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[alsa-devel] [PATCH] ALSA: usb/mixer.c: remove unused variable
by Daniel Mack 31 May '10

31 May '10

This was a development left-over. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/mixer.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index fd3e08d..a35dd74 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1970,8 +1970,6 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, int attribute, int value, int index) { - struct usb_mixer_elem_info *info; - switch (attribute) { case UAC2_CS_CUR: snd_usb_mixer_notify_id(mixer, index); -- 1.7.1

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Re: [alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by Jaroslav Kysela 30 May '10

30 May '10

On Thu, 27 May 2010, VDR User wrote: > On Thu, May 27, 2010 at 12:37 AM, Jaroslav Kysela <perex(a)perex.cz> wrote: >> Check any suspicious lines in /var/log/messages and follow proc xrun_debug >> settings noted in bug above. I need at least 20 first lines after the >> playback is started. > > By this do you mean compile the drivers with pcm-xrun-debug > --with-debug=verbose --enable-verbose-procfs options? Also, is there > a way to disable all the debug logging in maybe > /etc/modprobe.d/alsa-base.conf? My system is installed on a CF and I > can't have alsa hammering it with writes is they're not needed but the > only way I know to turn the debug logging off is to not compile it in > the first place. I asked for one run, so you can recompile ALSA driver without debugging support after test. I think that the problem might be that the WALLCLK register gives wrong values for your hardware (but the Intel specification says that implementation of this register is MANDATORY). Here is a patch which might show the bug: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170610e..a1e7c97 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1891,6 +1891,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) int stream; wallclk = azx_readl(chip, WALLCLK) - azx_dev->start_wallclk; + printk("wallclk: %u, %u\n", wallclk, azx_readl(chip, WALLCLK)); if (wallclk < (azx_dev->period_wallclk * 2) / 3) return -1; /* bogus (too early) interrupt */ >> Also, use 'aplay -Dplughw:0' from the alsa-utils package to trigger the >> playback. > > The hdmi device I'm using is 1,3: > > test:~$ aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: NVidia [HDA NVidia], device 0: ALC883 Analog [ALC883 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: NVidia [HDA NVidia], device 1: ALC883 Digital [ALC883 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: NVidia_1 [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > So what exactly do I need to do then with aplay? Do I need some file > to play? Sorry, I've never used Linux as a desktop so I'm only > familiar with running VDR from command line. If you use HDMI device, use 'aplay plughw:1,0 <some_wav_file>'. Some_wav_file means a .wav type audio file. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] How to support software mixing on QUALCOMM 72XX without mmap?
by Berg.Tian 29 May '10

29 May '10

Hello,guys. I'm porting ASLA1.0.23 to qualcomm 72xx base on ARM11. The first instance,"aplay test.wav", works now, but the second instance should fail with "device busy" error message. I have tried plug dmix and dshare "aply -D plug:mycard test.wav", but met a fail again for the soc chip doesn't support mmap. Of course,pcm_mycard is defined in /etc/asound.conf. In my pc ,both above cmds work well . Though,the document shows that: NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix for analogue output. Dmix is enabled by default for soundcards which don't support hardware mixing. You still need to set it up for digital outputs. My questions are: 1)Is software mixing implemented in ALSA Driver,or in ALSA LIB? 2)What could I do to make the software mixing work? Is there any special configure which need to adjust? And when testing on pc whth HD audio, I noticed that A)when aplay instances runs WITHOUT dmix, the result of"cat /etc/one_of_my_aplay_pid/maps" should like: ... b331d000-b731e000 rw-s 00000000 00:15 862346 /dev/shm/pulse-shm-2882420837 ... B)when aplay instances runs WITH dmix,the result is: ... b7511000-b7519000 rw-s 00000000 00:0f 4560 /dev/snd/pcmC0D0p b7519000-b751a000 rw-s 81000000 00:0f 4560 /dev/snd/pcmC0D0p b751a000-b751b000 r--s 80000000 00:0f 4560 /dev/snd/pcmC0D0p ... So,the third question: 3) From A) and B),can I make a result that the mmap feacture is not necessary for the buildin software mixing ? Expect your help,please berg

