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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