July 2007 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] RE : RE : M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 13 Jul '07

13 Jul '07

> With the help of Hakan I made a fix and updated the doc accordingly. > Hakan could you test this fix (to be applied after my last one) and > report if it is ok for you ? Yes, seem to work quite nicely now with AC3/DTS also in device_setup=0x00 mode. /Håkan

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[alsa-devel] snd-hda-intel auto detect incorrect
by Matt Hirsch 12 Jul '07

12 Jul '07

I have an hp pavilion tx 1000 laptop running fedora 7. The auto detect feature of the snd-hda-intel dirver doesn't work on my sound card. As requested in Documentation/sound/alsa/ALSA-Configuration.txt I am forwarding the following to the developer mailing list: #uname -a Linux laptop 2.6.21-1.3228.fc7 #1 SMP Tue Jun 12 14:56:37 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux # lspci -s 00:10.1 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) # lspci -s 00:10.1 -nv 00:10.1 0403: 10de:026c (rev a2) Subsystem: 103c:30bf Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 17 Memory at c0000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Capabilities: [6c] HyperTransport: MSI Mapping # yum list alsa-\* Installed Packages alsa-lib.x86_64 1.0.14-0.4.rc3.fc7 installed alsa-lib.i386 1.0.14-0.4.rc3.fc7 installed alsa-utils.x86_64 1.0.14-0.7.rc2.fc7 installed With no options I get the following log message and the audio doesn't work: kernel: hda_codec: Unknown model for ALC660VD/ALC861VD, trying auto-probe from BIOS... When loading the module as follows, the audio works: modprobe snd-hda-intel model=3stack I believe the card was detected as 3stack-dig, as the alsamixer options match when either no option or model=3stack-dig are given. Hope this helped. Thanks for all your hard work! Matt

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[alsa-devel] Period Size in hw_params
by Paul Kavan 12 Jul '07

12 Jul '07

I finally have my basic audio applications working. I divided it into capture and playback and began with playback as I had a nice sinewave I could use to playback. Anyway, I think my hangup with getting the thing working was with the buffer and period information. My apps now do this: int periods = 2; /* Number of periods */ snd_pcm_uframes_t periodsize = 1000; /* Periodsize (bytes) */ /* Set number of periods */ if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) { fprintf(stderr, "Error setting periods.\n"); return(-1); } /* Set buffer size */ size = periodsize * periods; if (snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (size)>>2) < 0) { fprintf(stderr, "Error setting buffer size.\n"); return(-1); } I borrowed the period size from using aplay -v and seeing what it used. Setting number of periods to 2 was a little bit of guess work based on what others seem to use. I have looked at the alsa documentation and cannot see how I could have determined these numbers would have worked without trial and error. Can anyone either give me a explanation of how to determine these or a pointer to some decent documentation that talks about this? Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] [PATCH] hda: stac9202 mixer fix
by Matthew Ranostay 12 Jul '07

12 Jul '07

Added support for line out controls for STAC9202 cards, and fixed issue where master mixer control was being created twice for headphone and speaker outs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 273fac42ced9 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Jul 04 15:17:38 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Wed Jul 11 22:13:06 2007 -0400 @@ -334,8 +334,6 @@ static struct snd_kcontrol_new stac9200_ }; static struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0xe, 0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -1401,7 +1399,15 @@ static int stac92xx_auto_create_hp_ctls( continue; add_spec_dacs(spec, nid); } - + for (i = 0; i < cfg->line_outs; i++) { + nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) + continue; + add_spec_dacs(spec, nid); + } for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { static const char *pfxs[] = { "Speaker", "External Speaker", "Speaker2",

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[alsa-devel] Can I mmap buffer allocated using SNDRV_DMA_TYPE_CONTINUOUS?
by Pharaoh . 12 Jul '07

12 Jul '07

Hi, I have allocated the DMA buffer using snd_pcm_lib_malloc_pages() and it was preallocated as: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data (GFP_KERNEL), 128 * 1024, 128 * 1024); Can I mmap such a dma buffer? I think yes... -pharaoh.

