October 2008 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] 20K1 PCI bars, and mode switching info.
by James Courtier-Dutton 20 Oct '08

20 Oct '08

Vedran Miletić wrote: > Takashi added UAA mode to hda-intel driver with patch_creative.c, > IIRC. Why would this be necessary? > In time xfi mode will have far more features than the UAA mode. E.g. hardware mixing of over 64 channels at the same time. The UAA mode has no hardware mixing and handles only a few channels.

2 1

Re: [alsa-devel] Backported sbxfi driver (UNTESTED!)
by Takashi Iwai 19 Oct '08

19 Oct '08

At Fri, 17 Oct 2008 14:16:38 +0400, The Source wrote: > > Takashi Iwai пишет: > > At Fri, 17 Oct 2008 14:01:55 +0400, > > The Source wrote: > > > >> Takashi Iwai пишет: > >> > >>> At Fri, 17 Oct 2008 13:58:20 +0400, > >>> The Source wrote: > >>> > >>> > >>>> Takashi Iwai пишет: > >>>> > >>>> > >>>>> At Fri, 17 Oct 2008 11:57:08 +0400, > >>>>> The Source wrote: > >>>>> > >>>>> > >>>>> > >>>>>> Takashi Iwai пишет: > >>>>>> > >>>>>> > >>>>>> > >>>>>>> At Thu, 16 Oct 2008 22:18:07 +0400, > >>>>>>> The Source wrote: > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>>>>> Ok. OpenAL with alsa also seem to cause problems. > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>> In both cases, check the period_size and buffer_size values (shown in > >>>>>>>>> the kernel message, or /proc/asound/card0/pcm0p/sub0/hw_params). > >>>>>>>>> And, try to aplay with these parameters, whether you get the similar > >>>>>>>>> problem. > >>>>>>>>> > >>>>>>>>> % aplay -v --period-size=xxx --buffer-size=yyy foo.wav > >>>>>>>>> > >>>>>>>>> > >>>>>>>>> Takashi > >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > >>>>>>>> I'm sorry, but any attemp to play file with ossplay results in complete > >>>>>>>> system hang with error: > >>>>>>>> unable to handle NULL ponter dereference at address > >>>>>>>> 0000000000000008.....(hang, no more output). > >>>>>>>> I tried many wav formats. So I can't get error log or period and buffer > >>>>>>>> sizes, sorry. > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>> Can anyone confirm to reproduce Oops with OSS apps (ossplay)? > >>>>>>> > >>>>>>> I'm wondering whether this has anything to do with the capture. > >>>>>>> Can you record the sound, and change the capture mixer element properly? > >>>>>>> > >>>>>>> > >>>>>>> thanks, > >>>>>>> > >>>>>>> Takashi > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> I checked mplayer. It uses period size 1024 instead of 4096 and 16384 > >>>>>> buffer size (default). Sound is choppy (sound pauses is more frequent > >>>>>> when rate is lower). > >>>>>> However an attempt to play the same file with the same period and buffer > >>>>>> sizes with aplay results in complete system hang. > >>>>>> > >>>>>> > >>>>>> > >>>>> OK, that looks like a problem. Looks like the timer resolution can be > >>>>> short like that, or something racy in the timer handling. > >>>>> > >>>>> Can you check whether this happens with XXX_SYSTEM_TIMER, too? > >>>>> > >>>>> Or, does the patch below avoid the problem, at least? > >>>>> > >>>>> > >>>>> thanks, > >>>>> > >>>>> Takashi > >>>>> > >>>>> diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c > >>>>> index 26a6cd3..5ceb228 100644 > >>>>> --- a/sound/pci/sbxfi/sbxfi.c > >>>>> +++ b/sound/pci/sbxfi/sbxfi.c > >>>>> @@ -277,6 +277,7 @@ static void sbxfi_rearm_timer(struct sbxfi *chip, int ticks) > >>>>> #else > >>>>> > >>>>> #define MAX_TICKS ((1 << 13) - 1) > >>>>> +#define MIN_TICKS 1000 /* FIXME: really so? */ > >>>>> > >>>>> static void sbxfi_init_timer(struct sbxfi *chip) > >>>>> { > >>>>> @@ -287,6 +288,8 @@ static void sbxfi_set_timer(struct sbxfi *chip, int ticks) > >>>>> LOG(2, "SET TIMER TICKS = %d\n", ticks); > >>>>> if (ticks > MAX_TICKS) > >>>>> ticks = MAX_TICKS; > >>>>> + else if (ticks < MIN_TICKS) > >>>>> + ticks = MIN_TICKS; > >>>>> sbxfi_write(chip, TIMR, ticks | TIMR_IE | TIMR_IP); > >>>>> } > >>>>> static void sbxfi_stop_timer(struct sbxfi *chip) > >>>>> > >>>>> > >>>>> > >>>>> > >>>> After patch: > >>>> > >>>> Without system timer: > >>>> > >>>> aplay --period-size=1024 96000_16_Stereo.wav > >>>> Plays fine > >>>> > >>>> aplay --period-size=1024 22050_16_Mono.wav > >>>> BUG: unable to handle kernel NULL pointer dereference at 0000000000000008 > >>>> Hang > >>>> > >>>> With system timer: > >>>> > >>>> aplay --period-size=1024 96000_16_Stereo.wav > >>>> Superglitch. Each sample is played many times before advancing to next one. > >>>> > >>>> aplay --period-size=1024 22050_16_Mono.wav > >>>> Instant reboot. > >>>> > >>>> > >>> Just to be sure: you don't enable XXX_CONT_RATE, right? > >>> It's known to be buggy, so disabled as default now. > >>> > >>> > >>> Takashi > >>> > >>> > >>> > >> It is disabled for me too. > >> > > > > And the patch didn't help? > > > > > > Takashi > > > > > Only 96000Hz Stereo without system timer plays fine after patch. Please elaborate? You mean 22.5k still doesn't work with the patch? Does 22.5kHz ever work with other parameters? It'd be really helpful if you get the full Oops log... The patch affects only the emu20k1 timer. The system timer stuff isn't touched by it. The reboot implies that it's unlikely the timer but the driver wrote something wrong (unless you have the set up that the kernel traps and automatically reboot). But, it's nothing but a wild guess at this point. Anyway, I updated the driver code a bit again. Please check the latest one. thanks, Takashi

