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

Re: [alsa-devel] Ubuntu sound problem - http://ubuntuforums.org/showthread.php?t=1073090
by jesse fillion 14 Apr '09

14 Apr '09

Hello, I wish I was coming back to you with good news. I have tried the suggested addition to my modprobe. I added the line you gave me to the alsa-base file in my modprobe folder. I have attached it and you will notice that I took "mobile" out and replaced it with m4. Both options allow me to use audio through my headphone jack, but notice that the audio buttons on my laptop funtion properly when I have m4 instead. 1. Ubuntu 8.10 2. kernel 2.6.27-14-generic 3. Alsa 1.0.19 4. AlsaUpgrade-1.0.x-rev-1.16 I notice that when I try and update Alsa snapshot, I get errors. See alsa upgrade log. I have also included my alsa-base file, uxchecker for addtional alsa related information and the actual alsa updater script that I used. Nothing is muted from what I can see, I can hear audio through my headphone jack perfectly. Any help I could receive is truly greatly appreciated. Thank-you Date: Thu, 9 Apr 2009 21:11:03 -0400 From: geisj(a)MessageNetSystems.com To: rob.krakora(a)MessageNetSystems.com CC: jessefillion24(a)hotmail.com Subject: Re: [alsa-devel] Ubuntu sound problem - http://ubuntuforums.org/showthread.php?t=1073090 We added the following to /etc/modprobe.conf: options snd-hda-intel enable_msi=1 model=mobile Jerry Robert Krakora wrote: Jesse: My boss and I just had the same issue with the same hardware (a couple of months ago). We were able to resolve the issue. However, I am not at work right now so I will have to get back with you tomorrow on our resolution. On Thu, Apr 9, 2009 at 12:19 PM, jesse fillion <jessefillion24(a)hotmail.com> wrote: Good afternoon, My name is Jesse. Forgive me if this is not the correct address to be sending this type of request to, but I am looking for support in getting the Alsa drivers to work correctly with HP DV4 laptops. The DV4 1228ca has a Azalia RS780 soundcard. I have the ability to hear music through the headphone jack, but not the internal speakers (please see ubuntu forum for additional information). I would truly appreciate any information you could provide. Thank you. -Jesse Hi, I'm relatively new to Ubuntu and would be thrilled for any assistance anyone can provide me with this problem. I have an HP DV4 1228ca. Just like everyone, I did have sound through my front audio jack and could listen to with headphones. Unfortunately after trying all of your helpful suggestions, I don't even have audio in my front jack anymore. If I go into my sound properties, it doesnt seem to recognize my sound card anymore. I beleive this has to do with the entry in the alsa base file. Here are my results: "[code]#> ./configure --enable-dynamic-minors #> make" winblows@ubuntu:~/Desktop/AlsaDriver/alsa-driver$ make make dep make[1]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver' make[2]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' copying file alsa-kernel/core/info.c /home/winblows/Desktop/AlsaDriver/alsa-driver/utils/patch-alsa: 24: patch: not found make[2]: *** [info.c] Error 1 make[2]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' make[1]: *** [dep] Error 1 make[1]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver' make: *** [include/sndversions.h] Error 2 #> sudo make install-modules This is where my noobishness may shine through. I wasn't quite sure what to take from the make process above so I continued... winblows@ubuntu:~/Desktop/AlsaDriver/alsa-driver$ sudo make install-modules [sudo] password for winblows: find /lib/modules/2.6.27-11-generic/kernel/sound -name 'snd*.*o' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'snd*.*o.gz' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'ac97_bus.*o' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'ac97_bus.*o.gz' | xargs rm -f make[1]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' mkdir -p /lib/modules/2.6.27-11-generic/kernel/sound/acore cp snd-hrtimer.ko snd-hwdep.ko snd-page-alloc.ko snd-pcm.ko snd-rawmidi.ko snd-timer.ko snd.ko /lib/modules/2.6.27-11-generic/kernel/sound/acore cp: cannot stat `snd-hrtimer.ko': No such file or directory cp: cannot stat `snd-hwdep.ko': No such file or directory cp: cannot stat `snd-page-alloc.ko': No such file or directory cp: cannot stat `snd-pcm.ko': No such file or directory cp: cannot stat `snd-rawmidi.ko': No such file or directory cp: cannot stat `snd-timer.ko': No such file or directory cp: cannot stat `snd.ko': No such file or directory make[1]: *** [modules_install] Error 1 make[1]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' make: *** [install-modules] Error 1 #> sudo vim /etc/modprobe.d/alsa-base [add 'options snd_hda_intel model=hp-dv5' sans quotes to the bottom of the file, even if you have a dv4, etc.] I have tried each of the following, rebooting in between them: options snd_hda_intel model=hp-dv5 options snd_hda_intel model=hp-dv4 # Keep snd-pcsp from beeing loaded as first soundcard options snd-pcsp index=-2 alias snd-card-0 snd-hda-intel alias sound-slot-0 snd-hda-intel options snd-hda-intel model=dell-m4-1 options snd-hda-intel enable_msi=1 # Keep snd-pcsp from beeing loaded as first soundcard options snd-pcsp index=-2 alias snd-card-0 snd-hda-intel alias sound-slot-0 snd-hda-intel options snd-hda-intel model=hp=dv4 options snd-hda-intel enable_msi=1 I have also tried using the 3.4 compiler and "unstable" alsa snapshot package recommended by uticamarco. If anyone has any suggestions, I would be most grateful. Thank you very much. Jesse _________________________________________________________________ Internet Explorer 8 helps keep your personal info safe. http://go.microsoft.com/?linkid=9655581 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Best Regards, -- Jerry Geis MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 (317)663-0808 Fax _________________________________________________________________ Create a cool, new character for your Windows Live™ Messenger. http://go.microsoft.com/?linkid=9656621

