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

Re: [alsa-devel] Problems regarding alsa-info.sh
by Travis Place 10 Jun '08

10 Jun '08

Sorry for the quick 2nd post, but ive just been thinking about this.. If we can patch the GIT version with the patch i just emailed, and temporarily put that same file (version 0.4.46) into the HG repo (even if it has to be entered manually) then current script users will get the new script (which contains the new URLs). That will solve half the problem. Then we just need a friendly URL to post in wikis etc. Ive setup a very temporary tinyurl page.. http://tinyurl.com/5bz98h Seems to work ok, but we want something more perm, and more professional looking, really. Travis Place.

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[alsa-devel] What does 0dB refer to? (Logitech USB Speakers)
by Lennart Poettering 10 Jun '08

10 Jun '08

Heya! I think I discussed this with both James and Takashi a while back: it would be great if we could initialize the sound card mixers in a better way by default and not everything set to 0. i.e. for output only PCM should be active, at a sensible volume, no bass/treble/3d enabled or whatever. All other sources should be muted. And that for input the mic jack should be active and nothing else, again. Right now people too often have to invest a lot of time to find the right controls to toggle/change in the mixer to get something sensible out of the speakers. Some distributions have started to ship special scripts that try to initialize the volumes properly. I know for the Fedora case that this tool is very very simple, and just initializes PCM and Master to 70% and that's basically it. It would be highly preferable if we had a sane tool like this from alsa upstream, which would set the default values to 0dB, possibly as an extension to "alsactl". I started to hack something similar to this as part of PulseAudio: if PA finds that the PCM slider is not really useful for volume control (no attenuation to at least -60dB, too few steps, no seperate controls for all channels, ...) it falls back to software volume control, however tries to initialize the the slider to 0dB first. Now the thing is, this mostly works fine, however on one of my devices, a pair of external Logitech USB speakers (ID 046d:0a04 Logitech, Inc. V20 portable speakers), the dB scale seems to be totally bogus. According to ALSA volume ranges from -41 dB to +3dB. However, Every setting > -41 dB makes audio awfully loud. Really, really fucking loud that is. Disco loud. As loud that the speakers start jump around centimeters due to the massive vibrations. Setting the volume to 0dB with those speakers really makes you fear you are trashing them. I somehow expected that the magic "0dB" setting in ALSA would refer to some sane default, where the final output would have a sensible volume. In this case this is not so. I am thus wondering how ALSA actually defines 0dB? Is this a bug in the hardware that should be worked around in ALSA? Is this an ALSA bug? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] Fix buffer position for ATI SB4x0
by Mauro Carvalho Chehab 09 Jun '08

09 Jun '08

ATI SB4x0 doesn't need any fix at position. This patch solves the issue of receiving several clicks during capture on those devices. Tested with a Gateway Notebook MX-6453. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)infradead.org> diff -r 33d7308ade64 pci/hda/hda_intel.c --- a/pci/hda/hda_intel.c Fri May 23 17:36:59 2008 -0300 +++ b/pci/hda/hda_intel.c Thu May 29 11:04:56 2008 -0300 @@ -1860,6 +1860,7 @@ SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE), + SND_PCI_QUIRK(0x1002, 0x437b, "ATI SB4x0 HDA", POS_FIX_NONE), {} };

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[alsa-devel] ASOC PCM formats
by Jon Smirl 09 Jun '08

09 Jun '08

Why is the AC97 format set to S16 in all of the drivers? The link is 20b and my codec takes 20b, how do I get 20b data to the device? I tried using SNDRV_PCM_FORMAT_S20_3BE but all I get is static. If I use S24 for the format it works but volume is divided by 16. I think the bits have been right justified into the 24 bits so the bottom four bits are lost. AC97 link is Efika, MPC5200 Codec is STAC9766. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] [PATCH] opti93x: fix sound ouput for Opti930
by Krzysztof Helt 09 Jun '08