4 8

[alsa-devel] Digidesign Mbox 2 Mini
by Mark Hills 19 Oct '08

19 Oct '08

I have been trying to use a Digidesign Mbox 2 Mini (USB) without success, but with some positive signs. The device is detected. From the software side, playback appears to work -- both aplay and mplayer push audio to somewhere (no crashes, and at the correct rate). But there is no sound from any of the outputs. Recording is not so good. It does not error or crash, but arecord hangs and creates a 44 byte file containing the WAV header. In a test program, snd_pcm_readi() does not return and strace waits on poll(). I found two threads on alsa-devel referring to the Mbox interfaces (a different model to the one I am using though; different ID): http://mailman.alsa-project.org/pipermail/alsa-devel/2007-November/004394.h… http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg12273.html In the second one, which is from 2004, Clemens says: > Strange. But having two endpoints in one interface is not what the > driver was written for. I think part of the driver needs to be > rewritten before this device can work. Now this is a while ago, with a different interface. But the Mbox 2 mini also has two endpoints in one interface (see attached lsusb output), and I seem to be getting considerably more out of the interface than the original poster. So is it possible that the driver is now successfully getting audio to the playback endpoint only, and just needs some kind of mixer device control (or similar) for playback to work? Or is the 'success' with aplay/mplayer misleading -- how far off working is this? Thanks, -- Mark

1 0

[alsa-devel] [PATCH] snd-sbxfi: rename erroneous Master mixer control to "PCM Playback Volume".
by William Pitcock 19 Oct '08

19 Oct '08

The OSS4-provided master mixer control is actually relative to EMU20K1 port volume settings. So, it should be named PCM. The Master mixer control will be shortly readded as a control which sets the master volume. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 28 ++++++++++++++-------------- 1 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index c006278..3e99a97 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -155,7 +155,7 @@ struct sbxfi { int timer_reprogram; unsigned int base_rate; - int master_vol[2]; + int pcm_vol[2]; int capture_vol[2]; int capsrc; int micboost; @@ -1103,7 +1103,7 @@ static void sbxfi_setup_play_mixer(struct sbxfi *chip, int i, vol, audio_ch; for (i = 0; i < 2; i++) { - vol = port->vol[i] + chip->master_vol[i] - MAX_VOLUME; + vol = port->vol[i] + chip->pcm_vol[i] - MAX_VOLUME; if (vol < 0) vol = 0; vol = amop_vol_table[vol]; @@ -1124,9 +1124,9 @@ static int sbxfi_update_play_mixer(struct sbxfi *chip, int *vol) struct sbxfi_port *port; for (i = 0; i < 2; i++) { - if (vol[i] == chip->master_vol[i]) + if (vol[i] == chip->pcm_vol[i]) continue; - chip->master_vol[i] = vol[i]; + chip->pcm_vol[i] = vol[i]; spin_lock_irq(&chip->port_lock); list_for_each_entry(port, &chip->port_list, list) { if (port->substream->stream != @@ -1516,7 +1516,7 @@ static int __devinit sbxfi_create_pcm(struct sbxfi *chip) /* * Mixer interface */ -static int sbxfi_master_vol_info(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1526,16 +1526,16 @@ static int sbxfi_master_vol_info(struct snd_kcontrol *kcontrol, return 0; } -static int sbxfi_master_vol_get(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sbxfi *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = chip->master_vol[0]; - ucontrol->value.integer.value[1] = chip->master_vol[1]; + ucontrol->value.integer.value[0] = chip->pcm_vol[0]; + ucontrol->value.integer.value[1] = chip->pcm_vol[1]; return 0; } -static int sbxfi_master_vol_put(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sbxfi *chip = snd_kcontrol_chip(kcontrol); @@ -1543,7 +1543,7 @@ static int sbxfi_master_vol_put(struct snd_kcontrol *kcontrol, vol[0] = ucontrol->value.integer.value[0]; vol[1] = ucontrol->value.integer.value[1]; - if (!memcmp(vol, chip->master_vol, sizeof(vol))) + if (!memcmp(vol, chip->pcm_vol, sizeof(vol))) return 0; if (vol[0] < 0) @@ -1658,13 +1658,13 @@ static int sbxfi_micboost_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new mixers[] __devinitdata = { { - .name = "Master Playback Volume", + .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE /* | SNDRV_CTL_ELEM_ACCESS_TLV_READ*/), - .info = sbxfi_master_vol_info, - .get = sbxfi_master_vol_get, - .put = sbxfi_master_vol_put, + .info = sbxfi_pcm_vol_info, + .get = sbxfi_pcm_vol_get, + .put = sbxfi_pcm_vol_put, /*.tlv = { .p = db_master_vol }*/ }, { -- 1.5.5.4

3 4

[alsa-devel] snd-xfi programming, some guidelines.
by James Courtier-Dutton 19 Oct '08

19 Oct '08

Hi, I have not had time to review the current work on the snd-xfi, but I have provided input if I think it would be useful. I have one more piece of information that might help. The xfi has two ways to talk to all the registers. 1) ioports using ioport pointer offset write method as familiar with older emu10k1 cards. 2) iomem using simple memory writes. The same registers on the xfi card can be accessed via both methods. There is some guidance as to when to use each. I.e. Do not use iomem for everything. There are some registers that take longer to write to than others. Unfortunately, the datasheet does not tell me which these slow are, so trial and error might be required. Now, if one uses the iomem to write to these registers that take a long time to update, it ties up the entire PCI bus for the entire duration, so this is not a good idea. So, the recommendation is that one should use ioports to access these slow registers as this will not block the PCI bus. I suggest putting a timer round each ioport and iomem write to find the slow ones. Kind Regards James

1 0

[alsa-devel] [PATCH] snd-sbxfi: use SBXXXX model numbers instead of user-unfriendly names like UAA and ORIG.
by William Pitcock 19 Oct '08

19 Oct '08

This is just something that bothers me. There's no reason to accept or reject it, other than that it might be more informative about what hardware revision we are dealing with. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index 30ad25b..c006278 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -1702,10 +1702,10 @@ static int __devinit sbxfi_create_mixer(struct sbxfi *chip) */ static char *typenames[SBXFI_HW_TYPES] = { - [SBXFI_HW_ORIG] = "ORIG", - [SBXFI_HW_055x] = "055x", - [SBXFI_HW_073x] = "073x", - [SBXFI_HW_UAA] = "UAA" + [SBXFI_HW_ORIG] = "SB0760", + [SBXFI_HW_055x] = "SB0550", + [SBXFI_HW_073x] = "SB0730", + [SBXFI_HW_UAA] = "SB0886" }; #ifdef CONFIG_PROC_FS -- 1.5.5.4

2 1

[alsa-devel] [PATCH] snd-sbxfi: set up the DAC at card initialization, regardless of duplex mode
by William Pitcock 19 Oct '08

