Re: [Alsa-user] [alsa-devel] alsaloop problems; ALSA streaming tutorial (resend)
Dne 24.10.2018 v 10:01 frede...@ofb.net napsal(a): > Dear Mark (cc'ing ALSA-user, ALSA-devel) > > Thank you for your tutorial > > http://www.pogo.org.uk/~mark/trx/streaming-desktop-audio.html > > I found it helpful and clearly-written (although missing a "}" brace > and I think it should be "-P" instead of "-p"...). > > I got as far as trying the 'alsaloop' example. It seems very finicky, > sometimes I am able to play a test.wav cleanly through the loopback > device, and sometimes there are a lot of crackles and I get errors > like > > playback hw:0: change avail_min=2968 > playback hw:0: change avail_min=2972 > playback hw:0: change avail_min=2976 > underrun for playback hw:0 > last write before 9.8370ms, queued 10.ms/0.ms -> missing > -0.1630ms > expected 10.ms, processing 0.1910ms, max missing 0.0443ms > last wake 0.0350ms, last check 0.0350ms, avail_min 32.ms > max buf 40.ms, pfilled 0.ms, cfilled 10.ms > job started before 0.0110ms Could you retest with the latest? I tried to fix the avail_min issue here: http://git.alsa-project.org/?p=alsa-utils.git;a=commitdiff;h=8bc1bc53d0a8b3797337bddd30cd345ba1049817 Jaroslav -- Jaroslav Kysela Linux Sound Maintainer; ALSA Project; Red Hat, Inc. ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Create account on alsa-project wiki
Dne 30.12.2016 v 15:08 Adam Ward napsal(a): > Any update on this, or is the wiki actually broken ? I sent the password to your e-mail. Jaroslav -- Jaroslav Kysela Linux Sound Maintainer; ALSA Project; Red Hat, Inc. -- Check out the vibrant tech community on one of the world's most engaging tech sites, SlashDot.org! http://sdm.link/slashdot ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] ALSA package release v1.0.26 (user space)
Hello all, I would like to notify you, that ALSA packages version 1.0.26 are available for download: ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/utils/alsa-utils-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/tools/alsa-tools-1.0.26.1.tar.bz2 ftp://ftp.alsa-project.org/pub/plugins/alsa-plugins-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/pyalsa/pyalsa-1.0.26.tar.bz2 The alsa-firmware and alsa-oss packages have no changes and they were not released this time. The changelog is available at this URL: http://www.alsa-project.org/main/index.php/Changes_v1.0.25_v1.0.26 The alsa-driver package will be released in next weeks once the driver repository with the build code and the mirrored kernel code will be reorganized to be more "GIT pull" friendly for those using the whole kernel tree as the development platform. Jaroslav -- Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc. -- Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] ALSA release version 1.0.25
Hello all, the ALSA release version 1.0.25 is available for download at http://www.alsa-project.org . All packages were updated. The full changelog can be obtained from: http://www.alsa-project.org/main/index.php/Changes_v1.0.24_v1.0.25 Have fun with it, Jaroslav -- Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc. -- Keep Your Developer Skills Current with LearnDevNow! The most comprehensive online learning library for Microsoft developers is just $99.99! Visual Studio, SharePoint, SQL - plus HTML5, CSS3, MVC3, Metro Style Apps, more. Free future releases when you subscribe now! http://p.sf.net/sfu/learndevnow-d2d ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] HDA ANalyzer
Date 17.10.2011 15:12, Sebastian Bota wrote:
> Hello everyone. I am new to this mailing list, and maybe is not the
> right place to ask.
> So my question is:
>
> I Use HDA Analyzer to tune my internal microphone. I manage to make it
> work, but now i need to make the changes permanently ( this mean, even
> after reboot )
> Can anyone help me a little bit ?
The newest HDA-Analyzer has "Exp" (export) button which generates a
python script to change the HDA codec settings from a command line.
Jaroslav
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Re: [Alsa-user] plughw versus hw
Date 21.6.2011 13:55, Pierre Habraken wrote: > On 06/20/2011 10:06 PM, alsa-user-requ...@lists.sourceforge.net wrote: >> >> Date: Mon, 20 Jun 2011 22:34:46 +0400 >> From: Vladimir Mosgalin >> Subject: Re: [Alsa-user] plughw versus hw >> To: alsa-user@lists.sourceforge.net >> Message-ID:<20110620183446.ga14...@vm10124.spb.edu> >> Content-Type: text/plain; charset=us-ascii >> >> Hi Pierre Habraken! >> >> On 2011.06.20 at 19:32:28 +0200, Pierre Habraken wrote next: >> >>> I can imagine that this is a FAQ, but I could not find a clear answer : >>> which precise difference(s) distinguish(es) plughw and hw from each other ? >>> Does plughw apply sound processing that hw does not ? >> >> plughw *might* apply simple sound processing if needed, mostly channels >> conversion and rate conversion if required. It doesn't have to apply >> processing. >> hw doesn't support such processing only works when operating strictly in >> mode that audio card support. >> >> If you have device that supports only 2 channel, 16 bit 48000 mode then >> "hw" device won't be able to playback 2/16/44100 stream, or mono stream >> for example; you'll get an error when you try. But plughw will accept >> such streams and do the conversion. However, if you use plughw and >> output 2/16/48000 stream then no conversion is needed and most likely >> plughw won't be doing any processing. >> >> Note that using both hw and plughw can lead to specific problems, so >> it's best to use "default" device unless you have very specific >> requirements. > > Hello Vladimir, > > Thank you for your reply. > > I just bought an Asus Xonar DX sound card, for sending 24bits/96KHz > stereo flac files to an external DAC. > I am using Alsa 1.0.21 on a PC running Ubuntu 10.04 with Linux kernel > 2.6.32-32. > Running aplay, I can't use hw for reading 24/96 files: > > $ aplay -D hw:0,1 Prelude.wav > Playing WAVE 'Prelude.wav' : Signed 24 bit Little Endian in 3bytes, Rate > 96000 Hz, Stereo > aplay: set_params:990: Sample format non available > Available formats: > - S16_LE > - S32_LE > $ > > Adding the switch -f S32_LE does not help: > > $ aplay -D hw:0,1 -f S32_LE Prelude.wav > Warning: format is changed to S24_3LE > Playing WAVE 'Prelude.wav' : Signed 24 bit Little Endian in 3bytes, Rate > 96000 Hz, Stereo > aplay: set_params:990: Sample format non available > Available formats: > - S16_LE > - S32_LE > $ > > If I use plughw instead of hw, it works fine: > > $ aplay -D plughw:0,1 Prelude.wav > Playing WAVE 'Prelude.wav' : Signed 24 bit Little Endian in 3bytes, Rate > 96000 Hz, Stereo > ^CAborted by signal Interrupt... > $ > > Does it mean that the 24bits stream has to be converted to 16bits before > being sent to the device and then to the DAC ? No, use 'aplay -v' to see all plugins used by plughw. Your hw supports 24-bit samples encoded in the 32-bit words. So alsa-lib does 3-byte to 4-byte conversion of 24-bit samples - audio bits are not lost. Jaroslav -- Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc. -- EditLive Enterprise is the world's most technically advanced content authoring tool. Experience the power of Track Changes, Inline Image Editing and ensure content is compliant with Accessibility Checking. http://p.sf.net/sfu/ephox-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] "_snd_pcm_hw_open not defined" - ALSA plus MPlayer on ARM[AT91SAM9260]
