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

[alsa-devel] snd_pcm_hw_params_set_buffer_time_near: invalid argument
by Jean-Yves Avenard 04 Apr '11

04 Apr '11

Hi there. I'm a bit puzzled as to what values are considered valids and which one aren't when it comes to setting the buffer time with snd_pcm_hw_params_set_buffer_time_near. For example setting a value of 295ms : all ok. From 300ms to 450ms -> "Invalid argument". but 500ms for example is okay. What's the logic behind it ? It's obviously related to the sampling rate, the audio format and the number of channels ; as the values giving me "Invalid argument" change then. Any comments would be greatly appreciated. Thanks! JY

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[alsa-devel] [PATCH 1/2] ALSA: ASoc: TLV320AIC3X: ad SPI and clock on GPIO2 or BCLK
by Prchal Jiří 04 Apr '11

04 Apr '11

Hi, this patch ads SPI communication for codecs TLV320AIC3X and clock input on GPIO2 or BCLK. Tested on TLV320AIC3106 and AT91SAM9260. TODO: Set the model in SPI probe the right way. I don't know how. This codec communicates on SPI in 1st byte: 7 MSB is register address, 1 LSB is read/write, 2nd byte data. For this reason there is new functions in soc-cache.c snd_soc_7_8_*. Kernel version: 2.6.38 Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/codecs/Kconfig /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/Kconfig --- linux-2.6.38-vanilla/sound/soc/codecs/Kconfig 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/Kconfig 2011-03-17 08:37:14.997500191 +0100 @@ -37,7 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER - select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TLV320AIC3X if SND_SOC_I2C_AND_SPI select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE @@ -118,7 +118,7 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate - + config SND_SOC_ADS117X tristate diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/codecs/tlv320aic3x.c /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/tlv320aic3x.c --- linux-2.6.38-vanilla/sound/soc/codecs/tlv320aic3x.c 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/tlv320aic3x.c 2011-03-24 09:34:58.769370117 +0100 @@ -2,7 +2,9 @@ * ALSA SoC TLV320AIC3X codec driver * * Author: Vladimir Barinov, <vbarinov(a)embeddedalley.com> + * Jiri Prchal, <jiri.prchal(a)aksignal.cz * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * (C) 2011 AK signal Brno * * Based on sound/soc/codecs/wm8753.c by Liam Girdwood * @@ -42,6 +44,7 @@ #include <linux/regulator/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -984,6 +987,13 @@ static int aic3x_set_dai_sysclk(struct s { struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + u8 data; + + /* set external clock on GPIO2 or BCLK */ + data = snd_soc_read(codec, AIC3X_CLKGEN_CTRL_REG); + data &= 0x0f; + data |= ((clk_id << PLLCLK_IN_SHIFT) | (clk_id << CLKDIV_IN_SHIFT)); + snd_soc_write(codec, AIC3X_CLKGEN_CTRL_REG, data); aic3x->sysclk = freq; return 0; @@ -1371,9 +1381,12 @@ static int aic3x_probe(struct snd_soc_co aic3x->codec = codec; codec->dapm.idle_bias_off = 1; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); + if (aic3x->control_type == SND_SOC_I2C) + ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); + else + ret = snd_soc_codec_set_cache_io(codec, 7, 8, aic3x->control_type); if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; } @@ -1472,6 +1485,54 @@ static struct snd_soc_codec_driver soc_c .resume = aic3x_resume, }; +#if defined(CONFIG_SPI_MASTER) +static int aic3x_spi_probe(struct spi_device *spi) +{ + struct aic3x_pdata *pdata = spi->dev.platform_data; + struct aic3x_priv *aic3x; + int ret; + + aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); + if (aic3x == NULL) { + dev_err(&spi->dev, "failed to create private data\n"); + return -ENOMEM; + } + + aic3x->control_type = SND_SOC_SPI; + spi_set_drvdata(spi, aic3x); + + if (pdata) { + aic3x->gpio_reset = pdata->gpio_reset; + aic3x->setup = pdata->setup; + } else { + aic3x->gpio_reset = -1; + } + + aic3x->model = AIC3X_MODEL_3X; + + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); + if (ret < 0) + kfree(aic3x); + return ret; +} + +static int __devexit aic3x_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver aic3x_spi_driver = { + .driver = { + .name = "tlv320aic3x-codec", + .owner = THIS_MODULE, + }, + .probe = aic3x_spi_probe, + .remove = __devexit_p(aic3x_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * AIC3X 2 wire address can be up to 4 devices with device addresses @@ -1557,6 +1618,13 @@ static int __init aic3x_modinit(void) ret); } #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&aic3x_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register AIC3X SPI driver: %d\n", + ret); + } +#endif return ret; } module_init(aic3x_modinit); @@ -1566,6 +1634,9 @@ static void __exit aic3x_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&aic3x_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&aic3x_spi_driver); +#endif } module_exit(aic3x_exit); diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/codecs/tlv320aic3x.h /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/tlv320aic3x.h --- linux-2.6.38-vanilla/sound/soc/codecs/tlv320aic3x.h 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/codecs/tlv320aic3x.h 2011-03-22 13:37:54.751566089 +0100 @@ -178,6 +178,13 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_SHIFT 4 +#define CLKDIV_IN_SHIFT 6 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 + /* Software reset register bits */ #define SOFT_RESET 0x80 diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/soc-cache.c /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/soc-cache.c --- linux-2.6.38-vanilla/sound/soc/soc-cache.c 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/soc-cache.c 2011-03-24 09:23:00.984373772 +0100 @@ -99,6 +99,58 @@ static int snd_soc_4_12_spi_write(void * #define snd_soc_4_12_spi_write NULL #endif +/* special functions for codecs with 7 bit register address and LSB read/write (like TLV320AIC3X) */ +static unsigned int snd_soc_7_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + int ret; + unsigned int val; + + if (reg >= codec->driver->reg_cache_size || + snd_soc_codec_volatile_register(codec, reg)) { + if (codec->cache_only) + return -1; + + BUG_ON(!codec->hw_read); + return codec->hw_read(codec, ((reg << 1) | 1)); + } + + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; +} + +static int snd_soc_7_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + int ret; + + data[0] = (reg << 1); + data[1] = value; + + if (!snd_soc_codec_volatile_register(codec, reg) && + reg < codec->driver->reg_cache_size) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } + + if (codec->cache_only) { + codec->cache_sync = 1; + return 0; + } + + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -661,6 +713,11 @@ static struct { .spi_write = snd_soc_4_12_spi_write, }, { + .addr_bits = 7, .data_bits = 8, + .write = snd_soc_7_8_write, .read = snd_soc_7_8_read, + .spi_write = snd_soc_8_8_spi_write, + }, + { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, .spi_write = snd_soc_7_9_spi_write,