19 Oct '08

We should bring up the DAC at card initialization even in half-duplex mode, otherwise we have to deal with unfortunate glitches in the hardware. Constantly resetting the DAC can cause serious state desyncs, resulting in the driver eventually not working until a reboot occurs. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 8 +------- 1 files changed, 1 insertions(+), 7 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index d64d462..30ad25b 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -62,7 +62,7 @@ MODULE_PARM_DESC(debug, "Debug flag: 0 = disable"); /* continuous rate support */ #undef XXX_CONT_RATE -/* full-duplex support */ +/* full duplex support */ #undef XXX_FULL_DUPLEX /* multi playback - not implemented */ @@ -1364,10 +1364,6 @@ static int sbxfi_playback_prepare(struct snd_pcm_substream *substream) port->frag_count = runtime->period_size; port->need_update = 0; -#ifndef XXX_FULL_DUPLEX - sbxfi_init_dac(chip); - sbxfi_setup_i2s(chip); -#endif sbxfi_setup_play_pitch(chip, port); sbxfi_setup_play_mixer(chip, port); sbxfi_setup_play_mapper(chip, port); @@ -1924,10 +1920,8 @@ static int sbxfi_init_hw(struct sbxfi *chip) sbxfi_daoimap_clear(chip); -#ifdef XXX_FULL_DUPLEX sbxfi_init_dac(chip); sbxfi_setup_i2s(chip); -#endif return 0; } -- 1.5.5.4

2 1

[alsa-devel] [PATCH] snd-sbxfi: implement support for 192khz playback/capture.
by William Pitcock 19 Oct '08

19 Oct '08

Implements support for 192khz initialization for playback and capture. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index 4c65125..d64d462 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -514,7 +514,7 @@ static void sbxfi_write_amop(struct sbxfi *chip, static void sbxfi_playback_start(struct sbxfi *chip, struct sbxfi_port *port) { unsigned int start, loop; - unsigned int ctrl, ratec; + unsigned int ctrl, ratec = 0; struct snd_pcm_runtime *runtime; if (port->tlb_index < 0) @@ -523,12 +523,19 @@ static void sbxfi_playback_start(struct sbxfi *chip, struct sbxfi_port *port) start = port->tlb_index * SBXFI_PAGE_SIZE; loop = start + frames_to_bytes(runtime, runtime->buffer_size); + /* + * ratec is the rate selection mode, 0x4c = 48000hz, 0x5c = 96000hz, + * 0x6c = 192000hz. + */ switch (runtime->rate) { case 48000: - ratec = 0x4c; /* XXX what meaning? */ + ratec = 0x4c; break; case 96000: - ratec = 0x5c; /* XXX what meaning? */ + ratec = 0x5c; + break; + case 192000: + ratec = 0x6c; break; } ctrl = ratec; @@ -559,7 +566,7 @@ static void sbxfi_playback_stop(struct sbxfi *chip, struct sbxfi_port *port) static void sbxfi_capture_start(struct sbxfi *chip, struct sbxfi_port *port) { unsigned int start, loop; - unsigned int ctrl, ratec; + unsigned int ctrl, ratec = 0; struct snd_pcm_runtime *runtime; if (port->tlb_index < 0) @@ -575,6 +582,9 @@ static void sbxfi_capture_start(struct sbxfi *chip, struct sbxfi_port *port) case 96000: ratec = 0x5d | SRCCTL_RUN_STATE; break; + case 192000: + ratec = 0x6d | SRCCTL_RUN_STATE; + break; } ctrl = ratec; -- 1.5.5.4

3 3

[alsa-devel] [PATCH] snd-sbxfi: rename erroneous Master mixer control to "PCM Playback Volume".
by William Pitcock 19 Oct '08

19 Oct '08

The OSS4-provided master mixer control is actually relative to EMU20K1 port volume settings. So, it should be named PCM. The Master mixer control will be shortly readded as a control which sets the master volume. Signed-off-by: William Pitcock <nenolod(a)petrie.sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 28 ++++++++++++++-------------- 1 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index c006278..3e99a97 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -155,7 +155,7 @@ struct sbxfi { int timer_reprogram; unsigned int base_rate; - int master_vol[2]; + int pcm_vol[2]; int capture_vol[2]; int capsrc; int micboost; @@ -1103,7 +1103,7 @@ static void sbxfi_setup_play_mixer(struct sbxfi *chip, int i, vol, audio_ch; for (i = 0; i < 2; i++) { - vol = port->vol[i] + chip->master_vol[i] - MAX_VOLUME; + vol = port->vol[i] + chip->pcm_vol[i] - MAX_VOLUME; if (vol < 0) vol = 0; vol = amop_vol_table[vol]; @@ -1124,9 +1124,9 @@ static int sbxfi_update_play_mixer(struct sbxfi *chip, int *vol) struct sbxfi_port *port; for (i = 0; i < 2; i++) { - if (vol[i] == chip->master_vol[i]) + if (vol[i] == chip->pcm_vol[i]) continue; - chip->master_vol[i] = vol[i]; + chip->pcm_vol[i] = vol[i]; spin_lock_irq(&chip->port_lock); list_for_each_entry(port, &chip->port_list, list) { if (port->substream->stream != @@ -1516,7 +1516,7 @@ static int __devinit sbxfi_create_pcm(struct sbxfi *chip) /* * Mixer interface */ -static int sbxfi_master_vol_info(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1526,16 +1526,16 @@ static int sbxfi_master_vol_info(struct snd_kcontrol *kcontrol, return 0; } -static int sbxfi_master_vol_get(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sbxfi *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = chip->master_vol[0]; - ucontrol->value.integer.value[1] = chip->master_vol[1]; + ucontrol->value.integer.value[0] = chip->pcm_vol[0]; + ucontrol->value.integer.value[1] = chip->pcm_vol[1]; return 0; } -static int sbxfi_master_vol_put(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sbxfi *chip = snd_kcontrol_chip(kcontrol); @@ -1543,7 +1543,7 @@ static int sbxfi_master_vol_put(struct snd_kcontrol *kcontrol, vol[0] = ucontrol->value.integer.value[0]; vol[1] = ucontrol->value.integer.value[1]; - if (!memcmp(vol, chip->master_vol, sizeof(vol))) + if (!memcmp(vol, chip->pcm_vol, sizeof(vol))) return 0; if (vol[0] < 0) @@ -1658,13 +1658,13 @@ static int sbxfi_micboost_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new mixers[] __devinitdata = { { - .name = "Master Playback Volume", + .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE /* | SNDRV_CTL_ELEM_ACCESS_TLV_READ*/), - .info = sbxfi_master_vol_info, - .get = sbxfi_master_vol_get, - .put = sbxfi_master_vol_put, + .info = sbxfi_pcm_vol_info, + .get = sbxfi_pcm_vol_get, + .put = sbxfi_pcm_vol_put, /*.tlv = { .p = db_master_vol }*/ }, { -- 1.5.5.4

