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

[alsa-devel] Sound problem in Toshiba
by Furkan Cayci 17 Jun '07

17 Jun '07

Hi, I am using Ubuntu 7.04. After installing Ubuntu, I could't hear any sound from my Toshiba Satellite A205-S4577, so I did the things related with this article; http://pastebin.ca/522731 After restarting laptop, I didn't take any errors, but there was no sound again. Then I realized the output of the dmesg. $ dmesg|grep alsa [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 18.948000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 19.672000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 19.680000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) [ 33.572000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:1810: hda_codec: model 'auto' is selected [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 33.620000] ALSA /home/leonetix/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 34.360000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 34.368000] ALSA /home/leonetix/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) ------------------------- So, there is something wrong, but I didn't understand and I was wondering if you could solve or show a way to do it. Also, here is the output of lcpci -v "00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Toshiba America Info Systems Unknown device ff10 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at f0b40000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0" Thanks for your help, Furkan

1 0

[alsa-devel] Software mixing according to priority of streams
by Nobin Mathew 15 Jun '07

15 Jun '07

Hi, Is there any sound mixer which does mixing according to priority of streams, i.e. i will have streams with different priority like audio stream , bluettoth sco stream, gps navigation sound stream etc. Since my hardware will not support hardware mixing i will have to do this mixing in software. If bluetooth sco stream is opened while audio stream is playing then the application (sound server) should stop the playback of audio stream and should start playback of bletooth sco stream. priority order is bluettoth sco > gps navigation sound > audio Can dmix and Jack server the purpose. Nobin Mathew

3 8

[alsa-devel] [RFC][ASoC] New device/driver arch
by Liam Girdwood 15 Jun '07

15 Jun '07

I'd like to gather some feedback on some changes I've been making to ASoC recently. The changes were made to fit in better with the PPC device tree and to help with easier porting to new platforms. They were also introduced to make dealing with device probe errors (e.g. I2C probe failure) a little easier. The largest change is the introduction of an ASoC bus device. All component drivers (that is, codec, codec DAI, CPU DAI, and DMA PCM drivers) register with this bus. This should simplify device registration and hopefully better fit in with the PPC model. A component driver now typically consists of the standard driver ops and a set of ASoC/ALSA ops to configure audio. The new soc.h shows this here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… Example component drivers are here :- I2S http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… DMA http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… WM8753 Codec http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… Another major change is that there is no more snd_dai_link (structure that connected codec <--> CPU). This has been replaced by an ASoC PCM device that describes each PCM device in the machine. This new ASoC PCM device has it's own driver and registers with the bus (like the others). This allows an ASoC PCM driver to be defined like :- static struct snd_soc_device_driver hifi_pcm_drv = { .type = SND_SOC_BUS_TYPE_PCM, .driver = { .name = "mainstone-hifi-pcm", .owner = THIS_MODULE, .probe = hifi_pcm_probe, .remove = hifi_pcm_remove, }, .components = { .cpu_dai = pxa2xx_i2s, .codec = wm8753_codec, .codec_dai = wm8753_hifi_dai, .platform = pxa2xx_pcm, }, }; This now shows each component within the driver structure. The snd_soc_machine driver has changed and can now contain numerous ASoC PCM devices. It is now a platform driver. Example machine driver for Mainstone with WM8753 is here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… The machine driver now also does the I2C/SPI probing (instead of the codec) and also does any codec IO (to better support multiple controllers). Fwiw, I've not tested any of this code atm. I plan to start this tomorrow on pxa2xx and i.MX31 and would like to get some feedback before porting remaining platforms and codecs over. Liam

4 6

[alsa-devel] Documentation for params_xxx() macros?
by Timur Tabi 14 Jun '07

14 Jun '07

Is there documentation that describes each of the params_xxx() macros? If not, then which macro tells me how many periods there are? Is it params_periods()? -- Timur Tabi Linux Kernel Developer @ Freescale

1 0

[alsa-devel] aplay hangs with execve("/build/bin/aplay", ["aplay"], [/* 8 vars */
by Pharaoh . 14 Jun '07

14 Jun '07

Hi I am using alsa lib and alsa utils 1.0.14 on my omap based board. When I try to test my my alsa driver using aplay it just hangs at the very first step. The strace output shows nothing else except the following line, execve("/build/bin/aplay", ["aplay"], [/* 8 vars */ What I am missing?