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Re: [alsa-devel] Mixer volume/decibel mapping
by Clemens Ladisch 29 May '10

29 May '10

Raymond Yau wrote: > 2010/5/21 Clemens Ladisch <clemens(a)ladisch.de> > > DB_LINEAR means that the mixer control value is a multiplier for the > > PCM sample value, as James wrote. A DB_LINEAR mapping has > > 0 == -infinity dB, full scale == 0 dB, and the mapping between mixer > > control values and corresponding dB values it _not_ linear. > > pci/oxygen/oxygen.c:static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > pci/oxygen/hifier.c:static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > > Refer to AK4396 datasheet > > Output Volume > > The AK4396 includes channel independent digital output volumes (ATT) with > 256 levels at linear step including MUTE. > These volumes are in front of the DAC and can attenuate the input data from > 0dB to –48dB and mute. > > dB range is -48dB to 0dB only > > ATT7-0: Attenuation Level > ATT = 20 log10 (ATT_DATA / 255) [dB] > FFH : 0dB (Default) > 00H : Mute This is the section of the datasheet on which the above declarations are based. Now if you have some remark or question, you should remember to include it in your mail. Regards, Clemens

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Re: [alsa-devel] [PATCH] ALSA/davinci: Use the resource_size , __devinit and __devexit.
by Sergei Shtylyov 28 May '10

28 May '10

Hello. Wan ZongShun wrote: > Dear all, > Use the resource_size function instead of manually calculating the > resource size. This reduces the chance of introducing off-by-one > errors. > In addition, to use __devinit and __devexit is good too. "Good" isn't a good justification I think. Also, this just asks to be put into two patches instead of one as the patch does two completely different things. > Signed-off-by:Wan ZongShun<mcuos.com(a)gmail.com> WBR, Sergei

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[alsa-devel] [PATCH] ALSA, hda: Add support for another Lenovo ThinkPad Edge in conexant codec
by Andreas Herrmann 28 May '10

28 May '10

From: Andreas Herrmann <andreas.herrmann3(a)amd.com> On a Thinkpad Edge 13 "01972NG" I had the problem that speakers played sound although headphones were plugged in. Using model=ideapad with latest alsa-git kernel fixed this. So adding this quirk to use ideapad for another Thinkpad Edge variant seems sensible. Cc: Jerone Young <jerone.young(a)canonical.com> Signed-off-by: Andreas Herrmann <andreas.herrmann3(a)amd.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) Patch is against today's sound-2.6 git tree. Please apply. Thanks, Andreas diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f4a2bd6..2bf2cb5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2975,6 +2975,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), -- 1.7.1

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Re: [alsa-devel] [PATCH] ALSA, hda: Add support for Lenovo ThinkPad Edge 13 in conexant codec
by Takashi Iwai 28 May '10

28 May '10

At Thu, 27 May 2010 18:06:56 +0200, Andreas Herrmann wrote: > > From: Andreas Herrmann <andreas.herrmann3(a)amd.com> > > On a Thinkpad Edge 13 I had the problem that speakers played sound > although headphones were plugged. > > Using "model=olpc-xo-1_5" solved this problem. So it seems sensible to > add this quirk. (Using other models, like ideabad did not fix it.) > > Signed-off-by: Andreas Herrmann <andreas.herrmann3(a)amd.com> > --- > sound/pci/hda/patch_conexant.c | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > I don't know whether adding this quirk is the right/complete solution > to support sound on Thinkpad Edge. But at least it solves this > annoying problem. Well, Jerone Young already sent a quirk patch but he took model=ideapd. Could you guys check which one is really better? thanks, Takashi > > Please apply. > > > Thanks, > > Andreas > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index feabb44..f85f072 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -2848,6 +2848,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { > SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), > + SND_PCI_QUIRK(0x17aa, 0x21b3, "Thnkpad Edge 13", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), > {} > }; > -- > 1.6.4.4 >