On Thu, 31 Mar 2011, Sergei Steshenko wrote: > On Wed, 30 Mar 2011 17:20:20 +0200 (CEST) > Jaroslav Kysela wrote: > >> On Wed, 30 Mar 2011, Dennis Borgmann wrote: >> >>> Then I still needed ALSA for mplayer to run on my embedded board. This >>> is the configure line for the alsa-lib: >>> >>> # ./configure ... >>> ... --enable-shared --enable-static ... >>> # make >> >>> From INSTALL: >> >> """"" >> Compilation of static library >> - >> >> If you would like to use the static ALSA library, you need to use these >> options for the configure script: >> >> ./configure --enable-shared=no --enable-static=yes >> >> Unfortunately, due to bug in the libtool script, the shared and static >> library cannot be built together. >> """"" >> >> Jaroslav >> > > Well, I'm building a whole bunch of libs/apps (more than 300 targets) from > sources; a lot of targets use 'libtool' and a lot of targets are built with > _both_ static and dynamic libraries in just _one_ session. > > Are you sure the latest 'libtool' has this bug ? Have you filed a bug report > against 'libtool' ? > > I am subscribed to 'libtool' bugs mailing list and I do not remember such > a bug report. If I remember correctly, the problem is that libtool uses only one target object file for both '-fPIC -DPIC' shared and static libraries. We use -DPIC to determine the static build to resolve buildin "dynamic" symbols. In other words, we need to determine the static mode at the compile time (in gcc) which libtool does not allow. Sure, it may work for other libraries which don't use the symbol tables in way as alsa-lib does. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] "_snd_pcm_hw_open not defined" - ALSA plus MPlayer on ARM[AT91SAM9260]
On Wed, 30 Mar 2011, Dennis Borgmann wrote: > Then I still needed ALSA for mplayer to run on my embedded board. This > is the configure line for the alsa-lib: > > # ./configure ... > ... --enable-shared --enable-static ... > # make >From INSTALL: """"" Compilation of static library - If you would like to use the static ALSA library, you need to use these options for the configure script: ./configure --enable-shared=no --enable-static=yes Unfortunately, due to bug in the libtool script, the shared and static library cannot be built together. """"" Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] finally, ALSA 1.0.24 is out
Hello all, after very intensive testing, I released all ALSA packages with version number 1.0.24(+) : * alsa-driver-1.0.24 * alsa-lib-1.0.24.1 * alsa-utils-1.0.24.2 * alsa-tools-1.0.24 * alsa-firmware-1.0.24 * alsa-plugins-1.0.24 * pyalsa-1.0.24 The full list of changes is here: http://www.alsa-project.org/main/index.php/Changes_v1.0.23_v1.0.24 There is a ton of work in these parts (among others): * alsa-driver - HDA driver - ASoC tree - USB driver * alsa-library - Use Case Manager * alsa-utils - alsaloop and alsaucm introduction - alsactl improvements * alsa-tools - updated hdspmixer Please, report compilation problems directly to me or to the alsa-devel mailing list. Enjoy this release, Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! Finally, a world-class log management solution at an even better price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires February 28th, so secure your free ArcSight Logger TODAY! http://p.sf.net/sfu/arcsight-sfd2d ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Info needed.
On Thu, 6 Jan 2011, Sakthi Subramanian wrote:
> Hi,
> I am using the arecord utility to record the samples.
> How to get the PCM audio samples instead of having as WAV file from ALSA.?
Use '-f raw' for arecord to get just samples without any media file
headers.
Jaroslav
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Re: [Alsa-user] how to play a 24-bit/96kHz samples on ALC1200 (snd-hda-intel) ?
On Sun, 2 Jan 2011, Paweł Sikora wrote: Slave: Direct Stream Mixing PCM hmm, why it's resampled to 48kHz? does alsa support 96/192kHz sample rates? dmix has static parameters. Reconfigure dmix to 96kHz. Put this line to your ~/.asoundrc file: defaults.pcm.dmix.!rate 96000 Or use 'plug:front' device (but you'll lose the software mixing feature). Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Learn how Oracle Real Application Clusters (RAC) One Node allows customers to consolidate database storage, standardize their database environment, and, should the need arise, upgrade to a full multi-node Oracle RAC database without downtime or disruption http://p.sf.net/sfu/oracle-sfdevnl___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Playback getting stuck
On Fri, 19 Nov 2010, Ron Cococcia wrote: > Hello, > > I've been having a playback problem recently, and have been looking at > trying to determine the cause. On my PC, audio is playing continuously, > mostly for background noise. Recently I've been trying upgrade to a > newer PC, trying to keep things as similar as possible. For some reason > audio playback gets stuck after a long time (it repeats the last half > second of audio, and the player's time counter stops increasing). I can > stop the player and restart it again, and it will work for many hours. > I have the player set to use plughw:0,0 as the output. > > One of the behaviors I see in /proc/asound/card0/pcm0p/sub0/status is > that the hw_ptr and appl_ptr are very different: > > state: RUNNING > trigger_time: 154.109259874 > tstamp : 64590.994415958 > delay : 22048 > avail : 0 > avail_max : 0 > - > hw_ptr : 1444936032 > appl_ptr: 20352 > > > When I 'cat status' a number of times, the only field I see that changes > is the tstamp. I'm curious as to whether these ptr values are > reasonable after a long playback (and if an update to > alsa-lib/alsa-driver or tweaked parameter to a module might prevent it > from getting stuck). When I restart the player, I see that the ptr > values are more reasonable (hw_ptr + delay = appl_ptr). > > The OS is Debian Lenny, with alsa-driver and alsa-lib both at 1.0.19 (a > little older, will upgrade over the weekend if that should help) > The old PC had a VIA chipset/CPU, with the VIA 8237 controller and the > ALC655 codec. > The new PC has an Atom D525 chip, with the Realtek ALC662 rev1 HDA codec. Please, try the latest ALSA snapshot. There are many improvements in the ring buffer pointers handling. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Beautiful is writing same markup. Internet Explorer 9 supports standards for HTML5, CSS3, SVG 1.1, ECMAScript5, and DOM L2 & L3. Spend less time writing and rewriting code and more time creating great experiences on the web. Be a part of the beta today http://p.sf.net/sfu/msIE9-sfdev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] How to redirect input to output in .asoun drc?
On Sat, 13 Nov 2010, r w wrote:
> Hello.
> Googling on subject, only two solutions raising : to use sound server as jack
> or
> pulseaudio, or turn on hardware switch responsible for that. Since I want to
> insert
> few ladspa plugins between in and out, jack is less or more acceptable. But
> more less
> than more.
> Anyway, first coming in mind to solve this is to make plug or directly ladspa
> slave of
> hw device . I've tried that with no result, but restarting alsa show no
> errors, so is
> such configuration meaningful? Another way looking potential, is to use file
> plugin,
> something like this:
Use the alsaloop utility (it's in the latest alsa-utils snapshot). Some
program must copy the input stream to the output stream.