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[alsa-devel] Multiple sound problems with ALC888 (1025:013f) on Acer Aspire 5930
by Elisavet Vorgia 02 Apr '11

02 Apr '11

Hi All, I have sound issues with my laptop, I cannot get 5.1 sound and jack sensing does not work (but works on Windows 7; dual-boot). I have an Acer Aspire 5930 (chipset: ALC888) running Ubuntu 10.10 (64-bit). I have tested with Alsa 1.0.23 (which comes with Ubuntu 10.10) and also compiled Alsa 1.0.24. Here is the Alsa-info with Alsa 1.0.24 (latest), and "model=auto", http://www.alsa-project.org/db/?f=2cbab ... 31dc6a7d70 <http://www.alsa-project.org/db/?f=2cbab995516f4b56a7faecf537f61f31dc6a7d70> Here is alsa-info with "model=acer-aspire-6530g", http://www.alsa-project.org/db/?f=40992 ... 17160cde64 <http://www.alsa-project.org/db/?f=409923a3d8f042804e61f21c7b8ae917160cde64> However, on Linux, 1. the sound volume is too low compared to Windows 7. For example, 50% on the volume bar in Ubuntu is roughly equivalent to 10% in Windows 7. 2. the subwoofer control bar in Sound Preferences Settings seems to control the master volume bar. 3. it seems that only 2 speakers actually work (Front Left and Front Right) as tested with speaker-test -c6 -twav 4. when I plug in the headphones jack the internal speakers don't get muted, instead I have combined sound output. I tried to change the model from auto to "acer" (tested all the acer and the 6stack-dig models and settled down to acer-aspire-6530g as a popular suggestion) but it didn't change much. (You'll find the alsa-info.txt as an attachment to this message.) Is there any possible solution or suggestion for this problem? I am happy to try out patches. Thanks, Elisavet

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[alsa-devel] export pcm_debug_name?
by Eliot Blennerhassett 01 Apr '11

01 Apr '11

Currently defined locally in pcm_lib.c, this function would be useful for debug output from drivers too? Any objection to exporting this function? > static void pcm_debug_name(struct snd_pcm_substream *substream, > char *name, size_t len) > { > snprintf(name, len, "pcmC%dD%d%c:%d", > substream->pcm->card->number, > substream->pcm->device, > substream->stream ? 'c' : 'p', > substream->number); > } -- Eliot Blennerhassett AudioScience Inc.

