June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [Bug] ICE1712 analog outputs are unusable
by Peter Lawler 20 Jun '09

20 Jun '09

http://mailman.alsa-project.org/pipermail/alsa-devel/2009-January/014108.ht… https://bugzilla.redhat.com/show_bug.cgi?id=499435 http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018616.html The original patch to fix this was declined and thus the issue has been left to fester. I'm available to help test out any new patch proposals etc. Please don't hesitate to contact me etc. Pete.

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[alsa-devel] IDT quirks...where?
by Adam Williamson 20 Jun '09

20 Jun '09

Hi, guys. Following up a Fedora forum quirk report: http://forums.fedoraforum.org/showpost.php?p=1229209&postcount=16 elsewhere in the thread, the reporter is instructed to try the hp-dv5 model parameter, and reports that this works. The alsa-info output shows the subsystem ID as 103c:3621, and the codec as IDT 92HD71B7X. This is one I haven't seen before. I wanted to write up a bug report with a patch to add a quirk for this chipset, but I can't see which of the quirk files it should go into - none of the patch_*.c files seems to cover IDT codecs. What's the story with these? Where do their quirks go? Does current ALSA have one already? (I did try searching through each of the patch_*.c files for 3621 and found nothing, but I may have missed something). Thanks! -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Fedora Talk: adamwill AT fedoraproject DOT org http://www.happyassassin.net

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[alsa-devel] LSB inclusion of ALSA
by Robert Schweikert 19 Jun '09

19 Jun '09

Hi, Currently ALSA is a trial use module for LSB (http://www.linuxfoundation.org/collaborate/workgroups/lsb) 4.0 and it is the intention to make ALSA mandatory with the LSB 4.1 release targeted for Q1 2010. One of the requirements of a mandatory module is the existence of a test suite for the module that covers a good chunk of the interfaces provided by the LSB spec. An initial exploratory look at the existing test cases showed that tests appear to be tied to hardware drivers and are interactive. Is this initial impression correct? _From an LSB testing perspective, interface testing is very important; i.e. I call the interface with the arguments required and I get the expected result back. Further the tests need to run in an automated fashion and produce a certain output (automated processing). The output can be created by retrofitting existing tests or by specifically writing tests within the LSB test framework. IMHO it would be best to have tests that are useful to both, the project and the LSB and in a best case scenario we wouldn't have to start over as the existing tests already cover a good part of the interface. With this as background there are a few questions that have been raised while discussing this during LSB workgroup sessions. 1.) Is there interest from the community side to participate in this effort and accept patches? 2.) Is it reasonable to expect that the existing tests can be used in some way by using a dummy sound device or a sound loop device to verify output? One requirement for LSB testing is that a test has only two states, pass or fail, and a recovery or try again mode is not supported. Further a test must pass or fail. 3.) Would the community be more comfortable if this is an effort that creates new tests separate from the existing HW focused tests? 4.) Is anyone interested in helping with this effort, writing tests, answering questions about ALSA for those not familiar with the interface and or sound in general? Comments, thoughts, etc. are much appreciated. Thanks, Robert -- Robert Schweikert MAY THE SOURCE BE WITH YOU Software Engineer Consultant LINUX rschweikert(a)novell.com 781-464-8147 Novell Making IT Work As One

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Re: [alsa-devel] Laptop (Samsung Q45) speakers unmute when master channel is unmuted even with headphones connected
by Takashi Iwai 19 Jun '09

19 Jun '09

At Fri, 19 Jun 2009 11:56:29 +0100, Sam Morris wrote: > > On Thu, 2009-05-07 at 17:11 +0200, Takashi Iwai wrote: > > At Tue, 05 May 2009 11:58:39 +0100, > > Sam Morris wrote: > > > > > > On Tue, 2009-05-05 at 12:47 +0200, Takashi Iwai wrote: > > > > At Sat, 2 May 2009 13:14:42 +0100, > > > > Sam Morris wrote: > > > > > > > > > > Since Linux 2.6.28 and through to 2.6.30-rc4, the hda-intel driver has had a > > > > > bug on my laptop (a Samsung Q45). > > > > > > > > > > If I connect my headphones, the laptop's internal speakers are muted--fine. But > > > > > if I then mute and unmute the master mixer channel, the speakers come back to > > > > > life, even though the headphones are still connected. > > > > > > > > Could you run alsa-info.sh with --no-upload option, and attach the > > > > generated file? > > > > > > Here you go! > > > > Thanks. > > > > What if you use model=auto option? > > This is fixed in 2.6.30, but only if I use the model=auto option. :) What about with the later version, e.g. Linus git tree? Or you can use sound git tree git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git I think this was already fixed there. Takashi