Jaroslav
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Re: [Alsa-user] alsa-compile-script patches for supporting Debian and Ubuntu distro
On Wed, 3 Nov 2010, wanming.zh...@tieto.com wrote: > Sorry, the below is patch for support Debian and Ubuntu. I applied your patch (although it required manual corrections - probably tab/space issues). Please, test the recent script from www.alsa-project.org. Jaroslav ----- Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Achieve Improved Network Security with IP and DNS Reputation. Defend against bad network traffic, including botnets, malware, phishing sites, and compromised hosts - saving your company time, money, and embarrassment. Learn More! http://p.sf.net/sfu/hpdev2dev-nov ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsa-compile-script patches for supporting Debian and Ubuntu distro
On Wed, 3 Nov 2010, wanming.zh...@tieto.com wrote: Hi, I am newbie, when I run alsa-compile.sh, I got some error. Some like “Cannot install alsa-lib-devel for unsupported distribution Debian”. Then I read alsa-compile.sh, I found this shell script could not support Debian and Ubuntu. http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg26544.html, some people report this bug, but it has not been fixed. So, I changed this shell script and fixed it. The below is patch for supporting Debian and Ubuntu. Please, send your patch as attachment or better as in-line text in plain text e-mail. We cannot apply HTML patches. Thanks, Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Achieve Improved Network Security with IP and DNS Reputation. Defend against bad network traffic, including botnets, malware, phishing sites, and compromised hosts - saving your company time, money, and embarrassment. Learn More! http://p.sf.net/sfu/hpdev2dev-nov___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Native Instruments Audio Kontrol 1: Outputs 3/4 not available
On Wed, 20 Oct 2010, Tristan Strange wrote:
> Hi,
>
> I'm having some issues setting up my Native Instruments Audio Kontrol
> 1 card on Ubuntu 10.04 using snd-usb-caiaq.
>
> The first pair of output's work perfectly as do the two inputs on the
> front of the card.
>
> Unfortunately I can't get any sound out of outputs 3/4 at all.
>
> The card shows as two subdevices when I do aplay -l:
>
> card 1: AudioKontrol1 [Audio Kontrol 1], device 0: Audio Kontrol 1
> [Audio Kontrol 1]
> Subdevices: 2/2
> Subdevice #0: subdevice #0
> Subdevice #1: subdevice #1
>
> and I can play sound through outputs 1/2 with:
> aplay -D plughw:1,0 acoustic/samples/apian-sustain-a-3-11.wav
>
> because of this I guess you should be able to play out of outputs 3/4 with:
> aplay -D plughw:1,1 acoustic/samples/apian-sustain-a-3-11.wav
Subdevice is third number:
plughw:1,0,1
Jaroslav
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Re: [Alsa-user] Basic PCM Recording
On Sun, 3 Oct 2010, Paul Braman wrote: > Paul Braman wrote: >> >> I guess therein lies some of the fundamental difference with how I was >> previously thinking. I figured I could just ask the driver/device for >> some basic defaults that should work but, instead, I should *tell* it >> what basic defaults I can live with. (Set buffer size near 1s and >> period size near 125ms.) > > I've come to the conclusion that this is stupid as hell. ALSA is > fundamentally flawed if it requires me to set anything more than the > access/format/channels/rate to make basic recording work. I should be > able to ask what the period size is it uses by default for the > hardware and do reads of that size without fear of "xrun" conditions > on my otherwise-unloaded 533MHz CPU. You replied yourself. ALSA is HAL. ALSA does not and cannot determine the system latencies, CPU power and any other system parameters which can influence the processing latency. It's almost impossible. If you don't want to bother with xruns, set the DMA buffers to maximum in the driver (this parameter is configurable using procfs) and ask for these sizes in your application (set buffer size to maximum value and use period size with something like buffer_size / 4). Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] HowTo determine samplerate card and driver supports
On Thu, 30 Sep 2010, Konstantin Kletschke wrote: > Hello, > > currently I am experimenting with different audio players in putting out > different bitrates an sample sizes of music directly to the soundcards I > have plugged in. Use test functions for hw_params for every interested rate. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Basic PCM Recording
On Fri, 24 Sep 2010, Paul Braman wrote: > Well, so far no-one has been able to sufficiently answer my inquiry > about how proper default ALSA sound capture should be coded. I've done > enough research to know that my program is getting an underrun > condition on the read. Fine. > > My question remains... > > When I set the hardware parameters, who should I need to set the > period and/or buffer size/time? Isn't the default acceptable? If not, > why not and how do I code around it? > > I've seen "simple capture examples" that just set the period size to > "32" and leave everything else untouched. Is that a good default for > all soundcards? If not, how do I discover what that default should be? 32 is too small. Too many interrupts from hardware and it's usually useless. > Here's my angle ... using OSS I could ask the driver to set its own > block size for best performance and I would just ask for that size and > use it. Never any problem. ALSA doesn't seem to have that concept as > far as I can tell. So, since all I want to do is capture audio from > any soundcard I choose and record it to a file (I don't care much > about perfect latency behavior) what the heck should a generic ALSA > capture program do? Just try to set some reasonable timing values. For you case, buffer size around 1 sec and period size around 0.125sec . snd_pcm_hw_params_set_period_time_near(100) and snd_pcm_hw_params_set_buffer_time_near(125000) will help you. (That's reason to have the near like functions.). The problem is that we cannot predict the application usage. What's best settings? All settings which driver allows should work. If you use small periods, there is more processing in kernel, but the latency is smaller. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Card sequence (hardware numbering)
On Fri, 24 Sep 2010, immanuel litzroth wrote: > Hi, > Does this address your problem? > http://alsa.opensrc.org/index.php/MultipleCards Note that the simplest way is to rename the card identifier using udev or a custom script at boot and do not mangle the slot indexes: ALSA: add /sys/class/sound/card#/id (r/w) and card#/number (r/o) files http://git.alsa-project.org/?p=alsa-kernel.git;a=commit;h=9fb6198e8c574c6547fbfac0ae1eaf7894ddfdcc The usage is just simple. Everywhere where card index can be used, the card name can be used too, for example: aplay -D hw:0,0 ... might be: aplay -D hw:CARDID,0 ... Someone should update the wiki to add this preferred possibility. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Strange alsa behaviour
