORT_SYMBOL_GPL(ssm2604_dai); + +static int ssm2604_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; + + ssm2604_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2604_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(ssm2604_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + ssm2604_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ssm2604_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +static int ssm2604_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int reg, ret = 0; + + socdev->card->codec = ssm2604_codec; + codec = ssm2604_codec; + + ssm2604_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "ssm2604: failed to create pcms\n"); + goto pcm_err; + } + /*power on device*/ + snd_soc_write(codec, SSM2604_ACTIVE, 0); + /* set the update bits */ + reg = snd_soc_read(codec, SSM2604_LINVOL); + snd_soc_write(codec, SSM2604_LINVOL, reg | LINVOL_LRIN_BOTH); + reg = snd_soc_read(codec, SSM2604_RINVOL); + snd_soc_write(codec, SSM2604_RINVOL, reg | RINVOL_RLIN_BOTH); + + snd_soc_write(codec, SSM2604_APANA, APANA_SELECT_DAC); + snd_soc_write(codec, SSM2604_PWR, 0); + + snd_soc_add_controls(codec, ssm2604_snd_controls, + ARRAY_SIZE(ssm2604_snd_controls)); + ssm2604_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +/* remove everything here */ +static int ssm2604_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_ssm2604 = { + .probe = ssm2604_probe, + .remove = ssm2604_remove, + .suspend = ssm2604_suspend, + .resume = ssm2604_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2604); + +static int ssm2604_register(struct ssm2604_priv *ssm2604, enum snd_soc_control_type control) +{ + struct snd_soc_codec *codec = &ssm2604->codec; + int ret = 0; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "SSM2604"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ssm2604_set_bias_level; + codec->dai = &ssm2604_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(ssm2604_reg); + codec->reg_cache = kmemdup(ssm2604_reg, sizeof(ssm2604_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&ssm2604_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return ret; +} + +static void ssm2604_unregister(struct ssm2604_priv *ssm2604) +{ + struct snd_soc_codec *codec = &ssm2604->codec; + + ssm2604_set_bias_level(&ssm2604->codec, SND_SOC_BIAS_OFF); + kfree(codec->reg_cache); + snd_soc_unregister_dai(&ssm2604_dai); + snd_soc_unregister_codec(&ssm2604->codec); + kfree(ssm2604); + ssm2604_codec = NULL; +} + +static int ssm2604_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2604_priv *ssm2604; + struct snd_soc_codec *codec; + + ssm2604 = kzalloc(sizeof(struct ssm2604_priv), GFP_KERNEL); + if (ssm2604 == NULL) + return -ENOMEM; + codec = &ssm2604->codec; + snd_soc_codec_set_drvdata(codec, ssm2604); + + i2c_set_clientdata(i2c, ssm2604); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + ssm2604_codec = codec; + + return ssm2604_register(ssm2604, SND_SOC_I2C); +} + +static __devexit int ssm2604_i2c_remove(struct i2c_client *client) +{ + struct ssm2604_priv *ssm2604 = i2c_get_clientdata(client); + ssm2604_unregister(ssm2604); + return 0; +} + +static const struct i2c_device_id ssm2604_i2c_id[] = { + { "ssm2604", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2604_i2c_id); + +/* corgi i2c codec control layer */ +static struct i2c_driver ssm2604_i2c_driver = { + .driver = { + .name = "ssm2604", + .owner = THIS_MODULE, + }, + .probe = ssm2604_i2c_probe, + .remove = __devexit_p(ssm2604_i2c_remove), + .id_table = ssm2604_i2c_id, +}; + +static int __init ssm2604_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&ssm2604_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register ssm2604 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(ssm2604_modinit); + +static void __exit ssm2604_exit(void) +{ + i2c_del_driver(&ssm2604_i2c_driver); +} +module_exit(ssm2604_exit); + +MODULE_DESCRIPTION("ASoC SSM2604 driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2604.h b/sound/soc/codecs/ssm2604.h new file mode 100644 index 0000000..3b24a08 --- /dev/null +++ b/sound/soc/codecs/ssm2604.h @@ -0,0 +1,92 @@ +/* + * Header for ssm2604 sound codec + * + * Copyright 2010 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _SSM2604_H +#define _SSM2604_H + +/* SSM2604 Codec Register definitions */ + +#define SSM2604_LINVOL 0x00 +#define SSM2604_RINVOL 0x01 +#define SSM2604_APANA 0x04 +#define SSM2604_APDIGI 0x05 +#define SSM2604_PWR 0x06 +#define SSM2604_IFACE 0x07 +#define SSM2604_SRATE 0x08 +#define SSM2604_ACTIVE 0x09 +#define SSM2604_RESET 0x0f + +/* SSM2604 Codec Register Field definitions + * (Mask value to extract the corresponding Register field) + */ + +/* Left ADC Volume Control (SSM2604_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 (SSM2604_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 */ + + +/* Analogue Audio Path Control (SSM2604_REG_ANALOGUE_PATH) */ +#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) */ + +/* Digital Audio Path Control (SSM2604_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 (SSM2604_REG_POWER) + * (1=Enable PowerDown, 0=Disable PowerDown) + */ +#define PWR_LINE_IN_PDN 0x001 /* Line Input Power Down */ +#define PWR_ADC_PDN 0x004 /* ADC Power Down */ +#define PWR_DAC_PDN 0x008 /* DAC 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 (SSM2604_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 (SSM2604_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 (SSM2604_REG_ACTIVE_CTRL) */ +#define ACTIVE_ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */ + +/*********************************************************************/ + +#define SSM2604_CACHEREGNUM 9 + +#define SSM2604_SYSCLK 0 +#define SSM2604_DAI 0 + +struct ssm2604_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ssm2604_dai; +extern struct snd_soc_codec_device soc_codec_dev_ssm2604; + +#endif -- 1.7.2