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Re: [alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
by Liam Girdwood 19 Jun '09

19 Jun '09

On Fri, 2009-06-19 at 17:28 +0800, Barry Song wrote: > 1. add AD1938 codec driver (codec) > 2. add blackfin SPORT-TDM DAI and PCM driver (platform) > 3. add bf5xx board with AD1938 driver (machine) > Signed-off-by: Barry Song <21cnbao(a)gmail.com> > --- > include/sound/soc-dai.h | 1 + > sound/soc/blackfin/Kconfig | 46 +++- > sound/soc/blackfin/Makefile | 6 + > sound/soc/blackfin/bf5xx-ad1938.c | 178 ++++++++++++ > sound/soc/blackfin/bf5xx-tdm-pcm.c | 330 ++++++++++++++++++++++ > sound/soc/blackfin/bf5xx-tdm-pcm.h | 21 ++ > sound/soc/blackfin/bf5xx-tdm.c | 295 +++++++++++++++++++ > sound/soc/blackfin/bf5xx-tdm.h | 14 + > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/ad1938.c | 548 ++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/ad1938.h | 68 +++++ > 12 files changed, 1512 insertions(+), 1 deletions(-) > create mode 100644 sound/soc/blackfin/bf5xx-ad1938.c > create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c > create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h > create mode 100644 sound/soc/blackfin/bf5xx-tdm.c > create mode 100644 sound/soc/blackfin/bf5xx-tdm.h > create mode 100644 sound/soc/codecs/ad1938.c > create mode 100644 sound/soc/codecs/ad1938.h > Could you break this patch into smaller chunks. This should make it easier to review and apply. Thanks Liam

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[alsa-devel] RFC: new TLV types for dB range with min/max
by Takashi Iwai 19 Jun '09

19 Jun '09

Hi, while I was checking bug reports regarding USB mixer dB ranges, I found that my old code might not work always properly. In usbmixer.c, /* USB descriptions contain the dB scale in 1/256 dB unit * while ALSA TLV contains in 1/100 dB unit */ scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256; scale[3] = (convert_signed_value(cval, cval->res) * 100) / 256; Since the step size may have a rounding error, this can accumulate to a significant amount at the max value. A simple way to avoid this is to just give min and max values instead of min and step values, and interpolate the middle. For that, I create two new TLV types. Why two? Since the mute bit can't be packed into an integer (the dB range can be both positive and negative values) unlike the step size, I provided here both with and without mute, thus two. The kernel patch is attached below. This and a patch to usb-audio can be found in topic/tlv-minmax branch of sound-unstable GIT tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git topic/tlv-minmax I already wrote and test it with the modified alsa-lib, and the result looks good. If anyone has suggestions or objections to this, please let me know. If this change is OK, I'm going to merge this later for the next, 2.6.32 kernel. thanks, Takashi --- From 085f30654175a91c28d2b66b9ea6cceab627fed0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Tue, 16 Jun 2009 13:57:07 +0200 Subject: [PATCH] ALSA: Add new TLV types for dBwith min/max Add new types for TLV dB scale specified with min/max values instead of min/step since the resolution can't match always with the one a device provides. For example, usb audio devices give 1/256 dB resolution while ALSA TLV is based on 1/100 dB resolution. The new min/max types have less problems because the possible rounding error happens only at min/max. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/tlv.h | 14 ++++++++++++++ sound/core/vmaster.c | 8 ++++++-- 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/include/sound/tlv.h b/include/sound/tlv.h index d136ea2..9fd5b19 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -35,6 +35,8 @@ #define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */ #define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */ #define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */ +#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ +#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ #define TLV_DB_SCALE_ITEM(min, step, mute) \ SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ @@ -42,6 +44,18 @@ #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } +/* dB scale specified with min/max values instead of step */ +#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) } +#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) } + /* linear volume between min_dB and max_dB (.01dB unit) */ #define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \ SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \ diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 257624b..3b9b550 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -353,7 +353,8 @@ static void master_free(struct snd_kcontrol *kcontrol) * * The optional argument @tlv can be used to specify the TLV information * for dB scale of the master control. It should be a single element - * with #SNDRV_CTL_TLVT_DB_SCALE type, and should be the max 0dB. + * with #SNDRV_CTL_TLVT_DB_SCALE, #SNDRV_CTL_TLV_DB_MINMAX or + * #SNDRV_CTL_TLVT_DB_MINMAX_MUTE type, and should be the max 0dB. */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv) @@ -384,7 +385,10 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, kctl->private_free = master_free; /* additional (constant) TLV read */ - if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) { + if (tlv && + (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE || + tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX || + tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) { kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; memcpy(master->tlv, tlv, sizeof(master->tlv)); kctl->tlv.p = master->tlv; -- 1.6.3.2

