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[alsa-devel] Nvidia hdmi problems
by Nir Tzachar 21 Oct '08

21 Oct '08

Hello, I have been trying to get audio through hdmi to work for some time now. I have cloned the sound-2.6 repository to get the latest drivers, which should support hdmi, AFAIK. Motherboard is Gigabyte GA-73PVM-S2H, with nforce 630i and realtek alc889a. aplay lists the hdmi device: > aplay -L default:CARD=NVidia HDA NVidia, ALC885 Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) but, > aplay -D hdmi test.wav aplay: main:583: audio open error: No such file or directory fails when trying to open /dev/snd/pcmC0D3p only the following devices exists: /dev/snd/seq /dev/snd/controlC0 /dev/snd/pcmC0D0c /dev/snd/pcmC0D0p /dev/snd/pcmC0D1c /dev/snd/pcmC0D1p /dev/snd/pcmC0D2c /dev/snd/timer I am using alsa-lib and alsa-utils 1.0.17 . Does anyone have an idea? Thanks.

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[alsa-devel] snd_pcsp locking mess
by Stas Sergeev 21 Oct '08

21 Oct '08

bugme-daemon(a)bugzilla.kernel.org wrote: > ------- Comment #17 from tiwai(a)suse.de 2008-05-18 10:22 ------- > Anyway, what I meant is that the part touching I/O ports could be a faster path > than softirq. And the hrtimer code can use a bit simpler locks. The hrtimer > callback needs just one thing about DMA buffer - the buffer is allocated and > the address doesn't change during the callback. So, we'd need the lock for the > hr callback and for the PCM prepare, hwparams and free callbacks, i.e. to > assure that the hrtimer callback is finished when these PCM callbacks may > change the DMA buffer. Thus this lock doesn't have to be PCM substream lock. I wonder if it would be better to introduce such a lock in an alsa core rather than in a driver. The same way as snd_pcm_stream_lock(). > The current lock mess may come from the PCM trigger (STOP) called from > snd_pcm_period_elapsed(). So, putting snd_pcm_period_elapsed() out of hrtimer > lock is anyway necessary. A tasklet for snd_pcm_period_elapsed() is needed for > this reason, too. Wouldn't it be better to just make snd_pcm_period_elapsed() to not take the stream lock? There is a requirement right now to drop the stream lock before calling this. I still beleive this is the root of all the headache. If you need to drop it, then you either accept the race or need a second lock, and the troubles ensue.

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[alsa-devel] [PATCH1/2] hda: add access to jack functions
by Matthew Ranostay 21 Oct '08

21 Oct '08

Removed 'static' declaration from get_jack_location() to be accessed via the codec patches. Also added jack location and type reporting functions for use in the jack abstraction layer implementations. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4a08c31..10f26e0 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -383,6 +383,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, /* * get widget capabilities */ +const char *get_jack_connectivity(u32 cfg); +const char *get_jack_type(u32 cfg); +const char *get_jack_location(u32 cfg); static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) { if (nid < codec->start_nid || diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 64ab19f..b77c1b1 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -145,7 +145,7 @@ static void print_pcm_caps(struct snd_info_buffer *buffer, print_pcm_formats(buffer, stream); } -static const char *get_jack_location(u32 cfg) +const char *get_jack_location(u32 cfg) { static char *bases[7] = { "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", @@ -171,6 +171,26 @@ static const char *get_jack_location(u32 cfg) return "UNKNOWN"; } +const char *get_jack_connectivity(u32 cfg) +{ + static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; + + return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3]; +} + +const char *get_jack_type(u32 cfg) +{ + static char *jack_types[16] = { + "Line Out", "Speaker", "HP Out", "CD", + "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", + "Line In", "Aux", "Mic", "Telephony", + "SPDIF In", "Digitial In", "Reserved", "Other" + }; + + return jack_types[(cfg & AC_DEFCFG_DEVICE) + >> AC_DEFCFG_DEVICE_SHIFT]; +} + static const char *get_jack_connection(u32 cfg) { static char *names[16] = { @@ -206,13 +226,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer, int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; - static char *jack_types[16] = { - "Line Out", "Speaker", "HP Out", "CD", - "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", - "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" - }; - static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); @@ -274,8 +287,8 @@ static void print_pin_caps(struct snd_info_buffer *buffer, caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], - jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT], - jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3], + get_jack_type(caps), + get_jack_connectivity(caps), get_jack_location(caps)); snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps),