On Thu, 2 Sep 2010, Cyril Russo wrote: Le 02/09/2010 15:52, Jaroslav Kysela a écrit : On Thu, 2 Sep 2010, Cyril Russo wrote: Hi, I've an issue with my new Creative Audigy sound card. I'm using a Debian Squeeze (with official 2.6.32-5-amd64 kernel) system. I've done this step to ensure I'm using the latest version: sudo module-assistant auto-install alsa (which installed the driver from alsa-driver 1.0.23's package) The sound card is correctly detected and it's working, but I've an issue, in that each channel appears as a different device. So in all the software using Alsa I have to select a device and it outputs on a single stereo channel for this particular device. For example, this command lists: # aplay -l List of PLAYBACK Hardware Devices card 0: CA0106 [CA0106], device 0: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 1: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 2: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CA0106 [CA0106], device 3: ca0106 [CA0106] Subdevices: 1/1 Subdevice #0: subdevice #0 Here the /proc/asound dump: # find /proc/asound/ /proc/asound/ /proc/asound/CA0106 /proc/asound/card0 /proc/asound/card0/id /proc/asound/card0/midi0 /proc/asound/card0/iec958 /proc/asound/card0/ca0106_reg32 /proc/asound/card0/ca0106_reg16 /proc/asound/card0/ca0106_reg8 /proc/asound/card0/ca0106_regs1 /proc/asound/card0/ca0106_i2c /proc/asound/card0/ca0106_regs2 /proc/asound/card0/pcm3c /proc/asound/card0/pcm3c/sub0 /proc/asound/card0/pcm3c/sub0/prealloc_max /proc/asound/card0/pcm3c/sub0/prealloc /proc/asound/card0/pcm3c/sub0/status /proc/asound/card0/pcm3c/sub0/sw_params /proc/asound/card0/pcm3c/sub0/hw_params /proc/asound/card0/pcm3c/sub0/info /proc/asound/card0/pcm3c/info /proc/asound/card0/pcm3p /proc/asound/card0/pcm3p/sub0 /proc/asound/card0/pcm3p/sub0/prealloc_max /proc/asound/card0/pcm3p/sub0/prealloc /proc/asound/card0/pcm3p/sub0/status /proc/asound/card0/pcm3p/sub0/sw_params /proc/asound/card0/pcm3p/sub0/hw_params /proc/asound/card0/pcm3p/sub0/info /proc/asound/card0/pcm3p/info /proc/asound/card0/pcm2c /proc/asound/card0/pcm2c/sub0 /proc/asound/card0/pcm2c/sub0/prealloc_max /proc/asound/card0/pcm2c/sub0/prealloc /proc/asound/card0/pcm2c/sub0/status /proc/asound/card0/pcm2c/sub0/sw_params /proc/asound/card0/pcm2c/sub0/hw_params /proc/asound/card0/pcm2c/sub0/info /proc/asound/card0/pcm2c/info /proc/asound/card0/pcm2p /proc/asound/card0/pcm2p/sub0 /proc/asound/card0/pcm2p/sub0/prealloc_max /proc/asound/card0/pcm2p/sub0/prealloc /proc/asound/card0/pcm2p/sub0/status /proc/asound/card0/pcm2p/sub0/sw_params /proc/asound/card0/pcm2p/sub0/hw_params /proc/asound/card0/pcm2p/sub0/info /proc/asound/card0/pcm2p/info /proc/asound/card0/pcm1c /proc/asound/card0/pcm1c/sub0 /proc/asound/card0/pcm1c/sub0/prealloc_max /proc/asound/card0/pcm1c/sub0/prealloc /proc/asound/card0/pcm1c/sub0/status /proc/asound/card0/pcm1c/sub0/sw_params /proc/asound/card0/pcm1c/sub0/hw_params /proc/asound/card0/pcm1c/sub0/info /proc/asound/card0/pcm1c/info /proc/asound/card0/pcm1p /proc/asound/card0/pcm1p/sub0 /proc/asound/card0/pcm1p/sub0/prealloc_max /proc/asound/card0/pcm1p/sub0/prealloc /proc/asound/card0/pcm1p/sub0/status /proc/asound/card0/pcm1p/sub0/sw_params /proc/asound/card0/pcm1p/sub0/hw_params /proc/asound/card0/pcm1p/sub0/info /proc/asound/card0/pcm1p/info /proc/asound/card0/pcm0c /proc/asound/card0/pcm0c/sub0 /proc/asound/card0/pcm0c/sub0/prealloc_max /proc/asound/card0/pcm0c/sub0/prealloc /proc/asound/card0/pcm0c/sub0/status /proc/asound/card0/pcm0c/sub0/sw_params /proc/asound/card0/pcm0c/sub0/hw_params /proc/asound/card0/pcm0c/sub0/info /proc/asound/card0/pcm0c/info /proc/asound/card0/pcm0p /proc/asound/card0/pcm0p/sub0 /proc/asound/card0/pcm0p/sub0/prealloc_max /proc/asound/card0/pcm0p/sub0/prealloc /proc/asound/card0/pcm0p/sub0/status /proc/asound/card0/pcm0p/sub0/sw_params /proc/asound/card0/pcm0p/sub0/hw_params /proc/asound/card0/pcm0p/sub0/info /proc/asound/card0/pcm0p/info /proc/asound/pcm /proc/asound/timers /proc/asound/modules /proc/asound/cards /proc/asound/devices /proc/asound/version /proc/asound/seq /proc/asound/seq/timer /proc/asound/seq/clients /proc/asound/seq/queues /proc/asound/seq/drivers /proc/asound/oss /proc/asound/oss/sndstat /proc/asound/oss/devices # cat /proc/asound/cards 0 [CA0106 ]: CA0106 - CA0106 Audigy SE [SB0570] at 0xcf00 irq 18 # cat /proc/asound/devices 0: [ 0] : control 1:: sequencer 8: [ 0- 0]: raw midi 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 18: [ 0- 2]: digital audio playback 19: [ 0- 3]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture 26: [ 0- 2]: digital audio capture 27: [ 0- 3]: digital audio capture 33:: timer I've an empty .asoundrc (Using a more c
Re: [Alsa-user] Strange alsa behaviour
proc/asound/cards > 0 [CA0106 ]: CA0106 - CA0106 > Audigy SE [SB0570] at 0xcf00 irq 18 > > # cat /proc/asound/devices > 0: [ 0] : control > 1:: sequencer > 8: [ 0- 0]: raw midi > 16: [ 0- 0]: digital audio playback > 17: [ 0- 1]: digital audio playback > 18: [ 0- 2]: digital audio playback > 19: [ 0- 3]: digital audio playback > 24: [ 0- 0]: digital audio capture > 25: [ 0- 1]: digital audio capture > 26: [ 0- 2]: digital audio capture > 27: [ 0- 3]: digital audio capture > 33:: timer > > > I've an empty .asoundrc (Using a more complex .asoundrc I'm able to > duplicate a channel to all other channels, but it's only duplication, > it's the same sound on all channels) > As such the audio softwares usually list this while enumerating the devices: > ALSA device: hw:CA0106,0 outs=2-2 ins=2-2 rates=3 > ALSA device: hw:CA0106,1 outs=2-2 ins=2-2 rates=3 > ALSA device: hw:CA0106,2 outs=2-2 ins=2-2 rates=3 > ALSA device: hw:CA0106,3 outs=2-2 ins=2-2 rates=3 > > I want them to list a single device with 8 outputs, and not 4 devices > with 2 outputs. > I have other systems with different sound cards and it's the first time > I'm seeing this. Use surround71 device (like 'aplay -D plug:surround71'). It combines the stereo devices to a 8-channel device. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- This SF.net Dev2Dev email is sponsored by: Show off your parallel programming skills. Enter the Intel(R) Threading Challenge 2010. http://p.sf.net/sfu/intel-thread-sfd ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Input device captures output (fwd)
On Sun, 29 Aug 2010, Warren Dumortier wrote:
> I understand that buying a real card is the easiest solution, but i don't
> want to do it.Also some time ago it was running almost with no problems, the
> issue existed but it was minor, now it got worse, maybe an updated ALSA
> package caused it.
> If you would ask me it is certainly some kind of regression as two months
> ago i had no problems on Ubuntu Lucid, i will figure out if there have been
> updates.
>
> And what i meant by the audio card that fed 100%, in fact i mean that my
> sound volume is exactly the same volume as the recording volume.