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Re: [alsa-devel] RFC: PCM extra attributes
by Takashi Iwai 19 Jun '09

19 Jun '09

At Fri, 19 Jun 2009 10:54:15 +0200, Florian Faber wrote: > > Takashi, > > > An important note is that I'm planning to use this framework for > > getting/setting the PCM (surround) channel mapping. > > I don't understand what this means. Who maps what on what and why? It's a mapping of surround channels. Right now, ALSA assumes (usually) the order of channels like front-left, front-rear, surround-left, surround-rear, center, LFE, ... But this really depends on the device. Some (or many) need FL/FR/C/LFE/SR/SR order, for example. And, some devices can allow you even to reassign the channel map. This is likely an additional information, which not all drivers have to provide. Takashi

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[alsa-devel] RFC: "patch" loading for HD-audio driver
by Takashi Iwai 19 Jun '09

19 Jun '09

Hi, another changes I've worked recently on is a "patch"-loading mechanism for HD-audio. Here, a "patch" refers to a sort of firmware file, corresponding "*.INI" file on Windows. As many of you know, HD-audio hardwares have many problems, partly because of crappy broken BIOS and partly because of lack of information that the driver can detect automatically. Until now, these are provided statically in the driver. In the recent version of HD-audio driver, you can override the setup and reconfigure the driver via sysfs. For example, changing the default pin-configurations can be done via sysfs and then reconfigure. However, this procedure can be done only after the driver is set up, not before the driver initialization. So an idea of "patching" the setup before the driver initialization came up. A "patch" file is a text file looks like: ================================================================ [codec] 0x12345678 0xabcd1234 2 [model] auto [pincfg] 0x12 0x411111f0 [verb] 0x20 0x500 0x03 0x20 0x400 0xff [hint] hp_detect = yes ================================================================ This file can be provided in the standard firmware path, typically /lib/firmware, through a module parameter "patch", e.g. options snd-hda-intel patch=my-hda-patch where /lib/firmware/my-hda-patch is the patch file path. The whole patch (I mean program patch) series can be found in topic/hda-patch branch of sound-unstable GIT tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git topic/hda-patch If you are interested, please take a look. Any comments appreciated. This work is intended for 2.6.32 merge as well as other hacks there. thanks, Takashi

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Re: [alsa-devel] kerneloops.org report for the week of June 14 2009
by Arjan van de Ven 19 Jun '09

19 Jun '09

On Thu, 18 Jun 2009 13:08:39 -0400 Bob Copeland <me(a)bobcopeland.com> wrote: > On Sun, Jun 14, 2009 at 8:33 PM, Arjan van de > Ven<arjan(a)infradead.org> wrote: > > Rank 9: minstrel_get_rate (warning) > > Reported 64 times (384 total reports) > > Issue with the ath5k driver > > This warning was last seen in version 2.6.30-rc3, and first > > seen in 2.6.27.12. More info: > > http://www.kerneloops.org/searchweek.php?search=minstrel_get_rate > > I believe this is fixed by http://lkml.org/lkml/2009/6/5/269, now > in pre-2.6.31. Hmm, odd though, we should be seeing reports from > other wireless drivers since it's a general memory corruption > unrelated to ath5k (but not all use the mac80211 rate controllers). > well.. out of the 534 reports or so that I have by now... 525 have ath5k loaded..... it's not 100% but it'd pretty close. -- Arjan van de Ven Intel Open Source Technology Centre For development, discussion and tips for power savings, visit http://www.lesswatts.org

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Re: [alsa-devel] [PATCH v2] alsa-plugins/pulse: Implement 'pause'.
by Takashi Iwai 18 Jun '09

