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Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 01 Jan '09

01 Jan '09

At Tue, 23 Dec 2008 10:44:50 -0500, Kyler Laird wrote: > > My Panasonic CF-74 Toughbooks do not play audio through the headphone > jack starting with 2.6.28-rc5. The patch below is only to illustrate > what changes I made to restore headphone jack functionality. I do not > suggest using this patch as part of the mainline kernel because it > reverts some code that I'm sure is important to other users. > > Output of the ALSA script aadebug follows the patch. > > I'm continuing to investigate but I wanted to get this out in case the > solution is obvious to someone else. It would be great to have a fix in > 2.6.28. (2.6.28 is the first series to support all of the hardware in > the Toughbook.) Please run alsa-info.sh with --no-upload option and attach the generated files on both the headphone plugged and unplugged. Also, run preferably on both (old) working and non-working kernels to compare. thanks, Takashi

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Re: [alsa-devel] [PATCH 11/13] sound/usb: use USB API functions rather than constants
by Takashi Iwai 01 Jan '09

01 Jan '09

At Mon, 29 Dec 2008 11:23:02 +0100 (CET), Julia Lawall wrote: > > From: Julia Lawall <julia(a)diku.dk> > > This set of patches introduces calls to the following set of functions: > > usb_endpoint_dir_in(epd) > usb_endpoint_dir_out(epd) > usb_endpoint_is_bulk_in(epd) > usb_endpoint_is_bulk_out(epd) > usb_endpoint_is_int_in(epd) > usb_endpoint_is_int_out(epd) > usb_endpoint_num(epd) > usb_endpoint_type(epd) > usb_endpoint_xfer_bulk(epd) > usb_endpoint_xfer_control(epd) > usb_endpoint_xfer_int(epd) > usb_endpoint_xfer_isoc(epd) > > In some cases, introducing one of these functions is not possible, and it > just replaces an explicit integer value by one of the following constants: > > USB_ENDPOINT_XFER_BULK > USB_ENDPOINT_XFER_CONTROL > USB_ENDPOINT_XFER_INT > USB_ENDPOINT_XFER_ISOC > > An extract of the semantic patch that makes these changes is as follows: > (http://www.emn.fr/x-info/coccinelle/) > > // <smpl> > @r1@ struct usb_endpoint_descriptor *epd; @@ > > - ((epd->bmAttributes & \(USB_ENDPOINT_XFERTYPE_MASK\|3\)) == > - \(USB_ENDPOINT_XFER_CONTROL\|0\)) > + usb_endpoint_xfer_control(epd) > > @r5@ struct usb_endpoint_descriptor *epd; @@ > > - ((epd->bEndpointAddress & \(USB_ENDPOINT_DIR_MASK\|0x80\)) == > - \(USB_DIR_IN\|0x80\)) > + usb_endpoint_dir_in(epd) > > @inc@ > @@ > > #include <linux/usb.h> > > @depends on !inc && (r1||r5)@ > @@ > > + #include <linux/usb.h> > #include <linux/usb/...> > // </smpl> > > Signed-off-by: Julia Lawall <julia(a)diku.dk> Applied now. Thanks. Takashi

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[alsa-devel] [PATCH] hda - fix name for ALC1200
by Wu Fengguang 01 Jan '09

01 Jan '09

On Thu, Dec 25, 2008 at 03:13:58PM +0200, Thomas Schneider wrote: > Hi, > > here is the output from alsa-info.sh > > http://www.alsa-project.org/db/?f=8282910dc807e1f141896886cd5f8fd574117000 > > I found another small mistake. In the Mixer my Soundcard is shown as "Realtek ALC888". > When i switch this to lines in patch_realtek.c then it is shown as "Realtek ALC1200". > > > static struct hda_codec_preset snd_hda_preset_realtek[] = { > ...... > > { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, > { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", > .patch = patch_alc883 }, > > ..... Good catch! Reversing the order should fix the name. Thanks, Fengguang --- hda - fix name for ALC1200 Move the more specific preset for ALC1200 above the general one for ALC888, so that it will have the chance to get matched and selected. Reported-by: Thomas Schneider <nailstudio(a)gmx.net> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -16638,9 +16638,9 @@ static struct hda_codec_preset snd_hda_p .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc883 }, + { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, {} /* terminator */ };

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[alsa-devel] [PATCH] ALSA: hda - Add HP Acacia detection
by Chris Bagwell 01 Jan '09

01 Jan '09

Add automatic mapping of HP Acacia motherboards to 3stack-hp. Allows for greater then 2 channel audio by enabling Channel Mode option in mixer. Motherboard specs: http://h10025.www1.hp.com/ewfrf/wc/document?docname=c01321559&lc=en&dlc=en&… Signed-off-by: Chris Bagwell <chris at cnpbagwell dot com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..410be84 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -8444,6 +8444,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),

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Re: [alsa-devel] 2.6.28: hda_codec: Unknown model for ALC883, trying auto-probe from BIOS...
by Takashi Iwai 01 Jan '09

