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[alsa-devel] [PATCH] Workaround for misdetected sample rate with CM6207
by Joris van Rantwijk 16 Feb '09

16 Feb '09

Summary: Workaround for misdetected sample rate with CM6207 The CM6207 incorrectly advertises its 96 kHz playback setting as 48 kHz in its USB device descriptor. This patch extends an existing workaround in usbaudio.c to also cover the CM6207. This resolves issue 0004249 in the ALSA bug tracker. Signed-off by: Joris van Rantwijk <jorispubl(a)xs4all.nl> diff -ur linux-2.6.28.4/sound/usb/usbaudio.c linux-2.6.28.4_j/sound/usb/usbaudio.c --- linux-2.6.28.4/sound/usb/usbaudio.c 2009-02-13 21:45:17.000000000 +0100 +++ linux-2.6.28.4_j/sound/usb/usbaudio.c 2009-02-13 22:00:44.000000000 +0100 @@ -2533,19 +2533,20 @@ } fp->nr_rates = nr_rates; - fp->rate_min = fp->rate_max = combine_triple(&fmt[8]); + fp->rate_min = fp->rate_max = 0; for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) { unsigned int rate = combine_triple(&fmt[idx]); /* C-Media CM6501 mislabels its 96 kHz altsetting */ if (rate == 48000 && nr_rates == 1 && - chip->usb_id == USB_ID(0x0d8c, 0x0201) && + (chip->usb_id == USB_ID(0x0d8c, 0x0201) || + chip->usb_id == USB_ID(0x0d8c, 0x0102)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; fp->rate_table[r] = rate; nonzero_rates |= rate; - if (rate < fp->rate_min) + if (r == 0 || rate < fp->rate_min) fp->rate_min = rate; - else if (rate > fp->rate_max) + else if (r == 0 || rate > fp->rate_max) fp->rate_max = rate; fp->rates |= snd_pcm_rate_to_rate_bit(rate); }

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[alsa-devel] ALSA layer in Linux 2.6.18 and OSS
by Viral Mehta 16 Feb '09

16 Feb '09

Hi All, I am trying to make an application that does Audio playback and record. I have configured Kernel and Re-built it. I want to use only ALSA features. I have said "Yes" to all ALSA related options and "No" to all OSS related things. When I booted my kernel, I found that there was no "/dev/dsp" device and so my application was not able to run. The application opens "/dev/dsp" device first and then does other things. Next time, when I enabled few of the OSS options in kernel's config file. I was able to run my application. My question is I do not want OSS at all. I just need ALSA then in that case what I am doing wrong. Thanks, Viral -- _____________________________________________________________________ Disclaimer: This e-mail message and all attachments transmitted with it are intended solely for the use of the addressee and may contain legally privileged and confidential information. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution, copying, or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately by replying to this message and please delete it from your computer. Any views expressed in this message are those of the individual sender unless otherwise stated.Company has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. __________________________________________________________________________

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[alsa-devel] Possible bug with ICE1712 driver
by Brendan Minish 16 Feb '09

16 Feb '09

Hello the driver for my M-Audio Delta66, snd_ice1712 fails when my machine resumes from suspend. I get no audio out and software accessing the the audio device appears to either hang or work normally (but with no output) I can always get sound working again by removing and reinserting the kernel module snd_ice1712 after the machine resumes if the ice1712 driver is compiled into the kernel then sound fails after resume and there appears to be no way to get it working again Platform Linux Intel x86_64 (core 2 E6700 ) Distro Fedora Custom kernel ver 2.6.27.12 issue also observed on distro kernel -2.6.27.12-170.2.5.fc10.x86_64 Attached dmesg output for suspend, resume and reprobing of snd_ice1712 Attached lspci output Attached lsmod output Anything else I should capture to help debug this ? Is this the right place to file this ? thanks for the help regards Brendan Minish

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[alsa-devel] how to set default alsa pcm device as bluetooth, so i can play flash in firefox to bluetooth headset?
by Ted T. Logan 16 Feb '09

16 Feb '09

I did this, but no go: pcm.bluetoothstereo { type bluetooth device 00:18:E4:0C:4D:17 #headset profile "auto" } pcm.!default { type plug slave.pcm "bluetoothstereo" } I can do things like mplayer -ao alsa:device=bluetoothstereo with no problem, and I hear fine through the headset. However, I'd like to be able to play flash audio from firefox through the headset. What am i doing wrong? rpmq bluez bluez-hcidump-1.42-2.fc10.i386 bluez-libs-4.22-2.fc10.i386 bluez-4.22-2.fc10.i386 bluez-gstreamer-4.22-2.fc10.i386 bluez-gnome-analyzer-1.8-11.fc10.i386 bluez-libs-devel-4.22-2.fc10.i386 bluez-gnome-1.8-11.fc10.i386 bluez-alsa-4.22-2.fc10.i386 bluez-cups-4.22-2.fc10.i386