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[alsa-devel] [PATCH - patch_sigmatel.c for HP dv4-1222nr laptop 9/9Patch to patch_sigmatel.c for HP dv4-1222nr laptop
by renidragsemaj＠yahoo.com 13 Apr '09

13 Apr '09

From: James Gardiner <renidragsemaj(a)yahoo.com> This patch to patch_sigmatel.c is for the HP dv4-1222nr laptop. It activates the included microphone(s) and the mute status change light for this computer. I don't know if this will work for any other dv4 laptop. The commit isn't perfect, for it doesn't seem to work too well with pulseaudio - the mic boost doesn't work with that audioserver. Otherwise, it works well with esound, even across reboots and suspend/hibernate/resume. The built-in microphone sits on node 0x18 and feeds through node 0x1c. The gpio for the mute light is GPIO #0. I borrowed the code from the HP HDX. Occasionally if you press the mute button too fast in succession it lowers the actual volume to 0 and you'll have to adjust it again...But if you are doing that you probably are getting a kick out of making me feel bad about the behavior. :) No autodetect of the laptop seems to work for me, but "model=hp-dv4-1222nr" as an option to modprobe will make the changes. Signed-off-by: James Gardiner <renidragsemaj(a)yahoo.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61996a2..4286155 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_M4, STAC_HP_DV5, STAC_HP_HDX, + STAC_HP_DV4_1222NR, STAC_92HD71BXX_MODELS }; @@ -1847,6 +1848,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", + [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -4534,6 +4536,28 @@ static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, } #endif +static int stac92xx_hp_dv4_1222nr_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + /* Shamelessly stolen from the function stac92xx_hp_hdx_check_power_status */ + /* by James Gardiner */ + struct sigmatel_spec *spec = codec->spec; + + if (nid == 0x10) { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + spec->gpio_data &= ~0x01; /* orange */ + else + spec->gpio_data |= 0x01; /* white */ + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, + spec->gpio_data); + } + + return 0; +} + static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; @@ -5202,6 +5226,19 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 1; break; + case STAC_HP_DV4_1222NR: /* James Gardiner */ + spec->num_dmics = 1; + spec->num_dmuxes = 2; + stac92xx_auto_set_pinctl(codec, 0x1c, 2); + stac92xx_auto_set_pinctl(codec, 0x18, AC_PINCTL_IN_EN); +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + spec->gpio_data |= 0x01; + codec->patch_ops.check_power_status = + stac92xx_hp_dv4_1222nr_check_power_status; +#endif + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); -- 1.5.6.3

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Re: [alsa-devel] no sound with the new imac
by Clemens Ladisch 13 Apr '09

13 Apr '09

Rafael J. Wysocki wrote: > Takashi, what's the appropriate list for ALSA-related things? alsa-devel (Takashi is on vacation until after the Easter weekend). > On Wednesday 08 April 2009, Justin Mattock wrote: > > Can somebody point me in the right direction > > to creating the correct pin numbers etc... > > for the new imac so I can have sound. What kernel version are you using? Are there any messages in the system log when you load the snd-hda-intel driver? What model do you mean with "the new imac"? What are the contents of /proc/asound/card0/codec#0? Best regards, Clemens

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[alsa-devel] Via vt8237 sometimes fails to load
by Manuel Naranjo 13 Apr '09