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Re: [alsa-devel] Audio on Linux is a mess
by Marek Vasut 20 Oct '08

20 Oct '08

You got what you paid for. Maybe if you change your attitude and supply some information, someone might consider looking in your issue, othervise it would be a problem since I forgot my crystal ball back at home. On Monday 20 of October 2008 16:41:45 Fred . wrote: > Yeah, fix the Linux audio so that if an application behaves wrong, or > does something stupid, then after I close the application, the system > should be silent. > It shouldn't continue make sound even after I quit the application. > And PulseAudio shouldn't crash because one application did something > even if it was stupid. > > On Mon, Oct 20, 2008 at 8:35 AM, Marek Vasut <marek.vasut(a)gmail.com> wrote: > > On Friday 17 of October 2008 15:24:20 Fred . wrote: > >> ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: > >> Timeout ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to > >> connect: Timeout AL lib: alsa.c:344: Could not open playback device > >> 'default': Connection refused AL lib: oss.c:179: Could not open > >> /dev/dsp: Device or resource busy File > >> "/var/lib/python-support/python2.5/pyglet/media/drivers/openal/__init__. > >>py" , line 340, in driver_init_device = alc.alcOpenDevice(device_name) > >> > >> Audio on Linux is a mess. > >> I launch an Python script that uses audio then it breaks, and when the > >> Python script is closed, I still hear the audio skipping and repeating > >> like, > >> "kakakakakakaka" in infinite loop. > >> And PulseAudio seems to be frozen. > >> Then I kill PulseAudio and finally the audio noise stops. > >> > >> Oh, and I heard OpenAL downsamples stereo to mono on Linux due to a bug. > >> > >> Whole audio on Linux is just a big mess. > > > > Please behave yourself. > > Is there any way we can help you (which I doubt, but ...)? > > > >> _______________________________________________ > >> Alsa-devel mailing list > >> Alsa-devel(a)alsa-project.org > >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] Audio on Linux is a mess
by Fred . 20 Oct '08

20 Oct '08

ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Timeout ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Timeout AL lib: alsa.c:344: Could not open playback device 'default': Connection refused AL lib: oss.c:179: Could not open /dev/dsp: Device or resource busy File "/var/lib/python-support/python2.5/pyglet/media/drivers/openal/__init__.py", line 340, in driver_init_device = alc.alcOpenDevice(device_name) Audio on Linux is a mess. I launch an Python script that uses audio then it breaks, and when the Python script is closed, I still hear the audio skipping and repeating like, "kakakakakakaka" in infinite loop. And PulseAudio seems to be frozen. Then I kill PulseAudio and finally the audio noise stops. Oh, and I heard OpenAL downsamples stereo to mono on Linux due to a bug. Whole audio on Linux is just a big mess.

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[alsa-devel] No pcm device?
by John Kacur 20 Oct '08

20 Oct '08

I'm not sure if the hardware I have makes this problem simple for me, or difficult, because it seems to differ from other ASoC code. Basically, I have a i2s as transport, connected to max5556 driving the output. (and a few other chips related to amplification and muting) There is however no sound card, thus no PCM device. Now the question is how to proceed? In other words what kind of minimal PCM layer do I need if any? Thanks

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[alsa-devel] [PATCH] snd-sbxfi: implement Master volume control
by William Pitcock 20 Oct '08

