December 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Rename rt562[1|2]_vol_snd_controls to alc562[1|2]_vol_snd_controls
by Axel Lin 20 Dec '11

20 Dec '11

The module desciption says this is ASoC alc5621/2/3 driver. Make the naming consistent with the reset of the code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index da97f02..6a9b621 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -99,7 +99,7 @@ static const unsigned int boost_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); -static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { +static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", @@ -110,7 +110,7 @@ static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { ALC5623_HP_OUT_VOL, 15, 7, 1, 1), }; -static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = { +static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", @@ -925,12 +925,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) switch (alc5623->id) { case 0x21: - snd_soc_add_controls(codec, rt5621_vol_snd_controls, - ARRAY_SIZE(rt5621_vol_snd_controls)); + snd_soc_add_controls(codec, alc5621_vol_snd_controls, + ARRAY_SIZE(alc5621_vol_snd_controls)); break; case 0x22: - snd_soc_add_controls(codec, rt5622_vol_snd_controls, - ARRAY_SIZE(rt5622_vol_snd_controls)); + snd_soc_add_controls(codec, alc5622_vol_snd_controls, + ARRAY_SIZE(alc5622_vol_snd_controls)); break; case 0x23: snd_soc_add_controls(codec, alc5623_vol_snd_controls, -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Use dai_fmt in edb93xx machine driver
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/ep93xx/edb93xx.c | 14 ++------------ 1 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index 6b90c75..9f6fecd 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -48,18 +48,6 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, else mclk_rate = rate * 64 * 2; - err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS); - if (err) - return err; - - err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS); - if (err) - return err; - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, SND_SOC_CLOCK_IN); if (err) @@ -80,6 +68,8 @@ static struct snd_soc_dai_link edb93xx_dai = { .cpu_dai_name = "ep93xx-i2s", .codec_name = "spi0.0", .codec_dai_name = "cs4271-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &edb93xx_ops, }; -- 1.7.5.4

4 4

[alsa-devel] [PATCH] ASoC: Remove export of s3c_pcm_dai
by Axel Lin 20 Dec '11

20 Dec '11

We don't need to export s3c_pcm_dai after multi-component patch. Thus remove export of s3c_pcm_dai and make it static. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/pcm.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index beef63f..786811d 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -478,7 +478,7 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .formats = SNDRV_PCM_FMTBIT_S16_LE, \ } -struct snd_soc_dai_driver s3c_pcm_dai[] = { +static struct snd_soc_dai_driver s3c_pcm_dai[] = { [0] = { .name = "samsung-pcm.0", S3C_PCM_DAI_DECLARE, @@ -488,7 +488,6 @@ struct snd_soc_dai_driver s3c_pcm_dai[] = { S3C_PCM_DAI_DECLARE, }, }; -EXPORT_SYMBOL_GPL(s3c_pcm_dai); static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) { -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: dsd1791: Introduce driver for TI DSD1791 stereo codec
by Michael Williamson 20 Dec '11