13 Apr '09

Hello guys, I have a KT880 based mobo made by MSI, I haven't used this computer for a while (a year or so) I remember that before the audio was working, even though sometimes it was initializated with noise and required me restarting alsa. Now it doesn't even load, and I get some weird errors in dmesg. I've checked the wiki and the bug tracker but still couldn't find a solution. This is lspci -vv -nn 00:11.5 Multimedia audio controller [0401]: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller [1106:3059] (rev 60) Subsystem: Micro-Star International Co., Ltd. Device [1462:0470] Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin C routed to IRQ 22 Region 0: I/O ports at a800 [size=256] Capabilities: <access denied> Kernel modules: snd-via82xx And dmesg ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:581: codec_read: codec 0 is not valid [0x1fc0000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:581: codec_read: codec 0 is not valid [0x1fe0000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:523: codec_ready: codec 0 is not ready [0x1000000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:523: codec_ready: codec 0 is not ready [0x1000000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:581: codec_read: codec 0 is not valid [0x1fc0000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:581: codec_read: codec 0 is not valid [0x1fe0000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/ac97/ac97_codec.c:2077: AC'97 0 access is not valid [0xffffffff], removing mixer. VIA 82xx Audio 0000:00:11.5: PCI INT C disabled VIA 82xx Audio: probe of 0000:00:11.5 failed with error -5 Thanks, Manuel

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[alsa-devel] [PATCH] ASoC: pxa-ssp.c fix clock/frame invert
by Daniel Ribeiro 13 Apr '09

13 Apr '09

>From the pxa27x developers manual, Table 8-8 SSPSP Bit Definitions: SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low) SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low) SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High) SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High) SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0). This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and DSP_B modes. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657..6768e4e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -568,7 +568,10 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_NB_IF: break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(3); + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; break; default: return -EINVAL; @@ -585,7 +588,13 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_NB_NF: sspsp |= SSPSP_SFRMP; break; + case SND_SOC_DAIFMT_NB_IF: + break; case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; break; default: return -EINVAL; -- tg: (577c9c4..) asoc/pxa-ssp-invert-clock.patch (depends on: master) total: 0 errors, 0 warnings, 24 lines checked asoc-pxa-ssp-invert-clock.patch has no obvious style problems and is ready for submission.

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Re: [alsa-devel] Question about pxa2xx-ac97-lib.c
by Mark Brown 12 Apr '09

12 Apr '09

On Sun, Apr 12, 2009 at 04:47:39PM +0200, Marek Vasut wrote: > One more thing, I'd like to ask how to change the reset_gpio in the above > file. Will I have to make copy of pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI] in > palm27x.c and change the .probe pointer to my own function that sets the > proper platform data? That is, something like the following? Thanks BTW, I meant to say in reply to the previous mail - you really should copy all audio discussion to at least alsa-devel. > +/* Palms use GPIO95 for AC97 reset */ > +static int palm_ac97_probe(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct platform_device *pd = to_platform_device(dai->dev); > + static struct pxa2xx_ac97_platform_data pdata = { > + .reset_gpio = 95, > + }; > + printk("%s[%i]\n", __FUNCTION__, __LINE__); > + pd->dev.platform_data = &pdata; > + return pxa2xx_ac97_hw_probe(pd); > +} Look in mainline - pxa2xx-ac97-lib already provides platform data based configuration of the GPIO to use for reset.

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Re: [alsa-devel] Question about pxa2xx-ac97-lib.c
by Robert Jarzmik 12 Apr '09

12 Apr '09

Marek Vasut <marek.vasut(a)gmail.com> writes: > One more thing, I'd like to ask how to change the reset_gpio in the above > file. Will I have to make copy of pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI] in > palm27x.c and change the .probe pointer to my own function that sets the > proper platform data? That is, something like the following? Thanks I'll add Mark to the discussion here, for the ASoC point of view. Mark, the original mail is in [1]. I originaly thought we would change pxa_set_ac97_info(), to add a parameter which will be the platform data for the gpio. IOW, have something like : void __init pxa_set_ac97_info(struct pxa2xx_ac97_platform_data *pdata) { pxa_register_device(&pxa_device_ac97, ops); } The trouble is, I saw references to the platform data as pxa2xx_audio_ops_t*, mostly to suspend and resume a card. I've seen them in pxa2xx_ac97_do_suspend(), pxa2xx_ac97_do_resume(). Mark, could you please have a glance at it, and tell us the right approach to have the struct pxa2xx_ac97_platform_data passed to pxa2xx-ac97 please ? Cheers. -- Robert [1] : > diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c > index 48a73f6..f134f95 100644 > --- a/sound/soc/pxa/palm27x.c > +++ b/sound/soc/pxa/palm27x.c > @@ -19,11 +19,13 @@ > #include <linux/gpio.h> > #include <linux/interrupt.h> > #include <linux/irq.h> > +#include <linux/platform_device.h> > > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/soc.h> > #include <sound/soc-dapm.h> > +#include <sound/pxa2xx-lib.h> > > #include <asm/mach-types.h> > #include <mach/audio.h> > @@ -168,6 +170,21 @@ static int palm27x_ac97_init(struct snd_soc_codec *codec) > return 0; > } > // this is the altered probe function > +/* Palms use GPIO95 for AC97 reset */ > +static int palm_ac97_probe(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct platform_device *pd = to_platform_device(dai->dev); > + static struct pxa2xx_ac97_platform_data pdata = { > + .reset_gpio = 95, > + }; > + printk("%s[%i]\n", __FUNCTION__, __LINE__); > + pd->dev.platform_data = &pdata; > + return pxa2xx_ac97_hw_probe(pd); > +} > + > +static struct snd_soc_dai hifi_cpu_dai; > + > static struct snd_soc_dai_link palm27x_dai[] = { > { > .name = "AC97 HiFi", > @@ -204,6 +221,8 @@ static int __init palm27x_asoc_init(void) > { > int ret; > // here it'll be copied > + pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI].probe = palm_ac97_probe; > + > if (!(machine_is_palmtx() || machine_is_palmt5() || > machine_is_palmld())) > return -ENODEV; > > > ------------------------------------------------------------------- > List admin: http://lists.arm.linux.org.uk/mailman/listinfo/linux-arm-kernel > FAQ: http://www.arm.linux.org.uk/mailinglists/faq.php > Etiquette: http://www.arm.linux.org.uk/mailinglists/etiquette.php