Try to play with hda-analyzer and see to http://www.alsa-project.org how
you can check issues with the snd-hda-intel driver.
Jaroslav
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Re: [Alsa-user] alsa-driver 1.0.23 fails to compile with kernel 2.6.35 (archlinux)
On Fri, 20 Aug 2010, Morgan Cox wrote:
Hi.
O.K - tried that
1st without the utils/setup-alsa-kernel script
:-
make[1]: Leaving directory
`/home/morgan/src/alsa-driver-1.0.23.62.g218f25.481.ga47ea'
make -C SUBDIRS=/home/morgan/src/alsa-driver-1.0.23.62.g218f25.481.ga47ea
CPP="gcc -E" CC="gcc" modules
make: *** SUBDIRS=/home/morgan/src/alsa-driver-1.0.23.62.g218f25.481.ga47ea:
No such file or directory. Stop.
make: *** [compile] Error 2
There was a problem in the configuration script. Please, try again in
this way (only run ./configure). It should be fixed now.
Jaroslav
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Re: [Alsa-user] alsa-driver 1.0.23 fails to compile with kernel 2.6.35 (archlinux)
On Fri, 20 Aug 2010, Morgan Cox wrote:
> Hi.
>
> Thank you for your response.
>
> I have tried using the latest snapshot
> -http://www.alsa-project.org/snapshot/files/alsa-driver-1.0.23.62.g218f25.48
> 1.ga47ea.tar.bz2
>
> I tried just using
>
> ./gitcompile
Don't use gitcompile for preconfigured tarballs, use directly the
'configure' script.
Jaroslav
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Re: [Alsa-user] alsa-driver 1.0.23 fails to compile with kernel 2.6.35 (archlinux)
On Fri, 20 Aug 2010, Morgan Cox wrote:
> Hi.
>
> I did send a previous message (but can't work out how to reply to my
> post...) , I had the same issue with a custom built kernel (with desktop
> patch.).
Use latest tarball snaphost (http://www.alsa-project.org/snapshot/). I'll
release next version of the alsa-driver package probably next week.
Jaroslav
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Re: [Alsa-user] ALSA library API docs - download?
On Sat, 31 Jul 2010, John Mills wrote:
> Hello alsa-users
>
> Can anyone direct me to a download of the ALSA library API documentation?
>
> I found it online at http://www.alsa-project.org/alsa-doc/alsa-lib/ but it
> would be much
> more convenient and greppable to have a local copy.
>
> I tried, on Ubuntu, installing alsa-source and doxygen, but can't see
> where that went, how to generate the docs or even whether it's the
> kernel or library API.
cd alsa-lib
./configure
make doc
and check alsa-lib/doc/doxygen tree...
It's documentation for alsa-lib's API.
Jaroslav
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Re: [Alsa-user] ftp.alsa-project.org down?
On Sat, 29 May 2010, James Shatto wrote:
> My debian distro comes with a 2.6.26-2-686 kernel. Which has version
> 1.0.17 of alsa. I was hoping to just install the 1.0.23 version from
> alsa-project.org. But the links to download the sources don't appear
> to work. Is the ftp site down? Is there some other way to get these
> sources without extracting them from another more recent kernel?
>
> wget -c ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2
>
> --2010-05-29 16:09:25--
> ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2
> => `alsa-driver-1.0.23.tar.bz2'
> Resolving ftp.alsa-project.org... 212.20.107.51
> Connecting to ftp.alsa-project.org|212.20.107.51|:21... connected.
> Logging in as anonymous ...
> Error in server response, closing control connection.
> Retrying.
The command works for me. It seems like a local issue in your network
(perhaps a broken NAT gateway)?
Jaroslav
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Re: [Alsa-user] problem with alsa-compile-script with "ebian" distro
On Tue, 11 May 2010, David Raleigh Arnold wrote:
> On Tuesday 11 May 2010 01:50:31 Jaroslav Kysela wrote:
>> On Mon, 10 May 2010, David Raleigh Arnold wrote:
>>> I do have Debian testing, "tainted" with an nvidia driver. I have
>>> no idea where "ebian" comes from.
>>
>> There is check_distribution() function in alsa-compile.sh:
>>
>> distrib=$(lsb_release -ds 2> /dev/null | cut -d ' ' -f 1)
>> local first=${distrib:0:1}
>> if test "$first" = "\""; then
>> distrib=${distrib:1}
>> fi
>>
>> What string 'lsb_release -ds' gives in your system?
>
> d...@hydra[tue May 11](10:45:20)~$ lsb_release -ds
> Debian GNU/Linux 5.0.4 (lenny)
Please, download the alsa-compile.sh script again. On web server is
updated version with code above. The previous release expected the string
in "" - something like "openSUSE 11.1 (i586)" - so first character was
stripped.
Jaroslav
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Re: [Alsa-user] Using the 8 channel input of a Rme96 as stereo channels
On Tue, 11 May 2010, Bernhard Walle wrote:
> Hi,
>
> although I'm using ALSA for many years as it just always worked, I'm new
> to the "advanced features" of ALSA. In a project, we want to switch from
> OSS to ALSA to have better support for consumer sound cards.
>
> So the question: The Wiki article
>
> http://alsa.opensrc.org/index.php/Rme96
>
> describes how to use the 8 channel ADAT *output* as 4 stereo channels
> with dmix. My question is how to do the same with the *input*.
Just replace 'type dmix' with 'type dsnoop'.
Jaroslav
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Re: [Alsa-user] problem with alsa-compile-script with "ebian" distro
On Mon, 10 May 2010, David Raleigh Arnold wrote:
> I do have Debian testing, "tainted" with an nvidia driver. I have no
> idea where "ebian" comes from.
There is check_distribution() function in alsa-compile.sh:
distrib=$(lsb_release -ds 2> /dev/null | cut -d ' ' -f 1)
local first=${distrib:0:1}
if test "$first" = "\""; then
distrib=${distrib:1}
fi
What string 'lsb_release -ds' gives in your system?
For kernel sources, a check for kernel package should be added to
check_kernel_source() function.
Jaroslav
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Re: [Alsa-user] problem with alsa-compile-script with "ebian" distro
On Mon, 10 May 2010, Paul Menzel wrote:
Am Sonntag, den 09.05.2010, 09:02 -0400 schrieb David Raleigh Arnold:
On Saturday 08 May 2010 20:33:55 Sergei Steshenko wrote:
On Sat, 8 May 2010 19:57:42 -0400
David Raleigh Arnold wrote:
hydra:/usr/local/src/alsa# bash utils_alsa-compile.sh
Using temporary tree /tmp/alsa-compile-script.
Detected Linux distribution 'ebian 5.0.4'.
File environment.base has been created.
cd /tmp/alsa-compile-script
utils_alsa-compile.sh: line 431: git: command not found
Trying to install package 'git':
Cannot install git for unsupported distribution ebian.
Install 'git' - look for it in your package management system.
The error messages clearly indicate it.
Thanks much. It would have taken a week for it to sink in.
Just for the record. Do you have Debian installed and is »ebian« a copy
and paste bug or did the script do that wrong?
"ebian" is probably wrong. I've tried to fix the script. But for Debian
full support (package handling), we need a volunteer (bash script
programmer) to add necessary pieces to the alsa-compile.sh .
