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

[alsa-devel] [PATCH 1/2] ASoC: uda134x: add DATA011 register found in codecs family
by Vladimir Zapolskiy 30 Jun '10

30 Jun '10

In UDA1340, UDA1344 and UDA1345 codecs there is one more functional register in part of DATA0 tranfser. For UDA1341 this register coincides with EA register. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> --- sound/soc/codecs/uda134x.c | 3 ++- sound/soc/codecs/uda134x.h | 5 +++-- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index b3039b1..e770029 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -57,7 +57,7 @@ static const char uda134x_reg[UDA134X_REGS_NUM] = { /* Extended address registers */ 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* Status, data regs */ - 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, + 0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00, }; /* @@ -116,6 +116,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, case UDA134X_DATA000: case UDA134X_DATA001: case UDA134X_DATA010: + case UDA134X_DATA011: addr = UDA134X_DATA0_ADDR; break; case UDA134X_DATA1: diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h index 94f4404..205f03b 100644 --- a/sound/soc/codecs/uda134x.h +++ b/sound/soc/codecs/uda134x.h @@ -23,9 +23,10 @@ #define UDA134X_DATA000 10 #define UDA134X_DATA001 11 #define UDA134X_DATA010 12 -#define UDA134X_DATA1 13 +#define UDA134X_DATA011 13 +#define UDA134X_DATA1 14 -#define UDA134X_REGS_NUM 14 +#define UDA134X_REGS_NUM 15 #define STATUS0_DAIFMT_MASK (~(7<<1)) #define STATUS0_SYSCLK_MASK (~(3<<4)) -- 1.7.0.4

2 10

[alsa-devel] [PATCH] ASoC: DaVinci: Added support for cpu clocking I2S
by Raffaele Recalcati 30 Jun '10

30 Jun '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added audio playback support with [frame sync master - clock master] mode and with [frame sync master - clock slave]. 1. SND_SOC_DAIFMT_CBS_CFS (the cpu generates clock and frame sync) I need to clarify how the McBSP is clocked internally. Reading sprufi3a-1.pdf (TMS320DM36x DMSoC Multichannel Buffered Serial Port (McBSP) Interface) at "2.5.3 Data Clock Generation" paragraph this info not appears really clear if we are in the case of "McBSP internal input clock". I hope some Ti developers can help me. There is a FIXME message in the patch. i2s_fast_clock switch can be used to have better approximate or symmetric waveforms. clk_input_pin board info can be used to select it depending on hw connections 2. SND_SOC_DAIFMT_CBS_CFM (the cpu get clock from external pin and generates frame sync) Clock slave can be important when the external codec need system clock and bit clock synchronized (tested with uda1345). 3. We haven't changed the evmdm365 support (due also to CPLD that doesn't help to understand) We don't know in this mode if audio stereo works on evmdm365. Probably it does. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 16 ++++ sound/soc/davinci/davinci-i2s.c | 126 ++++++++++++++++++++++++++---- 2 files changed, 128 insertions(+), 14 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..f9b6da2 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,16 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not simmetrical, but sampling + * frequency is better approximated + */ + int i2s_fast_clock; + + /* To be used when cpu gets clock from extenal pin */ + int clk_input_pin; + /* McASP specific fields */ int tdm_slots; u8 op_mode; @@ -78,6 +88,12 @@ enum { MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; +enum { + MCBSP_CLKR = 0, /* DM365 */ + MCBSP_CLKS, +}; + + #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..786ade6 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,21 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,6 +149,11 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + int i2s_fast_clock; + unsigned int fmt; + int clk_div; + int clk_input_pin; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -254,10 +264,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; + /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + dev->fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -268,11 +280,18 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + if (dev->clk_input_pin == MCBSP_CLKS) + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + else + /* + * McBSP CLKR pin is the input for the Sample Rate + * Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -372,6 +391,16 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + dev->clk_div = div; + return 0; +} + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,11 +409,12 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; + struct clk *clk; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -396,12 +426,56 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + if (master == SND_SOC_DAIFMT_CBS_CFS) { + clk = clk_get(NULL, "pll1_sysclk6"); + if (clk) + freq = clk_get_rate(clk); + freq = 122000000; /* FIXME ask to Texas */ + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } + clk_div &= 0xFF; + srgr |= clk_div; + } else if (master == SND_SOC_DAIFMT_CBM_CFS) { + /* Clock given on CLKS */ + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + } else { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,12 +500,29 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -442,6 +533,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,6 +595,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; @@ -552,6 +648,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->i2s_fast_clock = pdata->i2s_fast_clock; + dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { -- 1.7.0.4