1 0

[alsa-devel] alsa + resampler + jack: possible?
by Markus Grabner 19 Oct '08

19 Oct '08

Hi all! A while ago I started development of an ALSA driver for a device which natively only supports a sampling rate of 39062.5 Hz and 24 bit samples (see http://www.tanzband-scream.at/line6). The ALSA rate plugin does a perfect job for using this device with standard applications (such as xmms) to play audio files at "conventional" sample rates (e.g., 44.1 kHz) with a configuration like this: pcm.resampler { type rate slave { pcm hw:1 format S24_3LE rate 39062 } converter "samplerate" } However, this doesn't work for jack since jack requires a power-of-two period size, and ALSA can't find an integer period size for the raw device such that the target rate is met with sufficient precision. An example: the jack daemon can't be started with period size 512 and sample rate 48000 Hz, but it works for 47961 Hz since 512 * 39062.5 / 48000 = 416.666... ~= 417 bytes 512 / 417 * 39062.5 = 47961.63 ~= 47961 Hz When using this approach together with other audio data from a 48 kHz source, they would drift apart over time, which basically renders the alsa+jack setup useless for this device. I understand that resampling is easier to implement if the sample rate ratio equals the ratio of the (integer) period sizes of client and slave, but at least the underlying libsamplerate has no such restriction. I tried to find the relevant pieces in the ALSA source code (pcm_rate.c), but I'm not sure where these calculations are done and how much effort it would be to support arbitrary sample rate ratios for the resampler. Can this be done in a plugin, or does it require modifications in the ALSA library code base? Could the developers please give a few hints on that? Thanks & kind regards, Markus -- Markus Grabner - Computer Graphics and Vision Graz University of Technology, Inffeldgasse 16a/II, 8010 Graz, Austria Phone: +43/316/873-5041, Fax: +43/316/873-5050 WWW: http://www.icg.tugraz.at/Members/grabner

1 0

[alsa-devel] snd_pcm_sw_params_get_xfer_align deprecated?
by Erik de Castro Lopo 18 Oct '08

18 Oct '08

Hi all, I've got some code that gives me a waring that the above function is deprecated. What should I do: a) Call some other function? If so which one? b) Just drop the call to that function and everything will continue to work as it should? c) Something else? Cheers, Erik -- ----------------------------------------------------------------- Erik de Castro Lopo ----------------------------------------------------------------- "The very definition of hell is having to maintain someone else's Perl code." -- Anonymous

3 2

[alsa-devel] New BIOS, new vendor and device IDs
by Chris Pemberton 17 Oct '08

17 Oct '08

My machine is a Lenovo 3000 N100 laptop (0768 model). I had to recently update the BIOS. Old BIOS: lspci -> device id: [17aa:2066] which is a Realtek ALC861VD (found in patch_realtek.c) New BIOS: lspci -> device id: [8086:27d8] (found in hda_intel.c) In both cases, I use the module option "model=3stack" to get sound working: Laptop speakers work. Built-in mic works. External mic works Headphones work; but automatic speaker muting does not. I tried to debug why I needed this module option at all, several months ago (old BIOS), but I didn't get far and just became complacent with using the module option. If anyone is interested, I'll do what I can to help debug. It would be nice to be able to use Skype with headphones... Output of the alsa info script is at: http://www.alsa-project.org/db/?f=701ca6efa0500080dc78a555813f66c1d42096e6 That is with driver 1.0.17. I'll try to update to the latest snapshot and run the script again. Thanks, Chris

2 6

Re: [alsa-devel] Backported sbxfi driver (UNTESTED!)
by Takashi Iwai 17 Oct '08

17 Oct '08

At Thu, 16 Oct 2008 19:06:58 +0400, The Source wrote: > > Takashi Iwai пишет: > > At Thu, 16 Oct 2008 19:00:16 +0400, > > The Source wrote: > > > >> Takashi Iwai пишет: > >> > >>> At Thu, 16 Oct 2008 18:41:08 +0400, > >>> The Source wrote: > >>> > >>> > >>>> Takashi Iwai пишет: > >>>> > >>>> > >>>>> At Thu, 16 Oct 2008 18:15:45 +0400, > >>>>> The Source wrote: > >>>>> > >>>>> > >>>>> > >>>>>> Takashi Iwai пишет: > >>>>>> > >>>>>> > >>>>>> > >>>>>>> At Thu, 16 Oct 2008 17:36:04 +0400, > >>>>>>> The Source wrote: > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>>>>> And, which X-Fi model do you have? > >>>>>>>>>> Please show the lspci -nv output, too. > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>> I've got the X-Fi Elite Pro. > >>>>>>>>> That's The one with the external In/Out box. > >>>>>>>>> > >>>>>>>>> Speaking of which, the headphone jack on it does not output a signal > >>>>>>>>> yet, the signal only goes to line out. > >>>>>>>>> > >>>>>>>>> There's some relais on the card that seem to switch these, they click > >>>>>>>>> multiple times with the windows driver and not all all with yours, I > >>>>>>>>> think that's the reason :) > >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > >>>>>>>> Original OSS driver doesn't output to external block also, so it > >>>>>>>> wouldn't be easy to make this support I think. > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>> The values for port->conv[0] and [1] values in sbxfi_playback_open() > >>>>>>> might play some role. It's I2SA_L and I2SA_R, alias DAI_CH_I2SAL > >>>>>>> and DAI_CH_I2SAR, as default. You can try other values, such as, > >>>>>>> DAI_CH_I2SBL, DAI_CH_I2SA1L, and so on. > >>>>>>> > >>>>>>> > >>>>>>> Takashi > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> Latest snapshot has a bug: > >>>>>> make[3]: *** No rule to make target > >>>>>> `/mnt/e/temp/alsa-driver-unstable/acore/jack.o', needed by > >>>>>> `/mnt/e/temp/alsa-driver-unstable/acore/snd.o'. Stop. > >>>>>> > >>>>>> > >>>>>> > >>>>> Already fixed. > >>>>> > >>>>> > >>>>> Takashi > >>>>> > >>>>> > >>>>> > >>>>> > >>>> In file included from /mnt/e/temp/alsa-driver-unstable/acore/jack.c:3: > >>>> /mnt/e/temp/alsa-driver-unstable/acore/../alsa-kernel/core/jack.c: In > >>>> function ‘snd_jack_new’: > >>>> /mnt/e/temp/alsa-driver-unstable/acore/../alsa-kernel/core/jack.c:107: > >>>> error: ‘SW_MICROPHONE_INSERT’ undeclared (first use in this function) > >>>> /mnt/e/temp/alsa-driver-unstable/acore/../alsa-kernel/core/jack.c:107: > >>>> error: (Each undeclared identifier is reported only once > >>>> /mnt/e/temp/alsa-driver-unstable/acore/../alsa-kernel/core/jack.c:107: > >>>> error: for each function it appears in.) > >>>> /mnt/e/temp/alsa-driver-unstable/acore/../alsa-kernel/core/jack.c: In > >>>> function ‘snd_jack_report’: > >>>> /mnt/e/temp/alsa-driver-unstable/acore/../alsa-kernel/core/jack.c:157: > >>>> error: ‘SW_MICROPHONE_INSERT’ undeclared (first use in this function) > >>>> > >>>> > >>> Hmm, it seems broken for older kernels right now. > >>> The easy workaround is to pass --with-cards=sbxfi to configure. > >>> > >>> Anyway, I'll fix it now. > >>> > >>> thanks, > >>> > >>> Takashi > >>> > >>> > >>> > >> --Hey, man, this is cool! Plays just fine (volume ok, speed ok, no > >> --glitches) with 96000, 48000, 44100, 22050, 16/24 bit, Stereo and --Mono!! > >> --I didn't change anything in the source code, so I don't use system > >> --timer. Yes!! > >> > >> However oss (alsa-emulated) is unstable. I'll test more. > >> > > > > Could be due to 96kHz base-rate? Try base_rate=48000. > > If you get still problems, please show the kernel logs with debug=2. > > > > BTW, the jack.c compile error should have been fixed now (hopefully). > > Let me know if you still have the build errors. > > > > > > thanks, > > > > Takashi > > > > > Ok. OpenAL with alsa also seem to cause problems. In both cases, check the period_size and buffer_size values (shown in the kernel message, or /proc/asound/card0/pcm0p/sub0/hw_params). And, try to aplay with these parameters, whether you get the similar problem. % aplay -v --period-size=xxx --buffer-size=yyy foo.wav Takashi