3 6

Re: [alsa-devel] Microphone not working on PCI ID 8086:3b56
by Jeff Chua 18 Mar '11

18 Mar '11

On Tue, Sep 28, 2010 at 9:42 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > Enable all CONFIG_SND_HDA_*. Done that. Still no mic. On Tue, Sep 28, 2010 at 9:42 PM, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > It's easier to enable all of them. They are small modules. Tried just about everything, still no mic. Sounds works in all cases. I found that with SND_HDA_CODEC_INTELHDMI selected, SND_DYNAMIC_MINORS will be enabled by default which won't work in my case as I'm not using udev. Here's a patch to take away the default so non-udev will still works. Do I really need udev to get mic to work? Thanks, Jeff --- lx/sound/pci/hda/Kconfig.org 2010-09-14 16:34:52.000000000 +0800 +++ lx/sound/pci/hda/Kconfig 2010-09-14 16:35:09.000000000 +0800 @@ -145,7 +145,6 @@ config SND_HDA_CODEC_INTELHDMI bool "Build INTEL HDMI HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include INTEL HDMI HD-audio codec support in

3 16

[alsa-devel] some questions about sound system
by loody 17 Mar '11

17 Mar '11

Dear all: I just install Ubuntu9.10 on my laptop and I try to update my sound system and recompile the utility. there are some questions I met: 1. when configure the utility it says checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found and config.log is "conftest.c:10:42: error: CoreFoundation/CFPreferences.h: No such file or directory conftest.c: In function 'main': conftest.c:14: error: 'NULL' undeclared (first use in this function) conftest.c:14: error: (Each undeclared identifier is repor" the above error seems I miss install some package before. I google the name but I have no improvement. "conftest.c:20:28: error: ac_nonexistent.h: No such file or directory configure:7397: $? = 1" the above error seems the header located where the conftest.c doesn't expect. how could solve them all? 2. there are some configure file in share when I compile advance linux sound architecture. like alsa.conf. have anyone know where I can find the naming rule? appreciate your help, miloody

4 7

[alsa-devel] idt STAC927x adat supported (sigmatel HDA codec) ?
by Eric 03 Mar '11

03 Mar '11

Is the ADAT output of the IDT (sigmatel) STAC927x HDA codec supported in alsa ? If not could someone please try to add this incredibly nice feature in the existing alsa driver ? Thank you very much

4 13

[alsa-devel] Channel swapping problem on Atmel SSC audio
by Ryan Mallon 18 Feb '11

18 Feb '11

Hi all, I'm working on a custom board using the AT91SAM9G45 processor and a TLV320AIC26 SPI attached audio codec using I2S format audio. When doing audio playback on the board the left and right channels are occasionally flipped. I have verified this by generating a custom wav file which has constant data on one channel and an incrementing counter on the other channel. The digitial out from the CPU shows that no data is being lost, but around 1/5 times the audio channels will be flipped. The problem appears to be that the DMA is starting on the wrong edge of the LRCLK. I have using both SND_SOC_DAIFMT_CBM_CFM (Codec provides clocks) and SND_SOC_DAIFMT_CBS_CFS (CPU provides clocks) but the problem occurs in both situations. I have also tried changing TCMR:CKO to SSC_CKO_TRANSFER and TCMR:START to SSC_START_LOW_RF, but the problem still occurs. I have looked through the PCM/SSC DMA code and cannot find any obvious bugs. I have checked the errata for the At91SAM9G45 processor (our chip marking is CU-1004-A), but can't find anything which relates directly to the problem we are seeing. Has anybody else had this problem, or does anybody have any AT91/Atmel hardware that they could test to see if it occurs on other hardware also? Thanks, ~Ryan -- Bluewater Systems Ltd - ARM Technology Solution Centre Ryan Mallon 5 Amuri Park, 404 Barbadoes St ryan(a)bluewatersys.com PO Box 13 889, Christchurch 8013 http://www.bluewatersys.com New Zealand Phone: +64 3 3779127 Freecall: Australia 1800 148 751 Fax: +64 3 3779135 USA 1800 261 2934