1 0

[alsa-devel] Drivers for M-Audio Firewire 410
by Cédric 14 Jun '07

14 Jun '07

Hello, Im a french user of Linux (Ubuntu Studio 7.04) and cant install my new hardware (M-Audio Firewire 410). Ive send a mail to the M-Audios technical support, but they have no solution. They recommended me to contact yourself. Could you help me for this problem. Do you need more informations about my computer hardware ? Syncerly. Cédric

2 1

[alsa-devel] [PATCH]is_power_of_2-sound/pcm_params.h
by vignesh babu 14 Jun '07

14 Jun '07

Replacing (n & (n-1)) in the context of power of 2 checks with is_power_of_2 Signed-off-by: vignesh babu <vignesh.babu(a)wipro.com> --- diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 85cf1cf..dee298e 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -21,6 +21,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ +#include <linux/log2.h> int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, @@ -170,7 +171,7 @@ static inline int snd_mask_single(const struct snd_mask *mask) for (i = 0; i < SNDRV_MASK_SIZE; i++) { if (! mask->bits[i]) continue; - if (mask->bits[i] & (mask->bits[i] - 1)) + if (!is_power_of_2(mask->bits[i])) return 0; if (c) return 0; -- Vignesh Babu BM _____________________________________________________________ "Why is it that every time I'm with you, makes me believe in magic?" The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[alsa-devel] [PATCH] Fix build error with strict mm checking
by Martin Habets 13 Jun '07

13 Jun '07

Turning on STRICT_MM_TYPECHECKS in include/asm/page.h causes a build error for sparc32. This patch against 2.6.21 fixes this. Martin Signed-off-by: Martin Habets <errandir_news(a)mph.eclipse.co.uk> --- --- linux/sound/core/sgbuf.c.orig 2007-06-06 23:53:34.000000000 +0100 +++ linux/sound/core/sgbuf.c 2007-06-06 23:54:45.000000000 +0100 @@ -96,7 +96,7 @@ void *snd_malloc_sgbuf_pages(struct devi } sgbuf->size = size; - dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); + dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, __pgprot(PAGE_KERNEL)); if (! dmab->area) goto _failed; return dmab->area;

2 1

[alsa-devel] Trouble understanding ALSA's DMA buffers
by Timur Tabi 12 Jun '07

12 Jun '07

I'm writing an ALSA SOC driver for an I2S-based device, and I'm having a really hard time understanding how ALSA uses the DMA buffers. And yes, I've read the documentation and studied some sample source code. I used to write audio drivers for a living, but that was many years ago, and it wasn't for Linux. Perhaps the concepts in my head are outdated, but I just don't see enough explanation as to how DMA buffers are supposed to work. Back then, audio drivers used "ping pong" DMA buffers. A single DMA buffer is allocated, and the audio hardware is programmed to read from that buffer in a loop. That is, it would automatically restart reading from the beginning of the buffer without any reprogramming. The hardware would also be programmed to issue an interrupt when it got to the end of the buffer, and when it got to the half-way point. To start playback, you first filled the whole buffer with audio data, and then told the hardware to start playing. After the hardware got to the half-way point, it would issue an interrupt. You would then tell the OS you need more data, and you'd get it. You then copy that data into the first half of the buffer *while* the hardware was playing the second half. Later, the hardware would interrupt you when it got to the end of the buffer. You'd then copy more data to the 2nd half while the hardware is playing the first half. And so - hardware plays one half while you copy data to the other half. Hence, "ping pong". So how do I implement this in ALSA? The "Writing an ALSA Driver" document doesn't even contain the words "ping" or "pong". -- Timur Tabi Linux Kernel Developer @ Freescale

4 6

[alsa-devel] [ASoC] bug in soc_pcm_open
by Matthieu CASTET 12 Jun '07

12 Jun '07

Hi, it seems there is a bug in soc_pcm_open. In some case [1] we are in a error path, but we don't set the ret return value. This cause bug/crash from alsa core. Thanks Matthieu [1]http://hg-mirror.alsa-project.org/alsa-kernel/file/38926de0e79c/soc/soc-c… if (!runtime->hw.rates) { 218 printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", 219 codec_dai->name, cpu_dai->name); 220 goto machine_err; 221 } 222 if (!runtime->hw.formats) { 223 printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", 224 codec_dai->name, cpu_dai->name); 225 goto machine_err; 226 } 227 if (!runtime->hw.channels_min || !runtime->hw.channels_max) { 228 printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", 229 codec_dai->name, cpu_dai->name); 230 goto machine_err; 231 }

4 4