20 Dec '11

This patch introduces a (spi) codec driver for the Texas Instruments DSD1791 24 bit audio stereo DAC. http://www.ti.com/product/dsd1791 Testing for basic operation using 16 and 24 bit I2S mode has been performed using a MityDSP-L138 SOM and an Industrial I/O board from Critical Link. Signed-off-by: Michael Williamson <michael.williamson(a)criticallink.com> --- This patch incorporates changes from the original RFC as a result of comments received from Mark Brown, Leon Romanovsky, and Lars-Peter Clausen. Thanks. Summary of Changes: - Use devm_kzalloc() - use regmap cached I/O feature of framework - simplify code applying codec fmt configuration - clean up section attributes on local functions - make local functions/variables static where appropriate - remove set_sysclk method - remove unused tlv header file - remove chatter - strip "-codec" from driver name - fixup driver variable name for consistency - Add Left and Right volume control - Add digital mute sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dsd1791.c | 256 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 262 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/dsd1791.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..95b7969 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 + select SND_SOC_DSD1791 if SPI_MASTER select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C @@ -205,6 +206,9 @@ config SND_SOC_DA7210 config SND_SOC_DFBMCS320 tristate +config SND_SOC_DSD1791 + tristate + config SND_SOC_DMIC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..d6b5f6a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -21,6 +21,7 @@ snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o +snd-soc-dsd1791-objs := dsd1791.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o @@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_DSD1791) += snd-soc-dsd1791.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dsd1791.c b/sound/soc/codecs/dsd1791.c new file mode 100644 index 0000000..41bbc61 --- /dev/null +++ b/sound/soc/codecs/dsd1791.c @@ -0,0 +1,256 @@ +/* + * ALSA SoC codec driver for Texas Instruments DSD1791. + * + * Author: (C) Michael Williamson <michael.williamson(a)criticallink.com> + * Copyright: (C) 2011 Critical Link, LLC + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/spi/spi.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> + +#define DSD1791_REG_DIGATT_L 16 +#define DSD1791_REG_DIGATT_R 17 +#define DSD1791_REG_AUDFMT 18 +#define DSD1791_REG_SRST 20 + +#define DSD1791_FMT_16RJ (0<<4) +#define DSD1791_FMT_20RJ (1<<4) +#define DSD1791_FMT_24RJ (2<<4) +#define DSD1791_FMT_24LJ (3<<4) +#define DSD1791_FMT_16I2S (4<<4) +#define DSD1791_FMT_24I2S (5<<4) +#define DSD1791_FMT_MASK 0x70 + +/* DSD1791 register cache (16 through 23 are used) */ +static const u8 dsd1791_reg[] = { + [16] = 0xFF, + [17] = 0xFF, + [18] = 0x50, + [19] = 0x00, + [20] = 0x00, + [21] = 0x01, + [22] = 0x00, + [23] = 0x00, +}; + +struct dsd1791 { + struct spi_device *spi; + struct snd_soc_codec codec; + int dai_fmt; + int pcm_fmt; +}; + +static int dsd1791_set_format_word(struct dsd1791 *dsd1791, + struct snd_soc_codec *codec) +{ + u8 fmt = 0; + u8 reg; + + switch (dsd1791->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + + case SND_SOC_DAIFMT_I2S: + switch (dsd1791->pcm_fmt) { + case SNDRV_PCM_FORMAT_S16_LE: + fmt = DSD1791_FMT_16I2S; + break; + case SNDRV_PCM_FORMAT_S24_LE: + fmt = DSD1791_FMT_24I2S; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_RIGHT_J: + switch (dsd1791->pcm_fmt) { + case SNDRV_PCM_FORMAT_S16_LE: + fmt = DSD1791_FMT_16RJ; + break; + case SNDRV_PCM_FORMAT_S24_LE: + fmt = DSD1791_FMT_24RJ; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_LEFT_J: + switch (dsd1791->pcm_fmt) { + case SNDRV_PCM_FORMAT_S24_LE: + fmt = DSD1791_FMT_24LJ; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + reg = snd_soc_read(codec, DSD1791_REG_AUDFMT); + reg &= ~(DSD1791_FMT_MASK); + reg |= fmt; + return snd_soc_write(codec, DSD1791_REG_AUDFMT, reg); +} + +static int dsd1791_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg; + + reg = snd_soc_read(codec, DSD1791_REG_AUDFMT); + if (mute) + reg |= 1; + else + reg &= ~1; + return snd_soc_write(codec, DSD1791_REG_AUDFMT, reg); +} + +static int dsd1791_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec); + + dsd1791->pcm_fmt = params_format(params); + + return dsd1791_set_format_word(dsd1791, codec); +} + +static int dsd1791_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec); + + dsd1791->dai_fmt = fmt; + + return dsd1791_set_format_word(dsd1791, codec); +} + +#define DSD1791_RATES SNDRV_PCM_RATE_8000_192000 +#define DSD1791_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops dsd1791_dai_ops = { + .hw_params = dsd1791_hw_params, + .set_fmt = dsd1791_set_fmt, + .digital_mute = dsd1791_mute, +}; + +static struct snd_soc_dai_driver dsd1791_dai = { + .name = "dsd1791", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DSD1791_RATES, + .formats = DSD1791_FORMATS, + }, + .ops = &dsd1791_dai_ops, +}; + +static const struct snd_kcontrol_new dsd1791_snd_controls[] = { + SOC_SINGLE("Left Playback Volume", DSD1791_REG_DIGATT_L, 0, 255, 0), + SOC_SINGLE("Right Playback Volume", DSD1791_REG_DIGATT_R, 0, 255, 0), +}; + +static int dsd1791_probe(struct snd_soc_codec *codec) +{ + u8 reg; + int ret; + struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_SPI); + if (ret) { + dev_err(codec->dev, "Failed to set Cache I/O: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, DSD1791_REG_SRST, 0x40); + if (ret) { + dev_err(codec->dev, "Unable to reset device: %d\n", ret); + goto err; + } + + /* default format after reset */ + dsd1791->dai_fmt = SND_SOC_DAIFMT_I2S; + dsd1791->pcm_fmt = SNDRV_PCM_FORMAT_S24_LE; + + /* enable attenuation control */ + reg = snd_soc_read(codec, DSD1791_REG_AUDFMT); + reg |= 0x80; + snd_soc_write(codec, DSD1791_REG_AUDFMT, reg); + + snd_soc_add_controls(codec, dsd1791_snd_controls, + ARRAY_SIZE(dsd1791_snd_controls)); + return 0; +err: + return ret; +} + +struct snd_soc_codec_driver dsd1791_codec_driver = { + .probe = dsd1791_probe, + .reg_cache_size = ARRAY_SIZE(dsd1791_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = dsd1791_reg, +}; + +static int __devinit dsd1791_spi_probe(struct spi_device *spi) +{ + struct dsd1791 *dsd1791; + + dsd1791 = devm_kzalloc(&spi->dev, sizeof *dsd1791, GFP_KERNEL); + if (!dsd1791) + return -ENOMEM; + + spi_set_drvdata(spi, dsd1791); + + return snd_soc_register_codec(&spi->dev, + &dsd1791_codec_driver, &dsd1791_dai, 1); +}; + +static int __devexit dsd1791_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver dsd1791_spi_driver = { + .driver = { + .name = "dsd1791", + .owner = THIS_MODULE, + }, + .probe = dsd1791_spi_probe, + .remove = __devexit_p(dsd1791_spi_remove), +}; + +static int __init dsd1791_init(void) +{ + return spi_register_driver(&dsd1791_spi_driver); +} +module_init(dsd1791_init); + +static void __exit dsd1791_exit(void) +{ + spi_unregister_driver(&dsd1791_spi_driver); +} +module_exit(dsd1791_exit); + +MODULE_DESCRIPTION("ASoC DSD1791 codec driver"); +MODULE_AUTHOR("Michael Williamson"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