01 Jan '09

At Tue, 30 Dec 2008 07:28:14 -0500 (EST), Justin Piszcz wrote: > > On Tue, 30 Dec 2008, Justin Piszcz wrote: > > > The audio for this board works fine, there are no issues, but there are some > > cosmetic issues when the driver loads: > > > > 1. Unknown model. It's no bug. It's neither warning nor error message but a information message. > > 2. Wrong model number. They likely share the same ID and compatible. So don't worry about it. In short, nothing to fix, so far. thanks, Takashi -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH 0/2] ASoC: TWL4030: Add DAPM based capture implementation
by Peter Ujfalusi 31 Dec '08

31 Dec '08

Hello, It has been a long time... Finally I have managed to get the DAPM based capture implementation working. I have not counted, but there were several dead end trials before I ended up with the current implementation. I have tested the capture implementation with the following configs: digimic0, Headset mic, AUX connected (not on the same board). So far the capture path behaved in a sane way, powering/enabling only the needed components in the capture path. All of the user controls are showing under the 'Capture' tab in alsamixer. Example for the board file maintainers to get the capture working: Let's say the board have the following input selection: Headset mic, AUXL/R (Line in), digimic0 static const struct snd_soc_dapm_widget BOARD_dapm_widgets[] = { SND_SOC_DAPM_MIC("Digital Mic", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_LINE("Line in", NULL); }; static const struct snd_soc_dapm_route audio_map[] = { {"DIGIMIC0", NULL, "Mic Bias 1"}, {"Mic Bias 1", NULL, "Digital Mic"}, {"HSMIC", NULL, "Headset Mic Bias"}, {"Headset Mic Bias", NULL, "Headset Mic"}, {"AUXL", NULL, "Line in"}, {"AUXR", NULL, "Line in"}, }; To record from the Headset mic: amixer sset 'Analog Left' 'Headset mic' # select the Headset for Analog left amixer sset 'TX1' 'Analog' # Select the Analog path for TX1 amixer sset 'TX2' 'Digimic1' # will disable the TX2 path, since digimic1 is # not connected amixer sset 'Analog' 2,2 # set the analog capture volume Have a Happy New Year! --- Peter Ujfalusi (2): ASoC: TWL4030: Make the enum filter generic for twl4030 ASoC: TWL4030: DAPM based capture implementation sound/soc/codecs/twl4030.c | 377 ++++++++++++++++++++++---------------------- sound/soc/codecs/twl4030.h | 7 + 2 files changed, 196 insertions(+), 188 deletions(-)

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[alsa-devel] Is Asoc suitable for my device?
by Xuesong Yang 31 Dec '08

31 Dec '08

hi all: I'm new to ALSA and still study it. My problem is I don't know if it suitable for my device use Asoc. In my platform, there is only a I2S bypass module, It has a route, select one of the I2S source as input,(from HDMI or cpu data) delay the data for some time and then output it to the backend. In theory, this device is very sample, but I find I can't get a easy way to implement it in ALSA, I'm lost in ALSA. Any suggestion will be greatly appreciated. Thanks.

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[alsa-devel] problems with 2 outputs on snd-hda-intel
by Alex Ander 29 Dec '08

29 Dec '08

My configuration: http://www.alsa-project.org/db/?f=e33841b3aa560b031a3723ecf2e2c5040d2c1a16 Ever since I came back from the holidays I've been having this problem: My laptop has 3 jacks: 2 line out and 1 line in that can be switched to out. But, I can only get sound over the laptop-speakers and the line-in switched to out. The speaker stops playing when a jack is connected to one of the two outs, but I don't get any sound over these outputs. I have checked all mixer-settings in alsamixer, nothing works (except switching the mic to lineout, which doesn't fix the other 2 outs) As you can see in alsa-info I have forced a model-type to the module, I have found this on http://www.mjmwired.net/kernel/Documentation/sound/alsa/ALSA-Configuration.… I have tried all 4 modules listed at line 1015 (which should correspond to the SigmaTel STAC9228) of which 3stack seems to be the one set by default (with empty /etc/modprobe.conf) and is the only one that works with the mic-as-output option. I can give more info if you need it.

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[alsa-devel] MIDI In on Audigy 2 ZS Live Drive
by Carl Bauer 28 Dec '08

28 Dec '08

On the Live Drive CD Bay thingy: MIDI in does not work. MIDI out (and everything else on the Live Drive) does work. For details, I have posted this problem before: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg23618.html http://forums.fedoraforum.org/showthread.php?t=198240 and seen it elsewhere: http://ubuntuforums.org/showthread.php?t=476633 I don't know if it's an ALSA bug or a kernel thing. My aadebug output is in the first of the above links. Or, my alsa-info.sh output is here: http://www.alsa-project.org/db/?f=5990a1a1b7476778a6cc82fff91997567db43a14 Thanks. Forgive me if this is the wrong list to post to. -Carl

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[alsa-devel] [PATCH] alsa-info.sh - added extra checks
by Travis Place 28 Dec '08

28 Dec '08

Changelog: Added checks for Pulseaudio/Esound/aRts 'sound servers', Check if they are 1) installed 2) running A few small changes for the locale, and timezone - keep things more uniform. Bumped version to 0.4.53 Signed-off-by: Travis Place <wishie(a)wishie.net>

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