20 Oct '08

Unfortunately, the X-Fi does not have a hardware master volume control, so we have to create a virtual master and bind all playback controls to it. This patch depends on the patch that renames Master to PCM. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index 3e99a97..d8b32aa 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -1685,15 +1685,45 @@ static struct snd_kcontrol_new mixers[] __devinitdata = { }, }; +/* list of controls to attach to the master volume control. */ +static const char *slave_vols[] __devinitdata = { + "PCM Playback Volume", +}; + static int __devinit sbxfi_create_mixer(struct sbxfi *chip) { int i, err; + struct snd_kcontrol *vmaster; for (i = 0; i < ARRAY_SIZE(mixers); i++) { err = snd_ctl_add(chip->card, snd_ctl_new1(&mixers[i], chip)); if (err < 0) return err; } + + /* + * Unfortunately, it seems that there is no master control, unlike + * on the emu10k1 series, so we create a virtual master, and bind + * our playback controls to it. + */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", NULL); + if (vmaster == NULL) + return -ENOMEM; + + /* attach the master to the card, and attach all slaves to it. */ + snd_ctl_add(chip->card, vmaster); + for (i = 0; i < ARRAY_SIZE(slave_vols); i++) { + struct snd_kcontrol *slave; + struct snd_ctl_elem_id sid; + + memset(&sid, 0, sizeof(sid)); + strlcpy(sid.name, slave_vols[i], 44); /* XXX this should be a constant defined somewhere */ + sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + if ((slave = snd_ctl_find_id(chip->card, &sid)) != NULL) + snd_ctl_add_slave(vmaster, slave); + } + return 0; } -- 1.5.5.4

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[alsa-devel] Microphone array using ALSA
by Ola Benderius 20 Oct '08

20 Oct '08

Hi! http://en.wikipedia.org/wiki/Microphone_array Would it be possible to use several USB soundcards in GNU/Linux using ALSA to build a microphone array? I have searched the Internet but haven't found any straight answers. What I have read is that it might be hard to sync the captured streams even if it would be possible. So, three questions. 1. Is it theoretically possible? 2. Can it be done using several USB soundcards? 3. Do you know of any code or project in this area as it is? Think this might be a necessary advancement for the future, the applications are many. Even Windows Vista has support for it. Other thoughts, comments? Regards, Ola

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[alsa-devel] ALSA SOC in the ARM linux kernel - pxa2xx-pcm oops on invalid dma_ch.
by Oliver Ford 20 Oct '08

20 Oct '08

Hi, I had been working on a kernel for the iPAQ 21x series (PXA310 based with a WM9713) which I had originally (June 08) based on arm.linux.org.uk's git tree with wolfsonmicro.com/linux-2.6-asoc/asoc-v2-dev merged in. Recently (last week) I tried to pull the latest changes from arm.linux.org.uk and got too many conflicts in the ALSA SOC code, the mainline kernel code seems to have been cleaned up a lot compared to asoc-v2-dev. Due to this I'm now working from the ALSA SOC code as it comes in arm.linux.org.uk. This mostly works but I'm having trouble with oopses due to invalid DMA channels in pxa2xx-pcm.c. Is this the right place to ask about this, if so I'll post the details here (which I already posted on the arm-linux mailing list). Is there a branch on wolfsonmicro.com that would be closer to what is now in the main kernel? Thanks, Oliver Ford

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[alsa-devel] davinci_request_dma()
by John Kacur 20 Oct '08

20 Oct '08

I'm looking for a little bit more help reading the SOC code. In particular, I'm reading davinci-pcm.c, and function davinci_pcm_dma_request, which makes two calls to davinci_request_dma. However, when I grep for davinci_request_dma, the only places I can find it are in davinci_pcm_dma_request itself! [jkacur@mydcore linux-2.6.26]$ grep -rn davinci_request_dma * sound/soc/davinci/davinci-pcm.c:143: ret = davinci_request_dma(prtd->params->channel, prtd->params->name, sound/soc/davinci/davinci-pcm.c:150: ret = davinci_request_dma(PARAM_ANY, "Link", Either I'm missing something, or this code wouldn't even compile. I would appreciate any help in understanding this. Thank you.

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