Jaroslav
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Re: [Alsa-user] The volume buttons on Lenovo Thinkpad X61S (was: Re: Console beep on Lenovo Thinkpad X61S)
On Fri, 26 Mar 2010, Lars Bjørndal wrote:
The volume buttons up, down and mute, doesn't work from within the
console. How could that be fixed? Please note that I use Fedora 12,
and I don't use X.
It's a bit different thing. The ACPI events are translated as key-press
events to the /dev/input/event* interface. So you need an application
(daemon) watching for these key-press events and modifying the volume
using the ALSA mixer interface. The GUI programs do this.
Here is a short program in python reading input events and translating
them to amixer calls:
=== cut here ===
#!/usr/bin/python
import struct
import os
# use 'evtest' program to determine right /dev/input device and code
acpievents = "/dev/input/event1"
fmt = 'iihhi'
fd = open(acpievents, "rb")
event = fd.read(16)
while event:
(time1, time2, type, code, value) = struct.unpack(fmt, event)
if type == 1:
if code == 115: # keypress VOLUMEUP
os.system("amixer -q -c 0 set PCM 10%+")
elif code == 114: # keypress VOLUMEDOWN
os.system("amixer -q -c 0 set PCM 10%-")
event = fd.read(16)
fd.close()
cut here =
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Re: [Alsa-user] Console beep on Lenovo Thinkpad X61S
On Tue, 23 Mar 2010, Jaroslav Kysela wrote:
On Tue, 23 Mar 2010, kionez wrote:
#include// created 22/03/2010 13:49
And if you blacklist (do not load) the snd-hda-intel driver module?
Then I have beep from both, console and firmware.
I confirm that behaviour, when i load the snd-hda-intel module (it
doesn't matters what version, from kernel > 2.6.29, or what beep_mode
parameter I can pass)
If the recent initialization for HDA codec is broken, there are two ways to
do more debugging:
1) determine the patch (code change) which make things worse (look for
"git bisect" in Google)
2) try to play with hda-analyzer, maybe you find the proper codec setup
to enable the firmware beep
My hints were correct. hda-analyzer is your friend (node 0x20 - index 3
(Values 6-7 in hda-analyzer controls the analog beep input). Fortunately,
I had access to T61, so the patch fixing the analog beep input for
T61/X61 is here:
http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=0bf0e5a6f304ac1bc93a80cdd68b4d91f3519eb5
You may use alsa-compile.sh script to check the recent ALSA code. You can
control the beep volume in a mixer application now, too.
Jaroslav
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Re: [Alsa-user] Console beep on Lenovo Thinkpad X61S
On Tue, 23 Mar 2010, kionez wrote:
#include// created 22/03/2010 13:49
And if you blacklist (do not load) the snd-hda-intel driver module?
Then I have beep from both, console and firmware.
I confirm that behaviour, when i load the snd-hda-intel module (it
doesn't matters what version, from kernel > 2.6.29, or what beep_mode
parameter I can pass)
If the recent initialization for HDA codec is broken, there are two ways
to do more debugging:
1) determine the patch (code change) which make things worse (look for
"git bisect" in Google)
2) try to play with hda-analyzer, maybe you find the proper codec setup
to enable the firmware beep
Jaroslav
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Re: [Alsa-user] Console beep on Lenovo Thinkpad X61S
On Mon, 22 Mar 2010, Lars Bjørndal wrote: Jaroslav Kysela writes: On Mon, 22 Mar 2010, Lars Bjørndal wrote: lars.bjorn...@broadpark.no (Lars Bjørndal) writes: Jaroslav Kysela writes: On Mon, 22 Mar 2010, Lars Bjørndal wrote: Jaroslav Kysela writes: On Mon, 22 Mar 2010, Lars Bjørndal wrote: kionez writes: #include// created 24/02/2010 11:33 It seems that your codec is not properly initialized. Try hda-analyzer or other tool to set directly HDA codec widgets and try to find which widget controls the analog beep (assuming you're using beep_mode=0 in the latest ALSA driver). i play a little with hda-analyzer and the last kernel (2.6.33-gentoo), but i can't get the beep working, if i use a kernel > 2.6.29 every analog beeps disappears (as reported) and i lose the firmware beeps (i.e.: when the power chord is detached). is there's something to know about widgets in HDA codec? i try to see if there's something beep related in every node, but i can't find anything useful. What changed between 2.6.29 and 2.6.30 that could break console and firmware beep? On some other systems, I did need to switch from pcspkr module and instead load snd-pcsp. This does not help on the Lenovo X61S machine. Is there other modules to try? Have you tried beep_mode=0 with recent drivers? Or disable SND_HDA_INPUT_BEEP with previous kernels. I'm not sure if I do things right: I use kernel 2.6.32.9-70.fc12.i686.PAE. Tried putting the following in The beep_mode option is not in this kernel. You may compile own kernel with CONFIG_SND_HDA_INPUT_BEEP disabled or try to compile latest ALSA snapshot: http://www.alsa-project.org/main/index.php/Driver_Compilation I followed the steps, and was able to compile and install the new kernel modules. I tried beep_mode=0 as well as beep_mode=1, nothing gives me console beep. Other suggestions? Replying to myself. Actually, I got console beep working. I had to do 'rmmod pcspkr', and I did need to NOT have the line 'options snd-hda-intel beep_mode=0' in the modprobe config file. The firmware beep, e.g. the beep you get when removing or inserting the power cable, doesn't work. The real problem here is that you miss the "low battery" warning beep. You use the digital beep generated by the snd-hda-intel driver in this configuration. Have you tried beep_mode=0 with reboot? Keep the pcspkr driver. It doesn't work. Neither console beep nor firmware beep. And if you blacklist (do not load) the snd-hda-intel driver module? Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Download Intel® Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Console beep on Lenovo Thinkpad X61S
On Mon, 22 Mar 2010, Lars Bjørndal wrote:
lars.bjorn...@broadpark.no (Lars Bjørndal) writes:
Jaroslav Kysela writes:
On Mon, 22 Mar 2010, Lars Bjørndal wrote:
Jaroslav Kysela writes:
On Mon, 22 Mar 2010, Lars Bjørndal wrote:
kionez writes:
#include// created 24/02/2010 11:33
It seems that your codec is not properly initialized. Try hda-analyzer
or other tool to set directly HDA codec widgets and try to find which
widget controls the analog beep (assuming you're using beep_mode=0 in
the latest ALSA driver).
i play a little with hda-analyzer and the last kernel (2.6.33-gentoo),
but i can't get the beep working, if i use a kernel > 2.6.29 every
analog beeps disappears (as reported) and i lose the firmware beeps
(i.e.: when the power chord is detached).
is there's something to know about widgets in HDA codec? i try to see if
there's something beep related in every node, but i can't find anything
useful.
What changed between 2.6.29 and 2.6.30 that could break console and
firmware beep? On some other systems, I did need to switch from pcspkr
module and instead load snd-pcsp. This does not help on the Lenovo X61S
machine. Is there other modules to try?
Have you tried beep_mode=0 with recent drivers? Or disable
SND_HDA_INPUT_BEEP with previous kernels.