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[alsa-devel] [PATCH] staging: &&/|| typo
by Roel Kluin 16 Feb '09

16 Feb '09

&&/|| typo Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- // vi sound/oss/ad1848.c +83 #define MD_1845 4 ... #define MD_1845_SSCAPE 9 /* Ensoniq Soundscape PNP*/ diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 7cf9913..a5b8356 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -2107,7 +2107,7 @@ int ad1848_control(int cmd, int arg) switch (cmd) { case AD1848_SET_XTAL: /* Change clock frequency of AD1845 (only ) */ - if (devc->model != MD_1845 || devc->model != MD_1845_SSCAPE) + if (devc->model != MD_1845 && devc->model != MD_1845_SSCAPE) return -EINVAL; spin_lock_irqsave(&devc->lock,flags); ad_enter_MCE(devc);

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[alsa-devel] Bug: Mic problem, mostly noise, almost no signal, AD1988B on an Asus P5B
by Thomas Stenhaug 16 Feb '09

16 Feb '09

Current settings and info here: http://www.alsa-project.org/db/?f=021ee315857ae0928de665f1f0924cc84c609576 When I record with these settings, I get a /lot/ of static, and discernible signal when I tap the microphone on my desktop, or scream into it. The static gets better when I reduce the capture level, but the signal is still barely discernible. The microphone works in the same jack when the computer is running Windows. I've also tried a different mic, with the same results. Playback works flawlessly. I have seen some posts with people having problems that sound similar, and some of them seems to be resolved by enabling Mic bias. However, I don't have a control called that when I look at alsamixer. My system is running Debian Etch. When I started working on this problem, I used the packages from the "testing" (Etch) repository. I then pulled in alsa-base and alsa-utils from the "experimental" repository, and compiled the 1.0.18a drivers. This changed nothing in the recordings, nor did any new controls appear in alsamixer. I added "snd-hda-intel: model=6stack-dig" to /etc/modprobe.db/alsa-base, and there was no change in the visible controls. Regards, Thomas

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[alsa-devel] ctxfi/pulse audio problems and skype - only hear digital noise after a while in call
by Ted T. Logan 15 Feb '09

15 Feb '09

Help, I have x-fi live, using ctxfi driver. I had to start using pulse audio because after a recent alsa upgrade, no matter what I did, my creative x-fi would no longer show as default sound card, but was way down the list, no matter what I did. Anyway, my current problem is I use skype all the time, but after I am in a phone call for a while, when the other person talks, I just hear digital noise/garbage whenever they talk, so I have to hang up. It is causing me a lot of grief because I use skype for business. I am not telling skype to use pulseaudio even though the rest of my apps are, because it doesn't seem to work well with skype, so I just have it use alsa/ctxfi. I don't know what to do, I didn't have this problem a few months ago, and I'm not sure if it's: 1.Since I upgraded from f9 to f10 2.Since I started using pulseaudio 3.Since I switched from sbxfi driver to ctxfi My current system is f10 rpmq alsa alsa-oss-1.0.17-1.fc10.i386 alsa-oss-libs-1.0.17-1.fc10.i386 alsa-utils-1.0.19-1.fc10.i386 alsa-plugins-pulseaudio-1.0.18-2.fc10.i386 alsa-lib-1.0.19-2.fc10.i386 alsa-lib-devel-1.0.19-2.fc10.i386 alsa-tools-1.0.17-1.fc10.i386 bluez-alsa-4.22-2.fc10.i386 alsa-plugins-oss-1.0.18-2.fc10.i386 Thanks in advance!

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[alsa-devel] Applying Herton supplied patches ...
by Guy Stalnaker 14 Feb '09