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[alsa-devel] [RFC] SGI O2 MACE audio ALSA module
by TJ 12 Jul '07

12 Jul '07

Hi, It's been a while since I last posted something. Here is the latest version of my code. It is somewhat of a mess at the moment, but I plan to tidy it all up when it works better, any // comments are not permanent features. I have been using trial and error to find why it play too fast. I am glad to say that it does play sound correctly, bar one little issue that has me stuck at the moment, it plays too fast (for any sample frequency). Also how fast appears to depend on the player (tried aplay and ogg123). I have cc the ALSA dev too in the hope that between the two lists someone may spot something. Please can anyone who replies cc me. The patch was built against (applies to) linux-2.6.19.7 from linux-mips.org. It'll probably work on other versions. Notes on the module it self... There are two code 'paths' depending on sma_indirect=1|0, this var will be made a module load argument. When 0 ALSA will be writing directly to the MACE ring buffer, currently the state of this code is believed very broken I have not been working on it for a while. When 1 ALSA will be writing to a bounce buffer and the module will copy data to the ring buffer as per the original driver. Currently sma_indirect is hard coded 1 and it is this 'path' i'm currently interested in fixing. The module has some proc files for debugging, the contents of the ad1843 regs, and for values in in mace. There is a _spy version of the module (which wont exist in my final patch) that provides just these proc files without making any (init) changes to any of the hardware (ie read only). This allowed me to see what state the O2 jingle left the various bit in. The module has a limited mixer, which I want to do a lot more with in future, and the front volume buttons are supported. DAC1/ADC map to dev0 on the card, while DAC2 maps to dev1. I don't know what else to say, please have a listen and a look. It will play any samples 8k to 48k, and any bit format that ALSA knows about. Regards, Thorben

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[alsa-devel] [PATCH] Make s3c24xx_i2s_set_clkdiv() change the correct bits
by Matt Reimer 12 Jul '07

12 Jul '07

Make s3c24xx_i2s_set_clkdiv() change the correct bits. Signed-off-by: Matt Reimer <mreimer(a)vpop.net> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 8ca314d..39f0246 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -344,11 +344,11 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, DBG("Entered %s\n", __FUNCTION__); switch (div_id) { - case S3C24XX_DIV_MCLK: + case S3C24XX_DIV_BCLK: reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); break; - case S3C24XX_DIV_BCLK: + case S3C24XX_DIV_MCLK: reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); break; -- 1.5.1.4

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[alsa-devel] [PATCH] Add LG LW20 si3054 modem id
by Claudio Matsuoka 12 Jul '07

12 Jul '07

This patch adds the id of the LG LW20 laptop modem.

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[alsa-devel] Using speaker-test for sound cards which dont support mmap
by Pharaoh . 12 Jul '07

12 Jul '07

Is it possible to use speaker-test for sound cards which do not support mmap? I straced speaker-test and it uses mmap call, can we bypass this?

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[alsa-devel] Help with ALSA application development
by Paul Kavan 12 Jul '07

12 Jul '07

Hello: I am still trying to write my own basic application that captures and plays audio. I am working in an embedded linux environment and have the drivers working well. I can use arecord/aplay without any problem. However, I have been trying to follow some web examples for simple capture and playback without any success. Does anyone know of some sample code for S8 formatted, 8k rate capture and playback. I think my problem is two-fold. First is getting the hw-params set correctly. I have a function call that sets the interface up. I am trying to use the same params that aplay -v returns to me, but perhaps I have them wrong somehow. Secondly is buffering the samples. I am not sure I have that correct. Does anyone have some pointers on that? I have attached my code. If anyone has some time and insight into this, I would greatly appreciate it. Thanks. Paul

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