18 Jun '09

At Thu, 18 Jun 2009 14:55:21 +0100 (BST), Troy Moure wrote: > > Just cork or uncork the stream to pause or unpause it. > > Signed-off-by: Troy Moure <twmoure(a)szypr.net> Applied now. Thanks! Takashi > --- > On Wed, 17 Jun 2009, Lennart Poettering wrote: > > > Looks mostly good to me, but: > > > ... > > > > pa_stream_cork() will return a pa_operation object on success. The > > least you need to do is call pa_operation_unref() on it, which you can > > safely do right-away. Otherwise this will leak memory. > > > > Lennart > > Thanks, Lennart, for looking at the first verson I sent. I've added a > call to pa_operation_unref() as you suggested. > > pulse/pcm_pulse.c | 25 +++++++++++++++++++++++++ > 1 files changed, 25 insertions(+), 0 deletions(-) > > diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c > index db8d1e1..c276839 100644 > --- a/pulse/pcm_pulse.c > +++ b/pulse/pcm_pulse.c > @@ -739,6 +739,30 @@ static int pulse_close(snd_pcm_ioplug_t * io) > return 0; > } > > +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) > +{ > + snd_pcm_pulse_t *pcm = io->private_data; > + int err = 0; > + > + assert (pcm); > + assert (pcm->p); > + > + pa_threaded_mainloop_lock(pcm->p->mainloop); > + > + if (pcm->stream) { > + pa_operation *o; > + o = pa_stream_cork(pcm->stream, enable, NULL, NULL); > + if (o) > + pa_operation_unref(o); > + else > + err = -EIO; > + } > + > + pa_threaded_mainloop_unlock(pcm->p->mainloop); > + > + return err; > +} > + > static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .start = pulse_start, > .stop = pulse_stop, > @@ -750,6 +774,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .prepare = pulse_prepare, > .hw_params = pulse_hw_params, > .close = pulse_close, > + .pause = pulse_pause > }; > > > -- > 1.6.2.2 > > > > > > > > > -- > > Lennart Poettering Red Hat, Inc. > > lennart [at] poettering [dot] net > > http://0pointer.net/lennart/ GnuPG 0x1A015CC4 > > >

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Re: [alsa-devel] [PATCH v2] alsa-plugins/pulse: Implement 'pause'.
by Lennart Poettering 18 Jun '09

18 Jun '09

On Thu, 18.06.09 14:55, Troy Moure (twmoure(a)szypr.net) wrote: > > pa_stream_cork() will return a pa_operation object on success. The > > least you need to do is call pa_operation_unref() on it, which you can > > safely do right-away. Otherwise this will leak memory. > > > > Lennart > > Thanks, Lennart, for looking at the first verson I sent. I've added a > call to pa_operation_unref() as you suggested. Looks good to me now. > > pulse/pcm_pulse.c | 25 +++++++++++++++++++++++++ > 1 files changed, 25 insertions(+), 0 deletions(-) > > diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c > index db8d1e1..c276839 100644 > --- a/pulse/pcm_pulse.c > +++ b/pulse/pcm_pulse.c > @@ -739,6 +739,30 @@ static int pulse_close(snd_pcm_ioplug_t * io) > return 0; > } > > +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) > +{ > + snd_pcm_pulse_t *pcm = io->private_data; > + int err = 0; > + > + assert (pcm); > + assert (pcm->p); > + > + pa_threaded_mainloop_lock(pcm->p->mainloop); > + > + if (pcm->stream) { > + pa_operation *o; > + o = pa_stream_cork(pcm->stream, enable, NULL, NULL); > + if (o) > + pa_operation_unref(o); > + else > + err = -EIO; > + } > + > + pa_threaded_mainloop_unlock(pcm->p->mainloop); > + > + return err; > +} > + > static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .start = pulse_start, > .stop = pulse_stop, > @@ -750,6 +774,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .prepare = pulse_prepare, > .hw_params = pulse_hw_params, > .close = pulse_close, > + .pause = pulse_pause > }; > > Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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Re: [alsa-devel] Regarding internal mic on Aspire 6935G
by Karthik Ramgopal 18 Jun '09

18 Jun '09

Hi, How to check if it works in stereo or mono mode? I remember reading hector's entry on his blog that on account of some wierd sum/difference problem he had duplicated the + signal across both outputs so it was actually mono? Regards, Karthik

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[alsa-devel] preventing SNDRV_PCM_STATE_XRUN
by Sergey Lapin 18 Jun '09

18 Jun '09

Hi, all! I have uncontrollable data stream which I need to play through kernel via ALSA snd_pcm_lib_write call. Everything is fine, but once a few seconds I get SNDRV_PCM_STATE_XRUN on stream and thus, -EPIPE. Sound glitches are minor, but annoying. Is it possible to somehow prevent this from happenning? Is it possible to distinguish if my next chunk of data will fit? Thanks a lot, S.