3 4

[alsa-devel] [PATCH] ASoC: Convert uda1380 to table based DAPM and control init
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/uda1380.c | 24 ++++++++---------------- 1 files changed, 8 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 83e45d2..8f734d6 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -373,7 +373,7 @@ static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route uda1380_dapm_routes[] = { /* output mux */ {"HeadPhone Driver", NULL, "Output Mux"}, @@ -410,17 +410,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right PGA", NULL, "VINR"}, }; -static int uda1380_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -764,10 +753,6 @@ static int uda1380_probe(struct snd_soc_codec *codec) break; } - snd_soc_add_controls(codec, uda1380_snd_controls, - ARRAY_SIZE(uda1380_snd_controls)); - uda1380_add_widgets(codec); - return 0; err_free_gpio: @@ -802,6 +787,13 @@ static struct snd_soc_codec_driver soc_codec_dev_uda1380 = { .reg_word_size = sizeof(u16), .reg_cache_default = uda1380_reg, .reg_cache_step = 1, + + .controls = uda1380_snd_controls, + .num_controls = ARRAY_SIZE(uda1380_snd_controls), + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = uda1380_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda1380_dapm_routes), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Convert max9850 to table based DAPM and control init
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/max9850.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 7dfd6e8..47060d2 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -86,7 +86,7 @@ SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route max9850_dapm_routes[] = { /* output mixer */ {"Output Mixer", NULL, "DAC"}, {"Output Mixer", "Line In Switch", "Line Input"}, @@ -293,7 +293,6 @@ static int max9850_resume(struct snd_soc_codec *codec) static int max9850_probe(struct snd_soc_codec *codec) { - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -309,13 +308,6 @@ static int max9850_probe(struct snd_soc_codec *codec) /* set slew-rate 125ms */ snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0); - snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets, - ARRAY_SIZE(max9850_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - snd_soc_add_controls(codec, max9850_controls, - ARRAY_SIZE(max9850_controls)); - return 0; } @@ -328,6 +320,13 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .reg_word_size = sizeof(u8), .reg_cache_default = max9850_reg, .volatile_register = max9850_volatile_register, + + .controls = max9850_controls, + .num_controls = ARRAY_SIZE(max9850_controls), + .dapm_widgets = max9850_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9850_dapm_widgets), + .dapm_routes = max9850_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9850_dapm_routes), }; static int __devinit max9850_i2c_probe(struct i2c_client *i2c, -- 1.7.5.4

2 1

Re: [alsa-devel] [GIT PULL] ARM: amba: Enable module alias autogeneration for AMBA drivers
by Russell King - ARM Linux 19 Dec '11

19 Dec '11

On Mon, Dec 19, 2011 at 01:24:09PM +0000, Dave Martin wrote: > Hi Russell, > > This one isn't urgent, but I'm not seeing the amba modalias patches > anywhere yet. Did you have any outstanding concerns which need to be > resolved? > > If you can suggest when/if these are likely to merge that would be great. > > Since Alessandro is now expecting to have to rebase on top of the amba > additions anyway, we shouldn't need to worry about conflicting with his > patches. > > Of course, if you've already merged these somewhere, then there's no > problem. I have pulled it, I just haven't merged it in anywhere yet.

