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

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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[alsa-devel] newbie, arecord
by javi perez 28 Jun '10

28 Jun '10

I am trying to do a speech recognition system. In the aplay file I would like to conver the unsigned char buffer to a float buffer to be able to make comparisons the way I would like. The floats that I am getting after the conversion are these. How should I do the conversion from unsigned char to float array? thanks in advance. 819 -464604244644772930012428885483847680.000000 820 -542490915911345743841838949652758528.000000 821 -495758857691688295559156595670646784.000000 822 -480181610589352498484045935735209984.000000 823 -441239146465853748477054432634208256.000000 824 -376333296573780517983308202832297984.000000 825 -503547144523165508473277153076576256.000000 826 -485374263974618640302095603012206592.000000 827 -495758501164956981369637424722870272.000000 828 -433450265423157678580401923648651264.000000 829 -415277147190123267616207592952430592.000000 830 -425662216276168008459294146874572800.000000 831 -477585818686816398859299858518376448.000000 832 -394508395510877785555942372127277056.000000 833 -472393086073387742776912597697429504.000000 834 -482778551300245054941687119339323392.000000 835 -584029726534518193323939684778246144.000000 836 -511336897075649235277643191045586944.000000 837 -558067806486950226727430438640418816.000000 838 -513933085118997906224077235982172160.000000 839 -302397753578834695235772573852631040.000000 float* float_array; while (rest > 0 && recycle_capture_file == 0) { size_t c = (rest <= (off64_t)chunk_bytes) ? (size_t)rest : chunk_bytes; size_t f = c * 8 / bits_per_frame; if (pcm_read(audiobuf, f) != f) break; float_array=(float*)malloc(f); memcpy(float_array,audiobuf,f); int z=0; for(z=0; z<f/sizeof(float);z++){ printf("%d %f\n",z,float_array[z]); } /*if (write(fd, audiobuf, c) != c) { perror(name); prg_exit(EXIT_FAILURE); }*/ count -= c; rest -= c; fdcount += c; }

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Re: [alsa-devel] [PATCH RFC 1/3] ASoC: uda1380: use callbacks instead of gpiolib
by Mark Brown 28 Jun '10

28 Jun '10

On Mon, Jun 28, 2010 at 04:20:43PM +0200, pHilipp Zabel wrote: > The reset GPIO is connected directly to the UDA1380 RESET pin. The > power source is external. OK, the power source is definitely a regulator then.

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[alsa-devel] [PATCH RFC 0/3] asoc: uda1380 cleanup
by Vasily Khoruzhick 28 Jun '10

28 Jun '10

This patch series cleanups uda1380 codec driver a bit: - it introduces callbacks instead of direct gpiolib usage, as some machines requires more actions than just plain gpio toggle to enable/disable/reset codec; - it makes driver more powersave-friendly and fixes suspend/resume

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[alsa-devel] [PATCH] ASoC: ak4642: Add Playback Volume control
by Kuninori Morimoto 28 Jun '10

28 Jun '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch works for me. But I'm not sure it is correct settings. Please check it carefully 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 9a6157f..ce4e6ec 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -28,6 +28,7 @@ #include <linux/slab.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -103,6 +104,23 @@ 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("Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -190,7 +208,6 @@ 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 @@ -200,8 +217,6 @@ 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); @@ -446,6 +461,10 @@ static int ak4642_init(struct ak4642_priv *ak4642) return ret; } + /* default volume */ + ak4642_write(codec, L_DVC, 0x28); + ak4642_write(codec, R_DVC, 0x28); + return ret; } @@ -524,6 +543,9 @@ 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] [PATCH] sis7019: increase reset delays
by David Dillow 28 Jun '10

28 Jun '10

A few boards using this controller are reported to need a little extra time during their reset cycle. Reported-by: Michael Goeke <michael.goeke(a)icachip.de> Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- Wow, a horrible maintainer am I. Working on the virtual timing period reminded me of of another timing issue from two years back that I forgot to push upstream. This applies on top of the vperiod fixes, but should not conflict if the ordering is changed. diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 614ff6e..1b8f674 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1049,7 +1049,7 @@ static int sis_chip_free(struct sis7019 *sis) /* Reset the chip, and disable all interrputs. */ outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); - udelay(10); + udelay(25); outl(0, sis->ioport + SIS_GCR); outl(0, sis->ioport + SIS_GIER); @@ -1085,7 +1085,7 @@ static int sis_chip_init(struct sis7019 *sis) /* Reset the audio controller */ outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); - udelay(10); + udelay(25); outl(0, io + SIS_GCR); /* Get the AC-link semaphore, and reset the codecs @@ -1098,7 +1098,7 @@ static int sis_chip_init(struct sis7019 *sis) return -EIO; outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); - udelay(10); + udelay(250); count = 0xffff; while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)

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[alsa-devel] [PATCH] ASoC: ak4642: Add ak4642_fill_cache function
by Kuninori Morimoto 28 Jun '10