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[alsa-devel] Regarding internal mic on Aspire 6935G
by Karthik Ramgopal 18 Jun '09

18 Jun '09

Hi, Sorry for the confusion. My internal mic does work, but wierdly not in Gnome Sound recorder. Skype detects it just fine and I am able to talk without making any changes to the code. So, I guess 8930G and 6935G are same after all. Thanks and Regards, Karthik PS: Please also add the PCI ID's of 6920G (0x1025 0146) and 8920G (0x1025 0145) under the same quirk. They also use ALC889 chipset with the same pinconfig.

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Re: [alsa-devel] trace PCM data from usb bus to alsa lib
by Takashi Iwai 18 Jun '09

18 Jun '09

At Wed, 17 Jun 2009 16:10:26 +0800, Zero Luo wrote: > > Dear Takashi, > > Data from usb dongle is 16K*16 bit PCM data. And what is the format and channels you are using in alsa-lib? alsa-lib itself can convert formats and channels automatically if needed. Takashi > 2009/6/17, Takashi Iwai <tiwai(a)suse.de>: > > At Wed, 17 Jun 2009 07:25:25 +0000 (UTC), > ZeroCluo wrote: > > > > Hi > > > > I am trying to trace the Audio data from usb bus. > > in kernel: > > dump data in retire_capture_urb() ,found data format is like: > > AABB 0000 CCDD 0000 ; > > > > but in alsa library , > > dump data in snd_pcm_mmap_begin(),the "0000" is cut,only "AABB CCDD", > > > > Do you know when and where those "0000" is cut? > > Are you using the native format that the hardware runs? > > Takashi > >

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Re: [alsa-devel] snd-ctxfi oops
by The Source 18 Jun '09

18 Jun '09

On 15.06.2009 17:01, Takashi Iwai wrote: > At Mon, 15 Jun 2009 15:00:03 +0200, > I wrote: > >> At Mon, 15 Jun 2009 16:59:53 +0400, >> The Source wrote: >> >>> On 15.06.2009 16:56, Takashi Iwai wrote: >>> >>>> At Sat, 13 Jun 2009 10:07:49 +0200, >>>> I wrote: >>>> >>>> >>>>> >>>>> >>>>>> Tested the patch more. Really no oopses. Instead of them I get rare >>>>>> system lockups. >>>>>> >>>>>> >>>>> In which situation, exactly? >>>>> Could you check whether it happens with use_system_timer=1 module >>>>> option, too? >>>>> >>>>> >>>> Or try the very latest alsa-driver-snapshot (not daily tarball). >>>> I fixed a deadlock issue in cttimer.c now. >>>> >>>> >>>> thanks, >>>> >>>> Takashi >>>> >>>> >>>> >>> Should I apply that patch to this snapshot? >>> >> Which patch do you mean? >> Just grab the latest alsa-driver-snapshot.tar.gz from >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ >> > Also, check alsa-driver/alsa-kernel/HEAD whether the commit > 8dca419721d188bfee5f19fad45275856c619a5c > ALSA: ctxfi - Fix deadlock with xfi-timer > is included. > > > Takashi > > Tested the driver and it's fine so far. No craches with any application including Doom 3, which caused system lockups often with earlier versions of the driver.

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[alsa-devel] [PATCH] alsa-plugins/pulse: Implement 'pause'.
by Troy Moure 17 Jun '09