14 Feb '09

herton or Takashi, How does one go about applying a patch like the ones attached to Message 6? And to which files are they applied? I assume they are meant for the alsa-driver code, but that's as much as I know (well, I know how to run ./configure, make, and make sudo :). I'd very much like to see if the patches work for my HP dv4 1225. Regards, Guy > Message: 6 > Date: Sat, 14 Feb 2009 17:13:55 -0200 > From: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> > Subject: Re: [alsa-devel] [Fwd: HP DV4 1155 SE - No Sound through > speakers...] > To: alsa-devel(a)alsa-project.org > Cc: Takashi Iwai <tiwai(a)suse.de>, Tim Barnette > <tim.barnette(a)gmail.com>, Jean-Pierre Andr? > <jean-pierre.andre(a)wanadoo.fr> > Message-ID: <200902141713.55843.herton(a)mandriva.com.br> > Content-Type: text/plain; charset="iso-8859-1" > > On Saturday 14 February 2009 08:24:15 Takashi Iwai wrote: >> > At Fri, 13 Feb 2009 14:22:00 -0500, >> > Tim Barnette wrote: >>> > > >>> > > ..And it worked! I now have sound! Can you tell me what I actually did here? >> > >> > The problem was that your BIOS didn't set the correct default pin >> > configuration. The driver doesn't know which pin should be used for >> > the speaker output, as a consequence, it's set as INPUT. >> > Now you set it OUTPUT manually by hda-verb, and (fortunately) the >> > route to that pin was alive even though it wasn't declared, you get >> > the speaker output now. >> > >> > The next step is to implement a proper pin-config to override by >> > the driver. This won't be much work, but hey, wait for a while. >> > >> > (Or, did Herton already work on that?) > > I had added before pin configs for which source came from *.ini files of IDT > windows drivers provided by HP. But I don't know if they are at all necessary, > may be just setting config of 0x0d pin should suffice for all hp-dv5 models, > as all of them have speaker on 0x0d from all reports and also looking at the > *.ini files. > > I have three patches, you can choose one of them :) , see attached. The first > one is my previous pin configs work "let's set for all cases the pin configs > based on *.ini files", the others are same smaller change/less intrusive that > instead just deals with this hp dv4 case. > >> > >> > >> > thanks, >> > >> > Takashi >> > >>> > > >>> > > On Fri, Feb 13, 2009 at 2:16 PM, Tim Barnette <tim.barnette(a)gmail.com> wrote: >>> > > >>> > > Takashi, I was able to build hda-verb and when I executed the command >>> > > below, it responded with: >>> > > nid=0xd, verb=0x707, param=0x40 >>> > > value=0x0 >>> > > >>> > > On Mon, Feb 2, 2009 at 10:25 AM, Takashi Iwai <tiwai(a)suse.de> wrote: >>> > > >>> > > At Mon, 2 Feb 2009 10:17:03 -0500, >>> > > Tim Barnette wrote: >>> > > > >>> > > > Thanks, Takashi. I made this change and rebooted. >>> > > > >>> > > > I noticed right away the mute light is no longer on... but still no >>> > > sound >>> > > > through speakers, only headphones.. >>> > > > >>> > > > /sys/module/snd_hda_intel/parameters/model >>> > > > contains >>> > > > hp-dv5,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> >>> > > > >>> > > > attached is the alsa-info.txt you requested.. >>> > > >>> > > Did you try the latest alsa-driver snapshot? >>> > > >>> > > If it still doesn't work, I guess this is a BIOS problem. >>> > > The other HP dv laptops show the widget 0x0d as the speaker while >>> > > yours is unused. >>> > > >>> > > Try to run: >>> > > hda-verb /dev/snd/hwC0D0 0x0d SET_PIN_WID 0x40 >>> > > >>> > > Takashi > > -- > []'s > Herton >

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[alsa-devel] [PATCH 0/3] ASoC Blackfin updates
by Bryan Wu 14 Feb '09

14 Feb '09

Hi Mark, Please review these patches. Is it possible to add them to 2.6.30 queue? Thanks -Bryan

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Re: [alsa-devel] [PATCH 0/4] Intel HDMI audio fixes
by Takashi Iwai 14 Feb '09

14 Feb '09

At Fri, 13 Feb 2009 09:28:22 -0800, Shane W wrote: > > On Thu, Feb 12, 2009 at 12:41:11PM +0800, Wu Fengguang wrote: > > static const char *channel_name[MAX_CHANNELS] = { > > /* 0 */ N_("Front Left"), > > /* 1 */ N_("Front Right"), > > /* 2 */ N_("Rear Left"), > > /* 3 */ N_("Rear Right"), > > /* 4 */ N_("Center"), /* wav file is "Front Center.wav" */ > > /* 5 */ N_("LFE"), /* wav file is "Rear Center.wav" */ > > /* 6 */ N_("Side Left"), > > /* 7 */ N_("Side Right"), > > > > Whereas the HDMI channel mapping selected for 6-channel playback is > > > > { .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, > > > > Here the two mappings disagree on channels other than Front Left/Right. > > > > In theory the channel mapping should be adjusted via AC_VERB_SET_HDMI_CHAN_SLOT. > > However I found that verb takes no effect for G35/G45. So I'm afraid > > there's no trivial ways we can adjust the channel mapping. > > If the driver can't reorder the channels to what ALSA > needs, is it possible to do it somewhere else like > HDA-Intel.conf and use the routing plugin somehow to > reorder things. Say for plughw:0,3 or modify the > surround51 device? > > I don't know much about this interface but it looks like > channel routing is possible here. Well, it's a good question. Right now, surround51 is used for the analog output which usually has the "right" ALSA channel mapping. So, we need to detect and switch the mapping using a plugin on the fly, and that's not so trivial. Besides, the problem arises when both analog 5.1 and HDMI 5.1 are implemented in the same "card". Since a "card" is defined by a controller, not by a codec as now, one card may have both 5.1 outputs. We might need to switch to a codec-based card approach sometime later, but I've postponed it simply because it can break so easily the compatibility with older versions... Anyway, the channel mapping issue is a missing piece. There are a few more issues, some to be implemented and some to be improved and fixed. Takashi

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