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Re: [alsa-devel] Linux 2.6.38 freeze because of sound/core/pcm_lib.c commit 59ff878ffb26bc0be812ca8295799164f413ae88
by Clemens Ladisch 01 Apr '11

01 Apr '11

Christopher K. wrote: > Kernel 2.6.38 has been freezing ever since rc1 on my asus p7p55d. > After booting up, the kernel seems to use a lot of cpu-time and after > 20-30 seconds > the system comes to a complete halt. > > I just finished bisecting 2.6.37 - 2.6.38-rc1 and found this commit to > be responsible: > > 59ff878ffb26bc0be812ca8295799164f413ae88 Please try this patch: --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -425,6 +425,8 @@ static int snd_pcm_hw_params(struct snd_ runtime->no_period_wakeup = (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); + runtime->hw_ptr_buffer_jiffies = + max(1uL, (runtime->buffer_size * HZ) / runtime->rate); bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; @@ -866,8 +868,6 @@ static void snd_pcm_post_start(struct sn struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; - runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / - runtime->rate; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0)

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Re: [alsa-devel] Question about your DSP topic
by Patrick Lai 01 Apr '11

01 Apr '11

Hi Liam/Mark, I ran some tests on top of soc-dsp framework pulled from Liam's topic/dsp-upstream(http://git.kernel.org/?p=linux/kernel/git/lrg/asoc- 2.6.git;a=summary). I found there is a scenario that soc-dsp framework erroneously start pcm playabck/capture. Here is the scenario: In the platform driver, I have this route table defined FE1 Playback -> Mixer 1 -> BE1 BE2 -> FE1 Capture FE = Front-end, BE = Back-end While PCM playback is going from FE1 playback to BE1, I switch off FE1 playback to Mixer 1. This caused soc_dsp_runtime_update called. Framework correctly close BE1 as it is no longer needed. Eventually, framework finds BE2 is connected to FE1 capture. Framework, without checking if FE1 capture is activated by user-space application, simply goes ahead activate BE2. Since FE1 capture is never activated, runtime structure is not allocated. This inherently results NULL pointer dereference exception. For now, in soc-dsp.c be_connect function(), I have a check to make sure fe->dsp[stream].runtime is not NULL. I don't know if it's appropriate fix. Can you please take a look? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] Query - ASoC headphone detection stereo vs. mono
by Patrick Lai 31 Mar '11

31 Mar '11

I would like to get a better understanding on how to propagate channel mode of inserted headphone through soc-jack framework? I looked at jack.h and I don't see any definition of channel mode. I would assume that user-space needs this information in order to set up the path in the audio CODEC properly such as mixing stereo content into mono stream inside CODEC. How is use case handled on other platform? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] git trees
by Steve Calfee 31 Mar '11

31 Mar '11

Hi Mark, I am starting to look at a new codec driver. I have been looking at the alsa-project wiki describing the various git trees. I am confused because they seem to be lagging the linux releases (ie 2.6.38 is out, not in the two trees I have tried. The wiki proposed occasional developers use 3 trees - not clear what is in each. ie: git clone http://git.alsa-project.org/http/alsa-driver.git alsa-driver git clone http://git.alsa-project.org/http/alsa-kmirror.git alsa-kmirror if you like do only changes in code in alsa-kmirror, please, use alsa-kernel repository git clone git://git.alsa-project.org/alsa-kernel.git alsa-kernel I really don't understand the sentence: "if you like do only changes in code in alsa-kmirror, please, use alsa-kernel repository" When I built alsa-kernel, I kernel version 2.6.38-rc8+ - which built and boots on my target. I also cloned the kernel.org broonie/sound-2.6.git tree But when I look at branches I see: asoc$ git branch -r origin/ac97-runtime-pm origin/ads117x origin/asoc-regulator origin/asoc-v2-dev origin/bias-off origin/dapm origin/dapm-log origin/dev origin/ep93xx origin/for-2.6.29 origin/for-2.6.30 origin/for-2.6.31 origin/for-2.6.32 origin/for-2.6.33 origin/for-2.6.34 origin/for-2.6.35 origin/for-2.6.36 origin/for-2.6.37 origin/for-2.6.38 origin/for-2.6.39 origin/for-2.6.40 origin/for-next origin/init-card origin/new-maintainers origin/pending origin/pxa-ssp origin/refactoring origin/reg-cache origin/tegra-arch origin/twl4030-mfd origin/untested So I checked out origin/for-2.6.39 when i built that I got a kernel version 2.6.38-rc1 or something. (and it built but didn't boot) What I need to know is which tree and which branch or tag I should use to develop on and to submit alsa patches from. Thanks for the help. Regards, Steve

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[alsa-devel] config file documentation?
by Chris Furlough 31 Mar '11

31 Mar '11

Greetings fellow developers! I'm working on configuring a USB sound device, and running into the occasional difficulty. During these times of duress, I turn to the web to look for documentation on the specific facet that I'm struggling with at the moment. Without fail, I'll come across some feature of the config files that I didn't previously know of. My question is this. Does there exist a comprehensive compendium of config file settings, and options that I can peruse? My current config is a hacked together menagerie of bit and pieces gleaned from many internet sources. I'd like to do it *RIGHT*. Thanks!

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[alsa-devel] ALSA hw sample rate conversion support
by Aisheng Dong 31 Mar '11

31 Mar '11

Hi ALL, Does ALSA core support hw sample rate conversion now? Our soc has a HW sample rate converter module inside, I wonder if current ALSA core provide interface to support hw sample rate conversion? Thanks Regards Aisheng

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