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Re: [alsa-devel] [PATCH 3/4] ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
by Takashi Iwai 28 May '10

28 May '10

At Wed, 26 May 2010 13:46:57 -0700, Greg KH wrote: > > On Wed, May 26, 2010 at 06:17:46PM +0200, Daniel Mack wrote: > > On Wed, May 26, 2010 at 06:11:38PM +0200, Daniel Mack wrote: > > > UAC2 devices have their information about pitch control stored in a > > > different field. Parse it, and emulate the bits for a v1 device. > > > > > > A new struct uac2_iso_endpoint_descriptor is added. > > > > > > > > Signed-off-by: Daniel Mack <daniel(a)caiaq.de> > > > --- > > > include/linux/usb/audio-v2.h | 16 ++++++++++++ > > > > Greg, I copied you because of this patch, just to let you know we're > > having fun in your area of responsibility again :) > > Heh, no problem at all. > > Takashi, feel free to add an: > Acked-by: Greg Kroah-Hartman <gregkh(a)suse.de> > to this patch when you take it. OK, I put this to my tree. Will be included in the next pull request in this week. thanks, Takashi

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for Sony VPCS11V9E
by Daniel Chen 28 May '10

28 May '10

Hi, Attached is a patch against sound-2.6 master HEAD that resolves stuttering audio on the original reporter's hardware. Please apply. This patch is applicable to all stable trees of 2.6.3x. Best, -Dan

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[alsa-devel] usbaudio: Support for USB audio v2.0 devices
by Daniel Mack 27 May '10

27 May '10

Hi, I've been working on support for USB devices compliant to audio class v2.0 and would like to share what I got. First problem is that they literally changed every single descriptor type by adding or removing fields or changing their width, respectively. Even worse, some enum values were dropped and reassigned so their numerical value has a different meaning now. My first step was to clean up most of the existing descriptor parsers by moving away from anonymous byte array access to real structs so we can see which fields we're actually dereferencing. In a second step, I added definitions for the replacement descriptors and use them at appropriate places. The bInterfaceProtocol field must always be set correctly of course, otherwise we'll parse garbage. Other things that have changed from v1.0 to v2.0 are: * The number of streaming interfaces is now reported by a standard interface association descriptor. The old approach using a proprietary descriptor extension is deprecated. * The number of channels per interface is now stored in the AS_GENERAL descriptor (used to be part of the FORMAT_TYPE descriptor). * The list of supported sample rates is no longer stored in a variable length appendix of the FORMAT_TYPE descriptor but is retrieved from the device using a class specific GET_RANGE command. * Supported sample formats are now reported as 32bit bitmap rather than as a fixed value. For now, this is worked around by choosing just one of them. Eventually, the code should propagate the full ability of the device. * A devices needs to have at least one CLOCK_SOURCE descriptor which denotes a clockID that is needed as argument to the class request command. If can have more than one though and the host driver is in charge to choose the right one. There are also new types for clock selectors and sample rate converters that a device might report. This is all unsupported at the moment. We only parse the first CLOCK_SOURCE and take this one. * Many descriptors (format_type, ...) have changed their layout. Handle this by casting the descriptors to the appropriate structs. With the following patches applied, a v2.0 device is recognized and reported as ALSA device. I can even hear at least some kind of music, but the sample rate setting does not yet seem to be correct. Also, the existing mixer code does not work at all for v2.0 devices. This will be quite some work to support that and I fear the whole thing must be reimplemented for v2.0 specifically. For now, I worked around that by not parsing them at all but bailing out very early. However, I would like to make sure the changes I made so far won't cause any regression for v1.0 devices, and so it would be good to see them applied as a preliminary support. I'll continue working on this and send more updates along. Let me know what you think. Thanks, Daniel

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