17 Jun '09

Just cork or uncork the stream to pause or unpause it. Signed-off-by: Troy Moure <twmoure(a)szypr.net> --- I'm not very familiar with alsa or pulseaudio, so this might not be right. If it is, feel free to apply it, though. It *does* fix my problem, which was that pause and unpause didn't work in 'cmus' after I upgraded to Fedora 11. Since the pulse plug-in didn't support snd_pcm_pause, pausing in cmus would just cause cmus to stop writing to the buffer, leading to buffer underruns in pulseaudio. With this patch, pausing in cmus works perfectly for me. If this should be fixed another way, please let me know what it is! Thanks. pulse/pcm_pulse.c | 20 ++++++++++++++++++++ 1 files changed, 20 insertions(+), 0 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index db8d1e1..c322884 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -739,6 +739,25 @@ static int pulse_close(snd_pcm_ioplug_t * io) return 0; } +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) +{ + snd_pcm_pulse_t *pcm = io->private_data; + int err = 0; + + assert (pcm); + assert (pcm->p); + + pa_threaded_mainloop_lock(pcm->p->mainloop); + + if (pcm->stream) + if (!pa_stream_cork(pcm->stream, enable, NULL, NULL)) + err = -EIO; + + pa_threaded_mainloop_unlock(pcm->p->mainloop); + + return err; +} + static const snd_pcm_ioplug_callback_t pulse_playback_callback = { .start = pulse_start, .stop = pulse_stop, @@ -750,6 +769,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { .prepare = pulse_prepare, .hw_params = pulse_hw_params, .close = pulse_close, + .pause = pulse_pause }; -- 1.6.2.2

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[alsa-devel] ICE1712 softdevices fixes
by Stefan Kost 17 Jun '09

17 Jun '09

hi, > The following patches fixes the default device as they appear for > ICE1712-based cards. Without this cards, 'aplay -L' will show a series of > devices like surround40 and surround71 that are not meaningful for this card, > and a front device that would not work. > > Diego E. 'Flameeyes' Pettenò (2): Remove definition for surround sound > devices on ICE1712. Make the front:idx device work for ICE1712. thanks for making those. this was exactly what I needed to get pulseaudio working with my card again. I also agree that even though that my card (Hoontech DSP 24) has 8/10 physical in/outs, those are essentially mono channels. The surround mapping seem to be more harm that use to me. If a musician indeed wires 6 3.5 jack outs somehow to his 5.1 amp, then its probably okay to uncomment such mappings. So what about making diegos patches the default? Stefan

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[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: fix legacy input streaming
by Daniel Mack 17 Jun '09

17 Jun '09

Seems that nobody recently tried the input on the very first supported sound card model, RK2. This patch fixes the byte offset to make it running again. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 5 +++-- sound/usb/caiaq/device.c | 2 +- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b144513..8f9b60c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -199,8 +199,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; } else { - dev->period_in_count[index] = BYTES_PER_SAMPLE; - dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE; + int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2; + dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos; + dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos; } if (dev->streaming) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 2240624..0e5db71 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- 1.6.3.1

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[alsa-devel] sound device is buzy to run pcm.c
by Guilherme Raymo Longo 17 Jun '09

17 Jun '09

I am getting this problem when running the pcm.c demo program at http://alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html This is the execution of the program: guilherme@lap:/home/guilherme/Experimentos# ./PCM Playback device is plughw:0,0 Stream parameters are 44100Hz, S16_LE, 1 channels sine wave rate is 0 0.0000Hz Using transfer method: write Playback open error: Device or resource busy I tried to check if the device is really buzy: root@lap:/home/guilherme/Experimentos# lsof |grep snd lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system */ home/guilherme/*.gvfs Output information may be incomplete. esd 5433 guilherme mem CHR 116,16 6947 /dev/snd/pcmC0D0p esd 5433 guilherme 5r CHR 116,33 6787 /dev/snd/timer esd 5433 guilherme 6w CHR 116,16 6947 /dev/snd/pcmC0D0p esd 5433 guilherme 7r CHR 116,0 6987 /dev/snd/controlC0 and it seems to be ok. That's my card root@lap:/home/guilherme/Experimentos# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Is there a way to free the device or smth is wrong with my code?? I am using slackware 12.2 kernel 2.6 Great Regads;

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[alsa-devel] OSS support
by The Source 17 Jun '09

17 Jun '09

Is oss emulation still supported in latest alsa-driver snapshot? I installed that driver in Fedora 11 on both my static machine (X-Fi platinum fatality champion) and laptop (HDA Intel) and I don't have /dev/dsp or /dev/mixer. Should I compile the driver with some specific flag?