I'm not sure if I do things right: I use kernel
2.6.32.9-70.fc12.i686.PAE. Tried putting the following in
The beep_mode option is not in this kernel. You may compile own kernel
with CONFIG_SND_HDA_INPUT_BEEP disabled or try to compile latest ALSA
snapshot:
http://www.alsa-project.org/main/index.php/Driver_Compilation
I followed the steps, and was able to compile and install the new kernel
modules. I tried beep_mode=0 as well as beep_mode=1, nothing gives me
console beep. Other suggestions?
Replying to myself.
Actually, I got console beep working. I had to do 'rmmod pcspkr', and I
did need to NOT have the line 'options snd-hda-intel beep_mode=0' in the
modprobe config file. The firmware beep, e.g. the beep you get when
removing or inserting the power cable, doesn't work. The real problem
here is that you miss the "low battery" warning beep.
You use the digital beep generated by the snd-hda-intel driver in this
configuration. Have you tried beep_mode=0 with reboot? Keep the
pcspkr driver.
Jaroslav
-
Jaroslav Kysela
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.--
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Re: [Alsa-user] Console beep on Lenovo Thinkpad X61S
On Mon, 22 Mar 2010, Lars Bjørndal wrote:
Jaroslav Kysela writes:
On Mon, 22 Mar 2010, Lars Bjørndal wrote:
kionez writes:
#include// created 24/02/2010 11:33
It seems that your codec is not properly initialized. Try hda-analyzer
or other tool to set directly HDA codec widgets and try to find which
widget controls the analog beep (assuming you're using beep_mode=0 in
the latest ALSA driver).
i play a little with hda-analyzer and the last kernel (2.6.33-gentoo),
but i can't get the beep working, if i use a kernel > 2.6.29 every
analog beeps disappears (as reported) and i lose the firmware beeps
(i.e.: when the power chord is detached).
is there's something to know about widgets in HDA codec? i try to see if
there's something beep related in every node, but i can't find anything
useful.
What changed between 2.6.29 and 2.6.30 that could break console and
firmware beep? On some other systems, I did need to switch from pcspkr
module and instead load snd-pcsp. This does not help on the Lenovo X61S
machine. Is there other modules to try?
Have you tried beep_mode=0 with recent drivers? Or disable
SND_HDA_INPUT_BEEP with previous kernels.
I'm not sure if I do things right: I use kernel
2.6.32.9-70.fc12.i686.PAE. Tried putting the following in
The beep_mode option is not in this kernel. You may compile own kernel
with CONFIG_SND_HDA_INPUT_BEEP disabled or try to compile latest ALSA
snapshot:
http://www.alsa-project.org/main/index.php/Driver_Compilation
Jaroslav
-
Jaroslav Kysela
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.
--
Download Intel® Parallel Studio Eval
Try the new software tools for yourself. Speed compiling, find bugs
proactively, and fine-tune applications for parallel performance.
See why Intel Parallel Studio got high marks during beta.
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Re: [Alsa-user] Console beep on Lenovo Thinkpad X61S
On Mon, 22 Mar 2010, Lars Bjørndal wrote:
kionez writes:
#include// created 24/02/2010 11:33
It seems that your codec is not properly initialized. Try hda-analyzer
or other tool to set directly HDA codec widgets and try to find which
widget controls the analog beep (assuming you're using beep_mode=0 in
the latest ALSA driver).
i play a little with hda-analyzer and the last kernel (2.6.33-gentoo),
but i can't get the beep working, if i use a kernel > 2.6.29 every
analog beeps disappears (as reported) and i lose the firmware beeps
(i.e.: when the power chord is detached).
is there's something to know about widgets in HDA codec? i try to see if
there's something beep related in every node, but i can't find anything
useful.
What changed between 2.6.29 and 2.6.30 that could break console and
firmware beep? On some other systems, I did need to switch from pcspkr
module and instead load snd-pcsp. This does not help on the Lenovo X61S
machine. Is there other modules to try?
Have you tried beep_mode=0 with recent drivers? Or disable
SND_HDA_INPUT_BEEP with previous kernels.
Jaroslav
-
Jaroslav Kysela
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.
--
Download Intel® Parallel Studio Eval
Try the new software tools for yourself. Speed compiling, find bugs
proactively, and fine-tune applications for parallel performance.
See why Intel Parallel Studio got high marks during beta.
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Re: [Alsa-user] Silence after handling xrun (sometimes) - best way to handle?
On Wed, 17 Mar 2010, John Graham wrote: Thanks for the quick reply. Application can choose between two methods: 1) enable xrun (set stop_threshold to buffer_size or less) - you are using this method now 2) disable xrun (set stop_threshold to boundary) If you disable xrun checking in the driver and if application does not update the ring buffer contents in time, the old samples from the ring buffer will be replayed (clicks). Additionaly to reduce clicks, you may force the driver to fill the ring buffer with silence - see silence_threshold). Disabling xrun will have to do for now - dirty audio is better than no audio! I had a look at silence_threshold, but the wording in the API is slightly confusing to me, I'm not sure how that and silence_size work. I read to/write from buffers of 256 frames. I set alsa's buffer size to 256, and its period size to half of that. So I figure if I want absolutely no data re-transmitted because I've not filled up the ring buffer in time, I should set silence_threshold to 0 and silence_size to the buffer size? Or should silence_threshold be the size of a period and silence_size be (buffer size - period size)? Just set silence_size to boundary. The driver will silence the played (unused) portion of the ring buffer. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Download Intel® Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Silence after handling xrun (sometimes) - best way to handle?
On Wed, 17 Mar 2010, John Graham wrote:
> Hi there,
>
> Can anyone suggest a good, clean way of handling xruns? I'm using ALSA
> for an embedded platform (TX25 (ARM i.MX25 processor) using an
> SGTL5000 - I2S audio) and sometimes my duplex audio stream goes
> completely silent after an xrun. Then, after another xrun (or a
> couple, or a few, or a lot...) audio returns.
It looks like a bug in the driver. The audio should be recovered after
each prepare() call. I would ask the author of driver what's wrong.
> At the moment, to handle an xrun I call snd_pcm_drop() and
> snd_pcm_prepare(). The only other thing I can think of to guarantee
> the audio returns is to completely stop and restart the stream, but
> I'd much rather find a better solution!
>
> If anyone could suggest how to go about this, I'd be very grateful.
Application can choose between two methods:
1) enable xrun (set stop_threshold to buffer_size or less) - you are
using this method now
2) disable xrun (set stop_threshold to boundary)
If you disable xrun checking in the driver and if application does not
update the ring buffer contents in time, the old samples from the ring
buffer will be replayed (clicks). Additionaly to reduce clicks, you may
force the driver to fill the ring buffer with silence - see
silence_threshold).
Jaroslav
-
Jaroslav Kysela
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.
--
Download Intel® Parallel Studio Eval
Try the new software tools for yourself. Speed compiling, find bugs
proactively, and fine-tune applications for parallel performance.