5 22

Re: [alsa-devel] [PATCH] Fix debugfs_create_dir's error checking method for sound/soc/
by Mark Brown 17 Oct '08

17 Oct '08

On Fri, Oct 17, 2008 at 09:04:55PM +0800, Zhaolei wrote: > debugfs_create_dir() returns NULL if an error occurs, returns -ENODEV > when debugfs is not enabled in the kernel. ... > asoc_debugfs = debugfs_create_dir("asoc", NULL); > - if (!IS_ERR(asoc_debugfs)) > + if (!IS_ERR(asoc_debugfs) && asoc_debugfs) > debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs, > &pop_time); Hrmpf. This looks like something that should be fixed in debugfs - using both error reporting strategies is rather unhelpful and we're actually loosing information in the case where debugfs is built. I'll send a patch and see what people thing. Still in the meantime, Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

2 3

[alsa-devel] [PATCH] hda: add SW_LINEOUT_INSERT support
by Matthew Ranostay 17 Oct '08

17 Oct '08

Add support for detecting line out pin insertion and reporting back to userspace with the jack abstraction layer. Line outs are reported with the macro defined SW_LINEOUT_INSERT code. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/include/linux/input.h b/include/linux/input.h index a5802c9..7323d2f 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -644,6 +644,7 @@ struct input_absinfo { #define SW_RADIO SW_RFKILL_ALL /* deprecated */ #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ +#define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index b1b2b8b..7cb25f4 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -35,6 +35,7 @@ enum snd_jack_types { SND_JACK_HEADPHONE = 0x0001, SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, + SND_JACK_LINEOUT = 0x0004, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index bd2d9e6..4978db1 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -102,6 +102,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_HEADPHONE) input_set_capability(jack->input_dev, EV_SW, SW_HEADPHONE_INSERT); + if (type & SND_JACK_LINEOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_LINEOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); @@ -153,6 +156,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_HEADPHONE) input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_LINEOUT) + input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, + status & SND_JACK_LINEOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d106ea5..1b3cba8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -36,6 +36,7 @@ #include "hda_patch.h" #include "hda_beep.h" +#define STAC_LINEOUT_EVENT 0x10 #define STAC_PWR_EVENT 0x20 #define STAC_HP_EVENT 0x30 #define STAC_VREF_EVENT 0x40 @@ -217,7 +218,8 @@ struct sigmatel_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ /* jack detection */ - struct snd_jack *jack; + struct snd_jack *hp_jack; + struct snd_jack *lineout_jack; /* dynamic controls and input_mux */ struct auto_pin_cfg autocfg; @@ -3635,9 +3637,15 @@ static int stac92xx_init(struct hda_codec *codec) /* jack detection */ err = snd_jack_new(codec->bus->card, "Headphone Jack", - SND_JACK_HEADPHONE, &spec->jack); + SND_JACK_HEADPHONE, &spec->hp_jack); if (err < 0) return err; + err = snd_jack_new(codec->bus->card, + "Lineout Jack", + SND_JACK_LINEOUT, &spec->lineout_jack); + if (err < 0) + return err; + /* fake event to set up pins */ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); } else { @@ -3661,6 +3669,16 @@ static int stac92xx_init(struct hda_codec *codec) for (i = 0; i < spec->num_dmics; i++) stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], AC_PINCTL_IN_EN); + for (i = 0; i < cfg->line_outs; i++) { + int def_conf = snd_hda_codec_read(codec, cfg->line_out_pins[i], + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); + + if (def_conf && def_conf != AC_JACK_PORT_FIXED) + continue; + enable_pin_detect(codec, cfg->line_out_pins[i], + STAC_LINEOUT_EVENT); + } for (i = 0; i < spec->num_pwrs; i++) { int event = is_nid_hp_pin(cfg, spec->pwr_nids[i]) ? STAC_HP_EVENT : STAC_PWR_EVENT; @@ -3717,8 +3735,10 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; - if (spec->jack) - snd_device_free(codec->bus->card, spec->jack); + if (spec->hp_jack) + snd_device_free(codec->bus->card, spec->hp_jack); + if (spec->lineout_jack) + snd_device_free(codec->bus->card, spec->lineout_jack); if (spec->bios_pin_configs) kfree(spec->bios_pin_configs); @@ -3804,7 +3824,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } - snd_jack_report(spec->jack, + snd_jack_report(spec->hp_jack, presence ? SND_JACK_HEADPHONE : 0); if (presence) { @@ -3840,6 +3860,28 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); } +static void stac92xx_lineout_detect(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int presence, i; + + for (i = 0; i < cfg->line_outs; i++) { + int def_conf = snd_hda_codec_read(codec, + cfg->line_out_pins[i], 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); + + if (def_conf && def_conf != AC_JACK_PORT_FIXED) + continue; + presence = get_hp_pin_presence(codec, + cfg->line_out_pins[i]); + if (presence) + break; + } + snd_jack_report(spec->lineout_jack, presence ? SND_JACK_LINEOUT : 0); +} + static void stac92xx_pin_sense(struct hda_codec *codec, int idx) { struct sigmatel_spec *spec = codec->spec; @@ -3868,15 +3910,20 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) { struct sigmatel_spec *spec = codec->spec; int idx = res >> 26 & 0x0f; + int event = (res >> 26) & 0x70; - switch ((res >> 26) & 0x70) { + switch (event) { case STAC_HP_EVENT: stac92xx_hp_detect(codec, res); /* fallthru */ case STAC_PWR_EVENT: if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, idx); - break; + /* fallthru */ + case STAC_LINEOUT_EVENT: + if (event == STAC_HP_EVENT) + break; + stac92xx_lineout_detect(codec); case STAC_VREF_EVENT: { int data = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DATA, 0);