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[alsa-devel] [PATCH v2] ASoC: ak4642: Add Digital Playback Volume control
by Kuninori Morimoto 30 Jun '10

30 Jun '10

This patch add Digital Playback Volume control. And it moves default digital playback volume settings from ak4642_dai_startup to ak4642_init Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 o "Playback Volume" => "Digital Playback Volume" o Add explain on commit log sound/soc/codecs/ak4642.c | 28 +++++++++++++++++++++++++--- 1 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7528a54..7b96311 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -36,6 +36,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -111,6 +112,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -204,7 +222,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -214,8 +231,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -463,6 +478,10 @@ static int ak4642_init(struct ak4642_priv *ak4642) goto reg_cache_err; } + /* default volume */ + ak4642_write(codec, L_DVC, 0x28); + ak4642_write(codec, R_DVC, 0x28); + return ret; reg_cache_err: @@ -548,6 +567,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -- 1.7.0.4

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[alsa-devel] alsamixer-qt4 release
by Sebastian H. 30 Jun '10

30 Jun '10

Hello list Here is (yet another) mixer application for ALSA: alsamixer-qt4 It took a bit longer that expected but today I've erased the last items from the TODO list for the first release version 0.1.0. All files are hosted at sourceforge http://sourceforge.net/projects/alsamixer-qt4/files/ And here's a small web page http://xwmw.org/alsamixer-qt4 At the moment it is a volume only mixer and the dB value is only displayed but the scaling sliders are worth a look (The style is homebrewn ;). So far. Regards, Sebastian H.

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[alsa-devel] snddevices issue
by Adish Kuvelker 29 Jun '10

29 Jun '10

Hello, I find that the device nodes created in the /dev/snd for Control, Timer and pcmC0D0 have different minor numbers from the class devices created by the ALSA core. I just did kernel prints to capture the minor numbers of audio Control, Timer and pcmC0D0p creatd by ALSA core and found that they are different. My ALSA core version is ALSA 1.0.13 and the snddevices file is also from the ALSA 1.0.13. How do I do a workabout for this? Regards; Aadish

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[alsa-devel] alsa-utils configuration query
by Adish Kuvelker 29 Jun '10

29 Jun '10

Hello, I am cross-compiling alsa-utils for my ARM target. I have successfully configured and installed the alsa-lib (alsa-lib-1.0.16). Now to configure the alsa-utils(alsa-utils-1.0.16) i pass the following command LD=arm-linux-ld ./configure --with-alsa-prefix=/usr/lib --build=i686-linux --disable-nls The libasound.la & libasound.a are present in the /usr/lib but the above command gives an error checking for snd_ctl_open in -lasound... no configure: error: No linkable libasound was found Can anyone give me a pointer of how to tackle this? Regards; Aadish

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Re: [alsa-devel] [PATCH] - remove "front" , "surround" , "rear" , "surround40" , "surround51" from output of arecord -L
by Raymond Yau 29 Jun '10