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[alsa-devel] [PATCH 3/3] ASoC: pxa-ssp.c handle the I2S and LEFT_J cases.
by Daniel Ribeiro 17 Jun '09

17 Jun '09

Handles I2S and LEFT_J cases, if the user does _not_ call set_tdm_slot we assume that the codec is I2S compliant, and that we can use frame_width = sample_width * channels. If the user calls set_tdm_slot then we assume that the wire is _not_ networked, and that the user wants to set a non-standard frame format instead, in this case we set the frame duration (the LRCLK falling edge) to be half of the wire frame width. eg. * standard 2*[16|24|32|..] bits I2S/LEFT_J doesn't need to call set_tdm_slot * to use 2*16bits I2S/LEFTJ on 64bits frames, with the LRCLK falling edge at the 32nd bitclk (+1 for I2S) needs to set_tdm_slot(5, 5, 4, 16) (4 slots of 16 bits, first and third RX/TX slots active). This should work even if pxa is slave of SFRM. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 ++-- sound/soc/pxa/pxa-ssp.c | 121 +++++++++++++--------------- 2 files changed, 63 insertions(+), 72 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index 6a2ed35..060e23b 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -108,21 +108,21 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ -#if defined(CONFIG_PXA3xx) -#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ -#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ -#endif - #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ -#define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ #define SSPSP_SFRMDLY(x) ((x) << 9) /* Serial Frame Delay */ -#define SSPSP_DMYSTRT(x) ((x) << 7) /* Dummy Start */ #define SSPSP_STRTDLY(x) ((x) << 4) /* Start Delay */ #define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ #define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ #define SSPSP_SCMODE(x) ((x) << 0) /* Serial Bit Rate Clock Mode */ +/* NOTE: PXA3xx extends the bit number of dummy start and stop, the macros + * below are compatible with PXA25x/27x as long as the parameter is within + * the correct limits, driver code has to take care of this. + */ +#define SSPSP_DMYSTRT(x) ((((x) & 3) << 7) | ((((x) >> 2) & 3) << 26)) +#define SSPSP_DMYSTOP(x) ((((x) & 3) << 23) | ((((x) >> 2) & 7) << 28)) + #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d60492e..aac85fe 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -179,21 +179,6 @@ static void ssp_set_scr(struct ssp_device *ssp, u32 div) ssp_write_reg(ssp, SSCR0, sscr0); } -/** - * ssp_get_clkdiv - get SSP clock divider - */ -static u32 ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = ssp_read_reg(ssp, SSCR0); - u32 div; - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* * Set the SSP ports SYSCLK. */ @@ -487,17 +472,14 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_PSP; - sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - break; - case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + /* See hw_params() for I2S and LEFT_J */ break; default: @@ -561,6 +543,59 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif + sspsp = ssp_read_reg(ssp, SSPSP); + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* + * I2S and LEFT_J are stereo only, we have to send data for + * both channels. + */ + if (chn == 1) + frame_width *= 2; + + /* + * If the user did not use network mode, we assume the codec + * is I2S compliant. + */ + if (frame_width > 0) { + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + sspsp |= SSPSP_FSRT; + } else { + /* + * Otherwise we assume that it is a single TDM slot, and + * the user is abusing set_tdm_slot to support an + * out of spec codec. + */ + int slots = ((sscr0 & SSCR0_SlotsPerFrm(8)) >> 24) + 1; + + /* PXA2XX doesn't support DMYSTOP > 3 */ + if (slot_width != (width * 2) && !cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_DMYSTOP( + (slot_width * slots) / 2 - width - 1); + sspsp |= SSPSP_SFRMWDTH((slot_width * slots) / 2); + } + break; + + case SND_SOC_DAIFMT_LEFT_J: + if (chn == 1) + frame_width *= 2; + + if (frame_width > 0) { + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + } else { + int slots = ((sscr0 & SSCR0_SlotsPerFrm(8)) >> 24) + 1; + + sspsp |= SSPSP_SFRMWDTH((slot_width * slots) / 2); + } + break; + default: + break; + } + ssp_write_reg(ssp, SSPSP, sspsp); + if (frame_width > 0) { /* Not using network mode */ if (frame_width > 16) @@ -598,50 +633,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, ssp_write_reg(ssp, SSCR0, sscr0); - switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); - - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. - */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); - sspsp |= SSPSP_DMYSTRT(1); - } - - ssp_write_reg(ssp, SSPSP, sspsp); - break; - default: - break; - } - dump_registers(ssp); return 0; -- tg: (119677b..) asoc/leftj-and-i2s (depends on: asoc/ssp-internals) -- Daniel Ribeiro