3 8

[alsa-devel] SB X-Fi driver status update
by Takashi Iwai 17 Oct '08

17 Oct '08

Hi, now I'm stabilizing the driver a bit. I hope it won't hang up any more (so often), and make you happy testing. ** DOWNLOAD ** The snd-sbxfi driver is included in my sound-unstable git tree, and alsa-driver-unstable snapshot tarball. git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-unstable-2.6.git ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… The git commits can be browsed via http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable-2.6.git;a=su… The snapshot tarball includes HEAD and alsa-kernel/HEAD files. These contain alsa-driver and sound git commit logs. Check these files and compare with the above gitweb page whether you have the latest version. ** BUILD ** If you want to build the sbxfi driver on your 2.6.27 git tree, at best, do the following: - pull sound-2.6.git tree devel branch % git pull \ git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-2.6.git devel - pull sound-unstable-2.6.git tree topic/sbxfi branch % git pull \ git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-unstable-2.6.git\ topic/sbxfi In that way, you can update only the sound and keep the rest clean. (The master branches of sound and sound-unstable git trees are for the latest upsteram, i.e. 2.6.28-pre git tree.) In other versions, it'd be better to use alsa-driver-unstable snapshot tarball to build external modules. ** UPDATED NOTES ** Below are some updates: - As default, the driver works only as a constant sample rate, either 48000 or 96000 (default). The rate can be changed via the module option base_rate. For enabling the continuous rate support (it was on before), define XXX_CONT_RATE in sbxfi.c. - The debug module option is 1 as default, i.e. showing some debug messages but not too often. You can set debug=0 to suppress, or debug=2 or 3 to see more. If you have problems (especially regarding with the sound quality problem), run with debug=2. - Please check whether the default (base_rate=96000) works as is for different rates, formats, channels. Then try base_rate=48000. - If you are brave, try recording, too :) It's no full-duplex, so you can't play and record at the same time. Define XXX_FULL_DUPLEX in sbxfi.c to enable the (possible) full-duplex. - If playback works well, try to define XXX_USE_SG in sbxfi.c. This will make the driver using SG buffer instead of continuous pages. - Also, test SBXFI_DMA_MASK being DMA_32BIT_MASK, especially if you have RAM more than 2GB. - You can try dmix as well. E.g. simply run like % aplay -Dplug:dmix:0 foo.wav thanks, Takashi

4 5

Re: [alsa-devel] Backported sbxfi driver (UNTESTED!)
by Bjoern Olausson 17 Oct '08

17 Oct '08

I found a site with a large range of files: http://bellatrix.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html aplay plays all files it is capable to decode. No crackel, no noise, just fine. glass.wav Signed 16 bit Little Endian, Rate 8000 Hz, Mono M1F1-int16-AFsp.wav Signed 16 bit Little Endian, Rate 8000 Hz, Stereo M1F1-int24-AFsp.wav Signed 24 bit Little Endian in 3bytes, Rate 8000 Hz, Stereo M1F1-int32-AFsp.wav Signed 32 bit Little Endian, Rate 8000 Hz, Stereo M1F1-uint8-AFsp.wav Unsigned 8 bit, Rate 8000 Hz, Stereo stereol.wav Signed 16 bit Little Endian, Rate 22050 Hz, Stereo Utopia_Critical_Stop.wav Signed 16 bit Little Endian, Rate 22050 Hz, Mono Thought when I try to play the first one with mplayer ---> FREEZZZZZ Spiele glass.wav. Audio file-Dateiformat erkannt! ========================================================================== Öffne Audiodecoder: [pcm] Uncompressed PCM audio decoder AUDIO: 8000 Hz, 1 ch, s16le, 128.0 kbit/100.00% (ratio: 16000->16000) Ausgewählter Audiocodec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Amarok with backend ALSA plays all files it can decode without problems. All Signed 16 bit Little Endian, Rate 8000 to 96000 Hz, Stereo do work (with the same oszilating noise described in my previous post with the mono files [I just joned two to get stereo])

2 1

[alsa-devel] HD-audio reconfigure branch merged to master
by Takashi Iwai 16 Oct '08

16 Oct '08

Hi, FYI, I merged the long-pending hda-reconfig topic branch to master and devel branches of sound.git tree for 2.6.29 kernel (not 28). This allows you to reconfigure the HD-audio setup dynamically by triggering via sysfs entries. I'll give more details later... Anyway, let me know if you encounter any build problems or Oops by these changes. thanks, Takashi

1 0

[alsa-devel] Salsa-lib : _snd_pcm_hw_param_test is not defined
by Matthieu CASTET 16 Oct '08

16 Oct '08

Hi, it seems there is a missing implementation of _snd_pcm_hw_param_test function in src/pcm_params.c . It is used in pcm_macros.h by snd_pcm_hw_params_test_xxx functions. Thanks Matthieu

2 6

[alsa-devel] [PATCH 1/2] ASoC: Convert playpaq_wm8510 to bulk route registration API
by Mark Brown 16 Oct '08

16 Oct '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/at32/playpaq_wm8510.c | 12 +++--------- 1 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 98a2d58..b1966e4 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -304,7 +304,7 @@ static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to SPKOUT */ {"Ext Spk", NULL, "SPKOUTP"}, {"Ext Spk", NULL, "SPKOUTN"}, @@ -312,9 +312,6 @@ static const char *intercon[][3] = { {"Mic Bias", NULL, "Int Mic"}, {"MICN", NULL, "Mic Bias"}, {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, }; @@ -334,11 +331,8 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) /* * Setup audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + /* always connected pins */ -- 1.5.6.5

2 4

[alsa-devel] Status of HDMI on Nvidia 10de:0774 device (Asus M3N-H HDMI
by Blacky (alsa-devel) 16 Oct '08

16 Oct '08

Hi all, I am using the above model (MB Asus M3N-H/HDMI) and attempting to use the latest patches for nvhdmi. (By latest, I just checked them from CVS today, and they seem to be very recent.. very bleeding edge) Do they support the 10de:0774 device? The source matches them against 0x10de0004, but I've not been able to identify if it is matching or not. (There are mask values in there which I've not worked out yet). So far, I've failed abysmally at getting HDMI (audio) output on the above mentioned MB. Can anyone tell me if the code is even there to support it? I'm starting to suspect it isn't. I'll follow up with details (aplay -L / -l, asoundconf and other details if there is interest.) Getting the HDMI audio working will be the final piece of the puzzle stopping me from putting the new system into the living room as a working mythtv box. Regards Blacky