09 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch fixes silenced output from the Opti930. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The bug was introduced by me in the patch which added QS1000 wavetable support: http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commit;h… Simply do not change digital input bits on Opti930. diff -urp linux-rc2/sound/isa/opti9xx/opti92x-ad1848.c linux-rc5/sound/isa/opti9xx/opti92x-ad1848.c --- linux-rc2/sound/isa/opti9xx/opti92x-ad1848.c 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-07 23:59:27.218950875 +0200 @@ -491,16 +491,9 @@ static int __devinit snd_opti9xx_configu break; #else /* OPTi93X */ - case OPTi9XX_HW_82C930: case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | - (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), - 0x34); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); - /* + /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. */ @@ -510,6 +503,13 @@ static int __devinit snd_opti9xx_configu * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); + case OPTi9XX_HW_82C930: /* FALL THROUGH */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | + (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), + 0x34); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); break; #endif /* OPTi93X */ --------------------------------------------------------------- Sprawdz jak zdobyc zdrowy usmiech! Kliknij >> http://link.interia.pl/f1e26

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[alsa-devel] [PATCH] Fix "alc262_sony_unsol[]" hda_verb array
by idak 09 Jun '08

09 Jun '08

I think that hda_verb array must have "terminator (empty array)". But alc262_sony_unsol[] does not have it. And it causes gcc-4.3's buggy behavior with snd_hda_sequence_write(). Signed-off-by: Akio Idehara <zbe64533(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) --- a/sound/pci/hda/patch_realtek.c 2008-06-08 19:23:57.000000000 +0900 +++ b/sound/pci/hda/patch_realtek.c 2008-06-08 19:42:19.000000000 +0900 @@ -8839,6 +8839,7 @@ static struct hda_verb alc262_sony_unsol {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} }; /* mute/unmute internal speaker according to the hp jack and mute state */

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[alsa-devel] [PATCH] Add Toshiba dynabook SS RX1 support
by Akio Idehara 09 Jun '08

09 Jun '08

I have Toshiba dynabook SS RX1 and this patch adds that support. Signed-off-by: Akio Idehara <zbe64533(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) --- a/sound/pci/hda/patch_realtek.c 2008-06-08 19:23:57.000000000 +0900 +++ b/sound/pci/hda/patch_realtek.c 2008-06-08 20:55:34.000000000 +0900 @@ -9718,6 +9718,8 @@ static struct snd_pci_quirk alc262_cfg_t SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),

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[alsa-devel] A little problem with main gitweb page
by Rafał Mużyło 08 Jun '08

08 Jun '08

Well, it would be nice if http://git.alsa-project.org/ listed alsa-plugins git too.

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[alsa-devel] Problems regarding alsa-info.sh
by Travis Place 08 Jun '08

08 Jun '08

Since the move from HG->GIT there is now a new download location for alsa-info.sh . There are 2 current problems with this though.. 1) The new URL is very ugly(and hard to post on wikis etc) The new link is http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=utils/alsa-in… 2) The current script version at HG repo (which is where people currently get it from, is an outdated version, that checks the HG repo for updates to the script. We need to put the new version of the script into HG (if its still accessible), with the updated 'update' URL, or, somehow redirect from the old URL to the new URL. Please let me know what should be done about this, and how to go about fixing it. Attached is the updated alsa-info.sh with the new GIT URLS (update and changelog) in place.

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[alsa-devel] Wrong period size interval with hda-intel makes snd_pcm_set_params() fail
by Christian Morales Vega 08 Jun '08

08 Jun '08

I looked deeper into the problem and looks like a hda-intel problem. This code fails with "ALSA lib pcm.c:7193:(snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" #include <alsa/asoundlib.h> int main(){ snd_pcm_t *handle; if(snd_pcm_open(&handle, "default:1", SND_PCM_STREAM_PLAYBACK, 0) < 0) { exit(1); } if(snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, 24000, 1, 100000) < 0) { exit(2); } } The "Invalid argument" comes from pcm_params.c:145. The test "!snd_interval_single(i)" fails since the interval for SND_PCM_HW_PARAM_PERIOD_SIZE ends being min=511 and max=513. I understand that this interval is set at the driver, correct? I don't know where exactly, so some help in finding it or a patch would be welcome. I'm using alsa-driver-kmp-default-1.0.16.20080608, libasound2-1.0.16.hg20080510 and alsa-1.0.16.hg20080510 RPM packages from multimedia:audio openSUSE Build Service repository (from Takashi Iwai) and an AD1988 (hda-intel) codec. A XUbuntu user also has the same problem with an hda-intel codec (that I suppose uses stable 1.0.16). With a SB Live! in the same computer there are no problems.

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