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[alsa-devel] trace PCM data from usb bus to alsa lib
by ZeroCluo 17 Jun '09

17 Jun '09

Hi I am trying to trace the Audio data from usb bus. in kernel: dump data in retire_capture_urb() ,found data format is like: AABB 0000 CCDD 0000 ; but in alsa library , dump data in snd_pcm_mmap_begin(),the "0000" is cut,only "AABB CCDD", Do you know when and where those "0000" is cut? btw,use mmap method map dma_area to user space; and use snd_pcm_readi to get audio data; Thank you very very much. Best Regards, Zero

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Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Wael Nasreddine (a.k.a eMxyzptlk) 17 Jun '09

17 Jun '09

All the information you need are here -> http://www.alsa-project.org/db/?f=6b588ea53af5cf2794fb8b4239311722809b3c34 2009/6/15 Jean-Pierre André <jean-pierre.andre(a)wanadoo.fr>: > Hi Wael, > > Wael Nasreddine (a.k.a eMxyzptlk) wrote: >> >> I don't know if this concerns my situation ( >> http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018262.html >> ) or not.. >> >> headphones work for me, front speaker work but LFE doesn't and the >> Irq22 stays on 0 too >> >> # cat /proc/interrupts | grep '22:' >> 22: 0 0 IO-APIC-fasteoi uhci_hcd:usb6, HDA Intel >> >> Takashi told me to play with hda-analyser and hda-verb though I don't >> what to do with hda-verb and I don't if what hda-analyser showing me >> is the current situation or the desired one.. >> >> Thanks > > So it looks like this a general situation on this hardware. > > If you get sound, at least from the front speakers, > without the IRQ22 being triggered, what are your > kernel and alsa versions, your boot parameters and > alsa (modprobe) parameters ? > > I have no knowledge about the hda-verb, etc. > so I am not to help you much in this area. > > Regards > > Jean-Pierre > > > > -- Wael Nasreddine Blog : http://wael.nasreddine.com E-mail : wael.nasreddine(a)gmail.com gTalk : wael.nasreddine(a)gmail.com Tel : +33.6.32.94.70.13 Skype : eMxyzptlk Twitter : @eMxyzptlk Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 .: An infinite number of monkeys typing into GNU emacs, would never make a good program. (L. Torvalds 1995) :.

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[alsa-devel] [PATCH 1/1] ALC889A chip identification
by Torben Schulz 17 Jun '09

17 Jun '09

With this patch the ALC889A chip will be identified correctly. The functions patch_alc882() and patch_alc883() have been altered to set the stream names correctly. Without this patch 'aplay -l' would identify an ALC889A chip as ALC885 for example. Apart from that a kernel info message will be printed if ALC885 or ALC889A chips are handled by the wrong patch function. Signed-off by: Torben Schulz <public(a)letorbi.de> --- alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2009-05-14 18:23:17.000000000 +0200 +++ alsa-driver-mod/alsa-kernel/pci/hda/patch_realtek.c 2009-05-15 01:08:16.000000000 +0200 @@ -7338,8 +7338,17 @@ setup_preset(spec, &alc882_presets[board_config]); if (codec->vendor_id == 0x10ec0885) { - spec->stream_name_analog = "ALC885 Analog"; - spec->stream_name_digital = "ALC885 Digital"; + if ((codec->revision_id == 0x100101) || + (codec->revision_id == 0x100103)) { + spec->stream_name_analog = "ALC889A Analog"; + spec->stream_name_digital = "ALC889A Digital"; + printk(KERN_INFO + "hda_codec: ALC889A handled by patch_alc882()," + " should be handled by patch_alc883()\n"); + } else { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + } } else { spec->stream_name_analog = "ALC882 Analog"; spec->stream_name_digital = "ALC882 Digital"; @@ -9231,6 +9240,27 @@ setup_preset(spec, &alc883_presets[board_config]); switch (codec->vendor_id) { + case 0x10ec0885: + if ((codec->revision_id == 0x100101) || + (codec->revision_id == 0x100103)) { + spec->stream_name_analog = "ALC889A Analog"; + spec->stream_name_digital = "ALC889A Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ + spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ + } else { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + printk(KERN_INFO + "hda_codec: ALC885 handled by patch_alc883()," + " should be handled by patch_alc882()\n"); + } + break; case 0x10ec0888: if (codec->revision_id == 0x100101) { spec->stream_name_analog = "ALC1200 Analog";

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