6 6

[alsa-devel] [PATCH - snd-usb-audio 1/1] Support for Power/Status LED on Creative USB X-Fi S51
by emailmandar＠gmail.com 13 Feb '11

13 Feb '11

From: Mandar Joshi <emailmandar(a)gmail.com> This patch adds support for Power/Status LED on Creative USB X-Fi S51. There is just one LED on the device. The LED can either be On or it can be set to Blink. There doesn't seem to be a way to switch it off. The control message to change LED status is similar to that of audigy2nx except that the index is to be set to 0 and value is 1 for Blink and 0 for On. The 'Power LED' control in alsamixer when muted will cause the LED to Blink continuously. When unmuted the LED will stay On. The Creative driver under Windows sets the LED to blink whenever audio is muted. This LED can be treated as the CMSS LED but I figured since there is just one LED, it should be treated as the Power LED. Is that alright? I've also changed the comment "Usb X-Fi" to "Usb X-Fi S51" as there are other external X-Fi devices from Creative like Usb X-Fi Go and Xmod. The volume knob and LED support patch doesn't apply to them. Signed-off-by: Mandar Joshi <emailmandar(a)gmail.com> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e7df1e5..aea92d6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -60,6 +60,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; @@ -182,10 +183,16 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (value > 1) return -EINVAL; changed = value != mixer->audigy2nx_leds[index]; - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - value, index + 2, NULL, 0, 100); + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + !value, 0, NULL, 0, 100); + else + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + value, index + 2, NULL, 0, 100); if (err < 0) return err; mixer->audigy2nx_leds[index] = value; @@ -224,8 +231,12 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { + /* USB X-Fi S51 doesn't have a CMSS LED */ + if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0) + continue; if (i > 1 && /* Live24ext has 2 LEDs only */ (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, @@ -364,6 +375,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { if ((err = snd_audigy2nx_controls_create(mixer)) < 0) return err; -- 1.7.1

4 5

[alsa-devel] Problem with VIA VT1708S and git versions of alsa-driver.
by Mark Goldstein 31 Jan '11

31 Jan '11

Hello, Recently I'm having some strange issues with my VIA VT1708S on-board sound card. The description might be a bit long, so I'll give a brief summary first. I have ASUS P5QL/EPU motherboard with VT1708S 8-channel HDA Codec. There are 3 OSs installed on this machine: openSUSE 11.1 (my main system), openSUSE 11.3, Kubuntu 10.04. Audio worked fine until some updates at the beginning of September. Since then I have 2 different issues. 1).On 11.1 - Line In control seems inactive, instead Front Mic control controls the level from Line In jack. Front Mic is constanly disabled. alsaconf does not show hda-intel in the list of detected cards. Latest version it happened with is git20100925. When I compiled stable 1.0.23 sources, everything started working as expected again. I reported this issue on bugtrack: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133 2) On oS 11.3 I start experiencing problems recording audio either from Mic or from Line. After some experiments I found out that the Capture1 control is somehow broken. Mute switch is reversed. When it is unmuted, there is no audio recorded. When it is muted, recording works. The same happened to at least one more user of openSUSE, see the thread "How to investigate the problem with MIC" on alsa-user mailing list. I've also added my comments to issue https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130 but now I think it is different issue. Even when I have audio recorded normally when Capture1 is muted, I still have very distorted voice from Mic in Skype. It worked fine before the problem started. More details: On oS 11.1 I had my audio working fine after updating to 1.0.23 versions of alsa driver. I'm using opensuse/multimedia: repository that provides binary rpm of alsa-driver git snapshots. I have only one current kernel 2.6.27.48 on 11.1. On oS 11.3 audio worked right out of the box, since it uses kernels with alsa-driver 1.0.23. I have 2-3 kernels installed at the same time - normal 2.6.34 from official repository and 2.6.35 from kernel build service. I'm not sure when exactly the problems started, but definitely at the beginning of September. I've initially saw that on oS 11.1 I can't hear the audio from my TV card connected to Line In. Then I noticed that Mic does not work in Skype on both 11.1 and 11.3, while it still works on Kubuntu. When I unmuted Front Mic on 11.1 I've suddenly found that it actually enabled Line In. I think now that something went bad with card detecting. Probably on 11.1 with latest git versions my card is not properly detected and some "defaults" are used. I'm attaching here 2 outputs of alsa-info.sh - 11.1_stable captured with alsa-driver compiled from stable 1.0.23 sources. 11.1_git25 captured with driver compiled with git20100925 version from opensuse/multimedia: repository. For me the problem with 11.1 is resolved, but probably it's worth checking why latest version from git behave this way. I'll continue with 11.3 investigation later on. Regards, -- Mark Goldstein