3 7

Re: [alsa-devel] Backported sbxfi driver (UNTESTED!)
by Alexey Bazhin 16 Oct '08

16 Oct '08

Hi, I have just compiled driver from git and try to run it. My card is pci: 1102:0005 Multimedia audio controller: Creative Labs SB X-Fi I got following: root@mayonaka:~# modprobe snd-sbxfi root@mayonaka:~# speaker-test -c 2 -r 96000 -t sine speaker-test 1.0.15 Playback device is default Stream parameters are 96000Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 96000Hz (requested 96000Hz) [ 262.522739] XFi DAC reset timeout Period size ran[ 262.564029] XFi DAC reset timeout ge from 32 to 16[ 262.620275] XFi DAC reset timeout 384 Using max b[ 262.676529] XFi DAC reset timeout uffer size 16384[ 262.732784] XFi DAC reset timeout Periods = 4 w[ 262.789038] XFi DAC reset timeout as set period_size = 4096 was set buffer_size = 16384 0 - Front Left [ 327.491997] BUG: soft lockup - CPU#0 stuck for 61s! [swapper:0] [ 327.492001] Kernel panic - not syncing: softlockup: hung tasks and i did hear the sine on the left channel. Hope it helps... -- Alexey Bazhin mailto:baz@irc.msk.ru ICQ 125125882

4 41

[alsa-devel] Backported sbxfi driver (UNTESTED!)
by Bjoern Olausson 16 Oct '08

16 Oct '08

>Hi, > >$SUBJECT is now on my sound-unstable git tree: > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound->unstable-2.6.git >together with other experimental patches. > Just wanted to help out with testing stuff. I compiled your snapshot on my gentoo Linux x86_64 Compilation, installation and loading the drivers went smooth. And, yes, there is sound... Here some infos: blub@freax $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ORIG [SB XFi (ORIG)], device 0: SB-XFi [SB-XFi] Subdevices: 1/1 Subdevice #0: subdevice #0 blub@freax $ lspci -v 01:00.0 Multimedia audio controller: Creative Labs SB X-Fi Subsystem: Creative Labs X-Fi Platinum Flags: bus master, medium devsel, latency 64, IRQ 21 I/O ports at 8c00 [size=32] Memory at eb800000 (64-bit, non-prefetchable) [size=2M] Memory at e4000000 (64-bit, non-prefetchable) [size=64M] Capabilities: [40] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Kernel driver in use: SB-XFi root@freax $ lsmod | grep snd snd_pcm_oss 40640 0 snd_mixer_oss 17224 1 snd_pcm_oss snd_seq_oss 33168 0 snd_seq_midi_event 9864 1 snd_seq_oss snd_seq 54056 4 snd_seq_oss,snd_seq_midi_event snd_seq_device 10076 2 snd_seq_oss,snd_seq snd_hda_intel 505276 0 snd_hwdep 10320 1 snd_hda_intel snd_sbxfi 23132 1 snd_pcm 81328 3 snd_pcm_oss,snd_hda_intel,snd_sbxfi snd_timer 24160 2 snd_seq,snd_pcm snd 65128 12 snd_pcm_oss,snd_mixer_oss,snd_seq_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_hwdep,snd_sbxfi,snd_pcm,snd_timer soundcore 10208 1 snd snd_page_alloc 11664 3 snd_hda_intel,snd_sbxfi,snd_pcm root@freax $ emerge --info Portage 2.1.4.5 (default-linux/amd64/2007.0/desktop, gcc-4.1.2, glibc-2.6.1-r0, 2.6.27-unpatched x86_64) So now the funny part: I was testing a few samplerates: Used a 1 kHz sine, 0 dBFS, 16 bit mono wave file, 96 kHz sample rate and rsampled it with sox: for i in 88200 50400 50000 48000 47250 44100 44056 32000 22050 11025 8000 ; do sox 0_16_96000.wav -r ${i} 0_16_${i}.wav ; done and played them with aplay: for i in 96000 88200 50400 50000 48000 47250 44100 44056 32000 22050 11025 8000 ; do echo "Now playing 0_16_${i}.wav" ; aplay 0_16_${i}.wav ; echo "DONE" ; done 96000 --> O.K. 88200 --> No Sound 50400 --> No Sound 50000 --> No Sound 48000 --> O.K. 47250 --> No Sound 44100 --> No Sound 44056 --> No Sound 32000 --> No Sound 22050 --> SYSTEM FREEZ Playing WAVE '0_16_22050.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono 11025 --> SYSTEM FREEZ Playing WAVE '0_16_11025.wav' : Signed 16 bit Little Endian, Rate 11025 Hz, Mono 8000 --> No Sound Let me know if I can help out testing some more stuff kind regards Bjoern

2 1

[alsa-devel] [PATCH] kernel docs: fix sound/core/ kernel-doc
by Randy Dunlap 16 Oct '08