1 0

Re: [alsa-devel] [LAU] kernel crashes with jackd (with Dell Precision M6600's integrated Intel HDA sound)
by Clemens Ladisch 19 Dec '11

19 Dec '11

Csillag Kristof wrote: > Lately, when experimenting with JACK, I began getting kernel crashes :( > > I am using stock Debian kernel v3.1.5. > > [ 7555.763914] BUG: unable to handle kernel NULL pointer dereference at 0000000000000028 > [ 7555.763952] IP: [<ffffffffa08ca7b8>] setup_bdle.isra.21+0x44/0xd9 [snd_hda_intel] > [ 7555.764735] RAX: 0000000000000000 RBX: ffff880221825e00 RCX: 0000000000000000 > [ 7555.765126] Call Trace: > [ 7555.765170] [<ffffffffa08caa5c>] ? azx_pcm_prepare+0x20f/0x3c7 [snd_hda_intel] > [ 7555.765589] Code: 8b 32 e9 94 00 00 00 81 7d 00 ff 00 00 00 0f 87 96 00 00 00 48 8b 83 c0 00 00 00 44 89 e2 44 89 e6 48 89 df 48 8b 80 d8 02 00 00 > [ 7555.765683] 8b 48 28 48 89 d0 81 e2 ff 0f 00 00 48 c1 e8 0c 48 c1 e0 04 7: 81 7d 00 ff 00 00 00 cmpl $0xff,0x0(%rbp) e: 0f 87 96 00 00 00 ja aa 14: 48 8b 83 c0 00 00 00 mov 0xc0(%rbx),%rax 1b: 44 89 e2 mov %r12d,%edx 1e: 44 89 e6 mov %r12d,%esi 21: 48 89 df mov %rbx,%rdi 24: 48 8b 80 d8 02 00 00 mov 0x2d8(%rax),%rax 2b: ?? 8b 48 28 mov 0x28(%rax),%r?? <-- crash here 2f: 48 89 d0 mov %rdx,%rax 32: 81 e2 ff 0f 00 00 and $0xfff,%edx 38: 48 c1 e8 0c shr $0xc,%rax 3c: 48 c1 e0 04 shl $0x4,%rax substream->runtime->dma_buffer_p is NULL. > https://bugzilla.redhat.com/show_bug.cgi?id=702056 > https://bugzilla.redhat.com/show_bug.cgi?id=718056 > https://bugzilla.redhat.com/show_bug.cgi?id=749186 Same bug. Regards, Clemens

2 1

[alsa-devel] [PATCH] atmel ac97c: using software reset instead hardware reset if not available
by voice 19 Dec '11

19 Dec '11

Using software reset instead hardware reset if not available Signed-off-by: Bo Shen <voice.shen(a)atmel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/atmel/ac97c.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 6e5adde..73516f6 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -899,6 +899,10 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) /* AC97 v2.2 specifications says minimum 1 us. */ udelay(2); gpio_set_value(chip->reset_pin, 1); + } else { + ac97c_writel(chip, MR, AC97C_MR_WRST | AC97C_MR_ENA); + udelay(2); + ac97c_writel(chip, MR, AC97C_MR_ENA); } } -- 1.6.3.3

2 1

[alsa-devel] FW: [PATCH] atmel ac97c: using software reset instead hardware reset if not available
by Shen, Voice 19 Dec '11

19 Dec '11

Hi Sir, Add patch to Atmel AC97C. Object: using software reset instead hardware reset if not available Best Regards, Voice Shen -----Original Message----- From: Nicolas Ferre [mailto:nicolas.ferre@atmel.com] Sent: Thursday, November 24, 2011 18:44 voice To: Shen, Voice Cc: linux-kernel(a)vger.kernel.org; linux-sound(a)vger.kernel.org; Lin, JM; Xu, Hong Subject: Re: [PATCH] atmel ac97c: using software reset instead hardware reset if not available On 11/23/2011 11:05 AM, voice : > Using software reset instead hardware reset if not available > > Signed-off-by: Bo Shen <voice.shen(a)atmel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> You may need to sent it to the sound subsystem mailing-list and maintainers. Anyway, I let you follow this patch... As it is a fixup and will benefit to existing platforms, I think that it can make its way to mainline during 3.2 phase. Best regards, > --- > sound/atmel/ac97c.c | 4 ++++ > 1 files changed, 4 insertions(+), 0 deletions(-) > > diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c > index 6e5adde..73516f6 100644 > --- a/sound/atmel/ac97c.c > +++ b/sound/atmel/ac97c.c > @@ -899,6 +899,10 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) > /* AC97 v2.2 specifications says minimum 1 us. */ > udelay(2); > gpio_set_value(chip->reset_pin, 1); > + } else { > + ac97c_writel(chip, MR, AC97C_MR_WRST | AC97C_MR_ENA); > + udelay(2); > + ac97c_writel(chip, MR, AC97C_MR_ENA); > } > } > -- Nicolas Ferre

2 1