2 21

Re: [alsa-devel] ALSA Multichannel through HDMI
by John Ettedgui 23 Jan '11

23 Jan '11

On Thu, Nov 25, 2010 at 6:21 PM, John Ettedgui <john.ettedgui(a)gmail.com> wrote: > On Thu, Nov 25, 2010 at 6:16 PM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >> On 26.11.2010 04:02, John Ettedgui wrote: >>> On Thu, Nov 25, 2010 at 5:08 PM, John Ettedgui <john.ettedgui(a)gmail.com> wrote: >>>> On Thu, Nov 25, 2010 at 4:59 PM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>> On 26.11.2010 02:53, John Ettedgui wrote: >>>>>> On Thu, Nov 25, 2010 at 3:34 PM, John Ettedgui <john.ettedgui(a)gmail.com> wrote: >>>>>>> On Thu, Nov 25, 2010 at 3:15 PM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>> On 26.11.2010 01:11, John Ettedgui wrote: >>>>>>>>> On Thu, Nov 25, 2010 at 2:58 PM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>>>> On 26.11.2010 00:42, John Ettedgui wrote: >>>>>>>>>>> On Thu, Nov 25, 2010 at 11:19 AM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>>>>>> On 25.11.2010 14:21, John Ettedgui wrote: >>>>>>>>>>>>> On Thu, Nov 25, 2010 at 3:40 AM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>>>>>>>> On 25.11.2010 13:09, John Ettedgui wrote: >>>>>>>>>>>>>>> On Thu, Nov 25, 2010 at 2:49 AM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>>>>>>>>>> On 25.11.2010 06:12, John Ettedgui wrote: >>>>>>>>>>>>>>>>> I'm actually trying to get a 5.1 setup working without passthrough, if >>>>>>>>>>>>>>>>> I use mplayer with passthrough my receiver behaves just fine, but I am >>>>>>>>>>>>>>>>> trying to send the 6 channels from the computer. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I was actually told that the current ALSA driver for radeon's hdmi >>>>>>>>>>>>>>>>> does not support more than 2 channels, and that would explain why I'm >>>>>>>>>>>>>>>>> getting this behavior. Is that true though? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Yes. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> However, strangely your log output seems to show the intel's hdmi parser >>>>>>>>>>>>>>>> being used instead: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> ALSA sound/pci/hda/patch_intelhdmi.c:842: hdmi_setup_stream: NID=0x2, >>>>>>>>>>>>>>>>> stream=0x1, new-format=0x11 >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Were you just trying some patch at the time? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> yes Wu gave me a patch to try against the kernel, I'm guessing this is >>>>>>>>>>>>>>> the reason you see something unexpected. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Did 6 channels work with that patch? (possibly with a wrong channel order) >>>>>>>>>>>>> Not any better as far as I remember. >>>>>>>>>>>> >>>>>>>>>>>> OK. Did 2 channel audio continue to work with it? >>>>>>>>>>>> >>>>>>>>>>>> If you are not sure, here's the patch doing the same thing against >>>>>>>>>>>> current alsa driver: >>>>>>>>>>>> http://stuff.onse.fi/0001-ALSA-hda-Use-generic-HDMI-code-for-ATI-HDMI-codec… >>>>>>>>>>>> >>>>>>>>>>>> Confirming that would help (it would allow to switch the ATI chip to use >>>>>>>>>>>> the generic parser instead of the ATI-specific one), though not with the >>>>>>>>>>>> multichannel issue (see below). >>>>>>>>>>>> >>>>>>>>>>>> [...] >>>>>>>>>>>> >>>>>>>>>>>>>>> Since I just moved to jack/rca cables I'm in no hurry anymore, but is >>>>>>>>>>>>>>> there any reason why the driver cannot work with more than 2 channels? >>>>>>>>>>>>>> >>>>>>>>>>>>>> I don't really know (it is simply not implemented). >>>>>>>>>>>>> Alright. >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Anything I could do to help with that? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Possibly. But see above, so that we won't try the same things you >>>>>>>>>>>>>> already tried :) >>>>>>>>>>>>>> >>>>>>>>>>>>> Sure :) >>>>>>>>>>>> >>>>>>>>>>>> Looking again at the the alsa info output you provided, I don't think I >>>>>>>>>>>> have any great ideas. It looks like the chip simply reports that it >>>>>>>>>>>> supports a maximum of 2 channels. >>>>>>>>>>>> >>>>>>>>>>>> One option could be hardcoding "chans = 8;" in generic_hdmi_build_pcms() >>>>>>>>>>>> in sound/pci/hda/patch_hdmi.c (assuming you have the above patch and 2 >>>>>>>>>>>> channel audio works with it). However, I find it rather unlikely that it >>>>>>>>>>>> would make any difference. >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> Anssi Hannula >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Anssi, >>>>>>>>>>> the patch fails on my kernel, I cannot find the hda_codec_preset >>>>>>>>>>> manually in there either. >>>>>>>>>>> Did you want me to try that against a 2.6.37 rc instead of a 2.6.36? >>>>>>>>>> >>>>>>>>>> For 2.6.36 I think you can just use your earlier patch. >>>>>>>>>> >>>>>>>>>> Note that on that kernel generic_hdmi_build_pcms() is >>>>>>>>>> intel_hdmi_build_pcms() in patch_intelhdmi.c. >>>>>>>>>> >>>>>>>>> ok >>>>>>>>>> BTW, I've just been informed by another ATI user that his card was >>>>>>>>>> actually stereo-only, while he thought earlier it was multichannel >>>>>>>>>> capable. So, are you sure your card is a multichannel one? >>>>>>>>>> (if not, it would explain why the card says it is not) >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anssi Hannula >>>>>>>>>> >>>>>>>>> Yes, it works fine in Window so I believe the hardware part is good. >>>>>>>> >>>>>>>> And you are sure it is multichannel PCM, and not e.g. multichannel DTS >>>>>>>> or multichannel AC-3 with on-the-fly compression, which are possible >>>>>>>> with stereo hdmi as well? >>>>>>>> >>>>>>>> Just making sure :) >>>>>>>> >>>>>>>> -- >>>>>>>> Anssi Hannula >>>>>>>> >>>>>>> >>>>>>> Well, I have not super tested it, but the reason I believe it is is >>>>>>> simple, when I use passthrough my receiver says Dolby digital or >>>>>>> something like that, but in Windows just going around or playing games >>>>>>> it says Multichannel. >>>>> >>>>> Sounds ok then. >>>>> >>>>>>> I'll get the patch applied in a few minutes, I was running out of >>>>>>> memory and my system was getting really slow... :) >>>>>>> >>>>>> >>>>>> I am not sure if I did it right but now I don't even get hdmi sound >>>>>> working anymore. >>>>>> It is not listed in aplay-l, and dmesg gives me this: >>>>>> [drm:radeon_dvi_detect] *ERROR* HDMI Type A-1: probed a monitor but >>>>>> no|invalid EDID >>>>>> >>>>>> I've attached the patch I used, since I could not used the previous >>>>>> one I had to hack it, maybe I broke something? >>>>>> (I have not set channels to 8 yet, just changed the parser) >>>>> >>>>> The patch looks ok. The above error message is weird, as it comes from >>>>> the display driver which should not be affected by the patch at all. >>>>> >>>>> I'll look at the code later (tomorrow maybe). Until then I'd advise you >>>>> to retry without the patch and with the patch again, just to make sure >>>>> this is not some unrelated issue... >>>>> Also, post the whole dmesg when it works (2 channel, without patch), and >>>>> when it doesn't (with patch). >>>>> >>>>> -- >>>>> Anssi Hannula >>>>> >>>> Following your message I restarted my computer and that error was >>>> gone, yet there is still nothing display in aplay -l / aplay -L about >>>> hdmi anymore. >>>> I'll go back to my non patched kernel and see what it does. >>>> >>> With the non-patched kernel I get back my hdmi in aplay -l, so it >>> looks like the patch messed up something. >> >> Indeed. >> >>> I really think that when I tried it before it did not do that, but it >>> was so long ago that I could be wrong... >> >> Well, the driver has changed since then, it could be more strict now :) >> >>> Anyway here are the 2 dmesgs (not sure if they'll help though). >> >> Indeed, there doesn't seem to be any difference. Maybe try with >> CONFIG_SND_DEBUG_VERBOSE? >> >> -- >> Anssi Hannula >> > > Sure I'll get that. > There you go.

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