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

[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

2 2

[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

1 0

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

1 0

[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

2 1

[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) {

1 0

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

2 3

[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; }

2 1

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.

1 0

[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

2 24

[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

3 5