29 Jun '10

2010/6/29 Jerone Young <jerone.young(a)canonical.com> > Hey Raymod, > A bunch of things wrong with this. > > 1) You need to post the whole diff patch .. not must the lines .. > the > diff patch will contain the file this is patching. > > 2) Needs a Signed-off-by: ME <me(a)somewhere.com> line. > > Jerone > > On Tue, 2010-06-29 at 12:05 +0800, Raymond Yau wrote: > > Since arecord -L will list "front", ..... devices which are not suitable > > for recording (e.g. HDA has softvol plugin in front device , emu10k1 and > > other drivers has hook ctl and lock ) > > > > > > > > > > while (*n != NULL) { > > name = snd_device_name_get_hint(*n, "NAME"); > > descr = snd_device_name_get_hint(*n, "DESC"); > > io = snd_device_name_get_hint(*n, "IOID"); > > + if ( stream == SND_PCM_STREAM_CAPTURE ) > > + if (( strstr(descr,"speaker") > 0 ) || ( > > strstr(descr,"Output") > 0 )) > > + goto __end; > > if (io != NULL && strcmp(io, filter) != 0) > > goto __end; > > printf("%s\n", name); > > if ((descr1 = descr) != NULL) { > I am not sure the fix should be at arecord or namehint since those "Output" and "speaker" may be translated to different language and use "front" device for capturing is not recommeded since there are many drivers have hook ctl and lock (e.g. "Channel Mode" , "Line-In As Surround", Surround Down Mix) in front or surroundxx device and emu10k1 have ctl of pcm_route/pcm_volume ATIIXP.conf: lock true ATIIXP-SPDMA.conf: lock true Audigy2.conf: lock true Audigy.conf: lock true Aureon51.conf: lock true Aureon71.conf: lock true CA0106.conf: lock true CMI8338.conf: lock true CMI8738-MC6.conf: lock true CMI8738-MC8.conf: lock true CMI8788.conf: lock true CS46xx.conf: lock true EMU10K1.conf: lock true EMU10K1X.conf: lock true ENS1370.conf: lock true ENS1371.conf: lock true HDA-Intel.conf: lock true ICE1712.conf: lock true ICE1724.conf: lock true ICH4.conf: lock true ICH.conf: lock true NFORCE.conf: lock true PS3.conf: lock true RME9652.conf: lock true SB-XFi.conf: lock true SI7018.conf: lock true TRID4DWAVENX.conf: lock true VIA686A.conf: lock true VIA8233A.conf: lock true VIA8237.conf: lock true VX222.conf: lock true VXPocket.conf: lock true YMF744.conf: lock true

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[alsa-devel] Card information in /proc or /sys
by Florian Faber 29 Jun '10

29 Jun '10

Hi, I want to expose more information about the sound card's ports (names, connectors etc.) to userspace in a standardized manner. Does ALSA have predefined locations for this in /sys, or would /proc (snd_card_proc_new) still be the way to go? Flo -- Machines can do the work, so people have time to think. public key DA43FEF4 x-hkp://wwwkeys.eu.pgp.net

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[alsa-devel] [PATCH] - remove "front" , "surround" , "rear" , "surround40" , "surround51" from output of arecord -L
by Raymond Yau 29 Jun '10

29 Jun '10

Since arecord -L will list "front", ..... devices which are not suitable for recording (e.g. HDA has softvol plugin in front device , emu10k1 and other drivers has hook ctl and lock ) while (*n != NULL) { name = snd_device_name_get_hint(*n, "NAME"); descr = snd_device_name_get_hint(*n, "DESC"); io = snd_device_name_get_hint(*n, "IOID"); + if ( stream == SND_PCM_STREAM_CAPTURE ) + if (( strstr(descr,"speaker") > 0 ) || ( strstr(descr,"Output") > 0 )) + goto __end; if (io != NULL && strcmp(io, filter) != 0) goto __end; printf("%s\n", name); if ((descr1 = descr) != NULL) {

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Re: [alsa-devel] [PATCH 1/4] dump_audiostreaming_interface(): fix ac-3 decoder
by Daniel Mack 28 Jun '10

28 Jun '10

Hi Greg, On Tue, Jun 01, 2010 at 10:06:58AM -0700, Greg KH wrote: > On Tue, Jun 01, 2010 at 06:52:12PM +0200, Daniel Mack wrote: > > Greg, > > > > I believe you're maintaining usb-utils now, right? At least it was > > you posting an announcement for a new release the other day. > > Yes I am. > > > Let me know if I should do anything else to get these patches merged :) > > Let me wake up and get a chance to get around to these patches? :) > > I'll get to them in a day or so, they look great to me. Just a reminder - could you have a look at them already? For easier access, the six patches I sent can now also be pulled from git://github.com/zonque/usbutils.git FWIW, I got positive feedback from people developing USB audio class 2 devices - my patches seem to display the right things for them :) Thanks, Daniel

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