16 Oct '08

From: Randy Dunlap <randy.dunlap(a)oracle.com> Add kernel-doc function short descriptions to sound/core functions that are missing this short description. Mostly this involves moving some of the function description onto the @funcname line. Also correct a few variable names and fix other kernel-doc notation. Signed-off-by: Randy Dunlap <randy.dunlap(a)oracle.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 48 ++++++++++++++++++++++++------------------------ sound/core/pcm_native.c | 24 ++++++++---------------- 2 files changed, 32 insertions(+), 40 deletions(-) --- lin2627-g5-kdoc.orig/sound/core/pcm_lib.c +++ lin2627-g5-kdoc/sound/core/pcm_lib.c @@ -908,12 +908,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_r EXPORT_SYMBOL(snd_pcm_hw_rule_add); /** - * snd_pcm_hw_constraint_mask + * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 32-bit mask parameter. */ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int32_t mask) @@ -928,12 +928,12 @@ int snd_pcm_hw_constraint_mask(struct sn } /** - * snd_pcm_hw_constraint_mask64 + * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the 64bit bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 64-bit mask parameter. */ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int64_t mask) @@ -949,7 +949,7 @@ int snd_pcm_hw_constraint_mask64(struct } /** - * snd_pcm_hw_constraint_integer + * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the integer constraint * @@ -964,7 +964,7 @@ int snd_pcm_hw_constraint_integer(struct EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); /** - * snd_pcm_hw_constraint_minmax + * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the range * @min: the minimal value @@ -995,7 +995,7 @@ static int snd_pcm_hw_rule_list(struct s /** - * snd_pcm_hw_constraint_list + * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the list constraint @@ -1031,7 +1031,7 @@ static int snd_pcm_hw_rule_ratnums(struc } /** - * snd_pcm_hw_constraint_ratnums + * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratnums constraint @@ -1064,7 +1064,7 @@ static int snd_pcm_hw_rule_ratdens(struc } /** - * snd_pcm_hw_constraint_ratdens + * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratdens constraint @@ -1095,7 +1095,7 @@ static int snd_pcm_hw_rule_msbits(struct } /** - * snd_pcm_hw_constraint_msbits + * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule * @runtime: PCM runtime instance * @cond: condition bits * @width: sample bits width @@ -1123,7 +1123,7 @@ static int snd_pcm_hw_rule_step(struct s } /** - * snd_pcm_hw_constraint_step + * snd_pcm_hw_constraint_step - add a hw constraint step rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the step constraint @@ -1154,7 +1154,7 @@ static int snd_pcm_hw_rule_pow2(struct s } /** - * snd_pcm_hw_constraint_pow2 + * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the power-of-2 constraint @@ -1202,13 +1202,13 @@ void _snd_pcm_hw_params_any(struct snd_p EXPORT_SYMBOL(_snd_pcm_hw_params_any); /** - * snd_pcm_hw_param_value + * snd_pcm_hw_param_value - return @params field @var value * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Return the value for field PAR if it's fixed in configuration space - * defined by PARAMS. Return -EINVAL otherwise + * Return the value for field @var if it's fixed in configuration space + * defined by @params. Return -%EINVAL otherwise. */ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) @@ -1271,13 +1271,13 @@ static int _snd_pcm_hw_param_first(struc /** - * snd_pcm_hw_param_first + * snd_pcm_hw_param_first - refine config space and return minimum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values > minimum. Reduce configuration space accordingly. * Return the minimum. */ @@ -1317,13 +1317,13 @@ static int _snd_pcm_hw_param_last(struct /** - * snd_pcm_hw_param_last + * snd_pcm_hw_param_last - refine config space and return maximum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values < maximum. Reduce configuration space accordingly. * Return the maximum. */ @@ -1345,11 +1345,11 @@ int snd_pcm_hw_param_last(struct snd_pcm EXPORT_SYMBOL(snd_pcm_hw_param_last); /** - * snd_pcm_hw_param_choose + * snd_pcm_hw_param_choose - choose a configuration defined by @params * @pcm: PCM instance * @params: the hw_params instance * - * Choose one configuration from configuration space defined by PARAMS + * Choose one configuration from configuration space defined by @params. * The configuration chosen is that obtained fixing in this order: * first access, first format, first subformat, min channels, * min rate, min period time, max buffer size, min tick time --- lin2627-g5-kdoc.orig/sound/core/pcm_native.c +++ lin2627-g5-kdoc/sound/core/pcm_native.c @@ -875,10 +875,8 @@ static struct action_ops snd_pcm_action_ }; /** - * snd_pcm_start + * snd_pcm_start - start all linked streams * @substream: the PCM substream instance - * - * Start all linked streams. */ int snd_pcm_start(struct snd_pcm_substream *substream) { @@ -926,12 +924,11 @@ static struct action_ops snd_pcm_action_ }; /** - * snd_pcm_stop + * snd_pcm_stop - try to stop all running streams in the substream group * @substream: the PCM substream instance * @state: PCM state after stopping the stream * - * Try to stop all running streams in the substream group. - * The state of each stream is changed to the given value after that unconditionally. + * The state of each stream is then changed to the given state unconditionally. */ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) { @@ -941,11 +938,10 @@ int snd_pcm_stop(struct snd_pcm_substrea EXPORT_SYMBOL(snd_pcm_stop); /** - * snd_pcm_drain_done + * snd_pcm_drain_done - stop the DMA only when the given stream is playback * @substream: the PCM substream * - * Stop the DMA only when the given stream is playback. - * The state is changed to SETUP. + * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. */ int snd_pcm_drain_done(struct snd_pcm_substream *substream) @@ -1065,10 +1061,9 @@ static struct action_ops snd_pcm_action_ }; /** - * snd_pcm_suspend + * snd_pcm_suspend - trigger SUSPEND to all linked streams * @substream: the PCM substream * - * Trigger SUSPEND to all linked streams. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend(struct snd_pcm_substream *substream) @@ -1088,10 +1083,9 @@ int snd_pcm_suspend(struct snd_pcm_subst EXPORT_SYMBOL(snd_pcm_suspend); /** - * snd_pcm_suspend_all + * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm * @pcm: the PCM instance * - * Trigger SUSPEND to all substreams in the given pcm. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend_all(struct snd_pcm *pcm) @@ -1313,11 +1307,9 @@ static struct action_ops snd_pcm_action_ }; /** - * snd_pcm_prepare + * snd_pcm_prepare - prepare the PCM substream to be triggerable * @substream: the PCM substream instance * @file: file to refer f_flags - * - * Prepare the PCM substream to be triggerable. */ static int snd_pcm_prepare(struct snd_pcm_substream *substream, struct file *file)

2 1

[alsa-devel] [PATCH] hda: add support for jack detection on IDT/Sigmatel
by Matthew Ranostay 16 Oct '08

16 Oct '08

Added support for jack detection reporting to userspace for IDT/Sigmatel codecs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/Kconfig b/pci/Kconfig index 7003711..7e40890 100644 --- a/pci/Kconfig +++ b/pci/Kconfig @@ -501,6 +501,7 @@ config SND_HDA_INTEL tristate "Intel HD Audio" select SND_PCM select SND_VMASTER + select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for Intel "High Definition Audio" (Azalia) motherboard devices. diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index a2ac720..f6c0b1d 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -30,6 +30,7 @@ #include <linux/pci.h> #include <sound/core.h> #include <sound/asoundef.h> +#include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" #include "hda_patch.h" @@ -216,6 +217,9 @@ struct sigmatel_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ + /* jack detection */ + struct snd_jack *jack; + /* dynamic controls and input_mux */ struct auto_pin_cfg autocfg; unsigned int num_kctl_alloc, num_kctl_used; @@ -3617,7 +3621,7 @@ static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - int i; + int i, err; snd_hda_sequence_write(codec, spec->init); @@ -3639,6 +3643,12 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], AC_PINCTL_OUT_EN); stac92xx_auto_init_hp_out(codec); + /* jack detection */ + err = snd_jack_new(codec->bus->card, + "Headphone Jack", + SND_JACK_HEADPHONE, &spec->jack); + if (err < 0) + return err; /* fake event to set up pins */ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); } else { @@ -3796,6 +3806,8 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } + snd_jack_report(spec->jack, + presence ? SND_JACK_HEADPHONE : 0); if (presence) { /* disable lineouts, enable hp */

2 1