28 Jun '10

ak4642 driver is used for ak4642/ak4643. So, reading real register value is better than using constant value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 50 +++++++++++++++----------------------------- 1 files changed, 17 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 8d56811..9a6157f 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -106,25 +106,19 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; + u8 reg_cache[AK4642_CACHEREGNUM]; }; static struct snd_soc_codec *ak4642_codec; -/* - * ak4642 register cache - */ -static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0001, 0x0000, - 0x0002, 0x0000, 0x0000, 0x0000, - 0x00e1, 0x00e1, 0x0018, 0x0000, - 0x00e1, 0x0018, 0x0011, 0x0008, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, -}; +static void ak4642_fill_cache(struct snd_soc_codec *codec) +{ + int i; + u8 *cache = codec->reg_cache; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + i2c_master_recv(codec->control_data, cache + i, 1); +} /* * read ak4642 register cache @@ -132,7 +126,7 @@ static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; if (reg >= AK4642_CACHEREGNUM) return -1; return cache[reg]; @@ -144,7 +138,7 @@ static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; if (reg >= AK4642_CACHEREGNUM) return; @@ -175,7 +169,7 @@ static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, static int ak4642_sync(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; int i, r = 0; for (i = 0; i < AK4642_CACHEREGNUM; i++) @@ -432,12 +426,9 @@ static int ak4642_init(struct ak4642_priv *ak4642) codec->dai = &ak4642_dai; codec->num_dai = 1; codec->hw_write = (hw_write_t)i2c_master_send; - codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); - codec->reg_cache = kmemdup(ak4642_reg, - sizeof(ak4642_reg), GFP_KERNEL); - - if (!codec->reg_cache) - return -ENOMEM; + codec->reg_cache_size = AK4642_CACHEREGNUM; + codec->reg_cache = ak4642->reg_cache; + ak4642_fill_cache(codec); ak4642_dai.dev = codec->dev; ak4642_codec = codec; @@ -445,23 +436,17 @@ static int ak4642_init(struct ak4642_priv *ak4642) ret = snd_soc_register_codec(codec); if (ret) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - goto reg_cache_err; + return ret; } ret = snd_soc_register_dai(&ak4642_dai); if (ret) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); snd_soc_unregister_codec(codec); - goto reg_cache_err; + return ret; } return ret; - -reg_cache_err: - kfree(codec->reg_cache); - codec->reg_cache = NULL; - - return ret; } #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -495,7 +480,6 @@ static int ak4642_i2c_remove(struct i2c_client *client) snd_soc_unregister_dai(&ak4642_dai); snd_soc_unregister_codec(&ak4642->codec); - kfree(ak4642->codec.reg_cache); kfree(ak4642); ak4642_codec = NULL; -- 1.7.0.4

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[alsa-devel] ALSA SoC device tree bindings
by Jeremy Kerr 27 Jun '10

27 Jun '10

Hi all, From some discussions at the previous Ubuntu Developer Summit, I've put up a page on the devicetree.org wiki with the ultimate goal of describing device tree bindings for ALSA System on Chip devices. At this stage, we're just looking at how to structure device in the device tree. The bindings for specific devices will come as required (ie, as we add device tree support for them). So - if you're interested in the ALSA and/or device tree areas, please take a look at the notes from UDS: http://devicetree.org/SoC_Audio - if you have any comments or suggestions, please let me know, or add to the page (or the discussion for the page). I intend for this document to become a more formal specification, but we still have a way to go before that happens. If you'd like an introduction to device trees, Grant has a good walkthrough- style document at: http://devicetree.org/Device_Tree_Usage Cheers, Jeremy

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[alsa-devel] Modifying the AD193X driver and integrating with the Blackfin platform
by Adam Rosenberg 27 Jun '10

27 Jun '10

I see that Liam just modified the code (bf5xx-ad193x.c) that I was looking at modifying to add support for the CS42448 CODEC. I would like to understand the architecture of the ASoC and Blackfin TDM PCM driver that it uses because I need to enable the secondary data lines to support multiple chips. Where can I find the documentation for this code? If I want to integrate this code into the uClinux-dist-2009R1.1-RC4 distribution, is it simply a matter of copying the files in "[linux-2.6-asoc.git]/sound/" to the linux-2.6/sound directory for uClinux? Or is it better if I git clone linux-2.6-asoc and move my custom board development files there? Is there a driver installation guide somewhere that I can read about this? I am concerned that the official Blackfin distribution (http://blackfin.uclinux.org/gf/download/frsrelease/473/7285/uClinux-dist-20…) will not be compatible with the ASoC distribution (git://opensource.wolfsonmicro.com/linux-2.6-asoc). Can you tell me if there are any differences between these? To sum up: 1. Where can I find the ASoC documentation? 2. Is there an installation guide to use ASoC with an existing Kernel? 3. What are the differences between the wolfsonmicro.com linux-2.6-asoc and blackfin.uclinux.org uClinux-dist-2009R1.1-RC4? Thank you, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

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[alsa-devel] Detecting if a pcm device is a digital or analog one
by Jean-Yves Avenard 25 Jun '10

25 Jun '10

Hi there... I was wondering if there was any way to accurately distinguish between audio device, if it has an analog or a digital output. I would prefer not parsing the name (like analog71 is obviously an analog device, while "hdmi" is digital) I see that on some devices the IEC958 bit is set, but this doesn't seem reliable (on another PC, the IEC958 bit isn't set). Any suggestions would be greatly appreciated... Thank you Jean-Yves

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