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[alsa-devel] [PATCH] ASoC: n810: replace BUG() with BUG_ON()
by Alexander Beregalov 12 Apr '09

12 Apr '09

Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/soc/omap/n810.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index a6d1178..e54e1c2 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -383,10 +383,9 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) - BUG(); - if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) - BUG(); + BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) || + (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);

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Re: [alsa-devel] Alsa driver compatible with 2.6.18 kernel
by reetesh chandran 11 Apr '09

11 Apr '09

Thanks a lot Lee. Which Alsa driver version is the latest stable version, or which is most widely used ? --- On Fri, 4/10/09, Lee Revell <rlrevell(a)joe-job.com> wrote: From: Lee Revell <rlrevell(a)joe-job.com> Subject: Re: [alsa-devel] Alsa driver compatible with 2.6.18 kernel To: "reetesh chandran" <reeteshchandran(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Date: Friday, April 10, 2009, 5:45 PM On Fri, Apr 10, 2009 at 12:54 PM, reetesh chandran <reeteshchandran(a)yahoo.com> wrote: > Hello all, > > I have a 2.6.18 kernel and I am using 1.0.12rc1 alsa driver. I am facing some issues and would like to get a newer alsa driver. I cant change the linux kernel version. Do you know which is the latest alsa driver which is supported on linux 2.6.18 ? How do I find that out ? The latest should work. New ALSA releases are backwards compatible with old kernels. Lee

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[alsa-devel] DELL SP2309W Monitor with Integrated microphone
by Phil Gorbett 11 Apr '09

11 Apr '09

Any luck anyone with this one yet? I too have the same issue as that posted previously (last month), and get the same results with the indicative testing. I have tried several fixes via googling - module loading and sound "card" indexing via alsa.base modprobe file and LOTS of others. Still no luck. I am running FC10 with pulse, but think it's probably a snd-usb-audio problem as the driver appears to be correctly loading and the webcam is clearly indicated in the sound prefs app and gconf/gstreamer. I have run thru the usb audio device classes, and all of the reported class values reported via lsusb -v seem to correctly correspond to possible values from the audio class parameters. I'm now stumped. I do get the following errors in the gnome sound recorder: "Could not negotiate format" followed by "Could not get/set settings from/on resource.". This jives with the error from the sound preference panel: "gconfaudiosrc ! audioconvert ! audioresample ! gconfaudiosink profile=chat: Could not get/set settings from/on resource." which does not reveal anything specific enough to track down the problem from searching the net. All other sound devices and services are performing correctly via pulseaudio. I have at one time removed p/a, but this did not correct the problem. I am running the webcam mic as device 0, but changing that to any other device number fails to solve the problem. Alsamixer shows the card "Monitor Webcam" mic, the chip "USB mixer", the recording levels at 74, the capture enabled, and the AGC and loudness parameters enabled, and although I have tried various combinations of these, none have worked. I am running version 1.0.19 of the alsa drivers, and compiled these directly with all everything enabled as well as using the rpm builds. Alsainfo has been uploaded to: http://www.alsa-project.org/db/?f=803829378929c58bcc86351c091719a12c79075e The video portion of the camera functions entirely correctly via uvc and v4l, but the audio listing from the video test app gucveiw does not list the webcam mic at all under the audio device list. The uvc dev group sent me here, naturally. Cheers, PG

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