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Re: [Alsa-user] ALSA dsnoop and RAT (Robust Audio Tool)
On Wed, 3 Mar 2010, Ушаков Андрей wrote: Hello all! I got a RAT (Robust Audio Tool) application and try to make works many instances of it properly. There is a my configuration: http://www.alsa-project.org/db/?f=e6185615279fc6dec5bbc81336ec6d346b9d0a9d (its default) By default I can start two or more instances of RAT, but sound from microphone is ugly (have a lot of jamming and interference) in this application (in arecord and aplay sound is perfect). I waste some time and found a problem. Problem is RAT cannot work properly with dsnoop plugin. I go to the my card config /usr/share/alsa/cards/HDA-Intel.conf and just change dsnoop plugin to hw: Have you tried to set variable defaults.pcm.dmix.rate to 16000 (or to frequency which RAT uses)? Just add this line to your ~/.asoundrc: defaults.pcm.dmix.!rate 16000 Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Download Intel® Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] virtual microphone
On Wed, 24 Feb 2010, Peter Lukac wrote:
> Hello everyone,
>
> I need debug my program with alsa driver and i need something like virtual
> microphone. I would like device "microphone" where i can put sound file and
> this file will be input for microphone stream. And in application i get this
> stream from this "microphone"
Use file plugin in alsa-lib or snd-aloop kernel module as loopback.
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
Jaroslav
-
Jaroslav Kysela
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.
--
Download Intel® Parallel Studio Eval
Try the new software tools for yourself. Speed compiling, find bugs
proactively, and fine-tune applications for parallel performance.
See why Intel Parallel Studio got high marks during beta.
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Re: [Alsa-user] Console beep on Lenovo Thinkpad X61S
On Wed, 24 Feb 2010, kionez wrote:
#include// created 22/02/2010 15:39
What could be done to get console beep back?
i haven't a fix for that, but i have collect many user report about this
issue, as seen in a mail sent to this ml [1].
i don't know where i should report that issue, everywhere i reported it
nobody seems cares about it... (ml, bugreport and so on..)
It seems that your codec is not properly initialized. Try hda-analyzer or
other tool to set directly HDA codec widgets and try to find which widget
controls the analog beep (assuming you're using beep_mode=0 in the latest
ALSA driver).
For digital beep (beep_mode=1 or 2), you need to unmute 'Beep' control in
the alsamixer or any other mixer application.
Jaroslav
-
Jaroslav Kysela
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.
--
Download Intel® Parallel Studio Eval
Try the new software tools for yourself. Speed compiling, find bugs
proactively, and fine-tune applications for parallel performance.
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Re: [Alsa-user] virtual sound card definition
On Fri, 12 Feb 2010, kapetr wrote:
> **
> ?? For what is good to define PCMs in my .asoundrc, if NO apps let me
> use them ??
> **
I think that it's problem of these applications, not ALSA itself. I would
be loud and ask authors of these apps why they hardwired something.
Could you send a list these apps? Perhaps we can push authors together.
> They use direct (?) the hardware (probably e.g. with "hw:0") - so I can
> not to force them to use (by me) defined pcm (e.g. with dmix, plug,
> ...).
You may overwrite the hw device, too. But it might have some consequences
in the system.
Jaroslav
-
Jaroslav Kysela
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.
--
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Re: [Alsa-user] Absolutely no sound from Intel Realtek ALC887--ever?
On Thu, 11 Feb 2010, David Raleigh Arnold wrote: > On Thursday 11 February 2010 12:35:41 David Raleigh Arnold wrote: >> On Thursday 11 February 2010 11:59:03 Jaroslav Kysela wrote: >>> On Thu, 11 Feb 2010, David Raleigh Arnold wrote: >>>> On Thursday 11 February 2010 11:41:45 Jaroslav Kysela wrote: >>> >>> You should install these kernel modules. If you don't know how, please, >>> post this question to support for your Linux distribution. It's not an >>> ALSA problem. >> >> Of course I should. > > But what version should I install? > > The latest alsa tarball has no ALC887 in Models. > > hydra:/home/dra/system-file-storage/alsa-driver-1.0.22.1/alsa- > kernel/Documentation# cat HD-Audio-Models.txt | grep ALC887 > hydra:/home/dra/system-file-storage/alsa-driver-1.0.22.1/alsa- > kernel/Documentation# cat HD-Audio-Models.txt | grep ALC > ALC880 > ALC260 > ALC262 > ALC267/268 > ALC269 > ALC662/663/272 > dell Dell with ALC272 > dell-zm1 Dell ZM1 with ALC272 > ALC882/883/885/888/889 > 3stack-2ch-dig 3-jack with SPDIF I/O (ALC883) > alc883-6stack-dig6-jack digital with SPDIF I/O (ALC883) > intel-alc889aIntel IbexPeak with ALC889A > intel-x58Intel DX58 with ALC889 > ALC861/660 > 3stack-660 3-jack (for ALC660) > ALC861VD/660VD > 3stack-660 3-jack (for ALC660VD) > 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) > hydra:/home/dra/system-file-storage/alsa-driver-1.0.22.1/alsa-kernel/Document > > Therefore, no matter what I do, there is no alsa driver > for this card, for which "support" was added about four > versions ago. Right or wrong? It is a "supported" card, > but it is not really supported. > > Sorry I bothered you. Regards, daveA Install the ALSA driver - it will use the generic code for new hardware. You can get a valid output from alsa-info.sh script when the driver is loaded then to add better support for your hw. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- SOLARIS 10 is the OS for Data Centers - provides features such as DTrace, Predictive Self Healing and Award Winning ZFS. Get Solaris 10 NOW http://p.sf.net/sfu/solaris-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] dual pdaudio-cf cards
On Thu, 11 Feb 2010, Burris T. Ewell wrote: > > On Feb 9, 2010, at 11:08 PM, Jaroslav Kysela wrote: > >> On Tue, 9 Feb 2010, Burris T. Ewell wrote: >> >>> I'm trying to capture 4 tracks using a pair of pdaudio-cf cards. >>> Everything seems OK until I try to capture from the route device (or >>> multi device.) The red lights on the cards will come on for a >>> moment, >>> then the machine will lock up spewing "PDAUDIOCF SRAM buffer overrun >>> detected!" to the console. >> >> No idea. Does one card work? The pdaudiocf card is reset after each >> operation to default state, so the SRAM buffer should be empty >> before the record starts. The interrupt should be triggered when 1/8 >> of SRAM buffer is filled, so there should be plenty of time to move >> samples from it. >> >> The reason might be a poor I/O performance through the PCMCIA bus. > > I'm wondering if the problem might be related to the fact that the > drivers don't support shared IRQ access. There is a warning at boot It would be quite easy to change driver to support shared interrupts, but I'm not sure if hardware is capable of this. > time that the drivers weren't able to be given exclusive access to the > IRQs. However, the kernel is reporting that the different cards have > different IRQs (17 and 18.) Maybe the interrupt is coming in for one > of the cards but is being handled by the other ISR? It's not possible, Each card has own structure which is passed to the interrupt routine. Just add some debug printk lines to pdacf_interrupt() and you'll see if interrupts are handled. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- SOLARIS 10 is the OS for Data Centers - provides features such as DTrace, Predictive Self Healing and Award Winning ZFS. Get Solaris 10 NOW http://p.sf.net/sfu/solaris-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound from Intel Realtek ALC887
On Thu, 11 Feb 2010, David Raleigh Arnold wrote: > On Thursday 11 February 2010 11:41:45 Jaroslav Kysela wrote: >> On Thu, 11 Feb 2010, David Raleigh Arnold wrote: >>> For some reason there is no alsa support at present for this onboard >>> card, as of the most recent alsa sources which I could figure out how >>> to look at. I see that there was some work on it several versions ago, >>> but for a long time the ALC887 has not been present in the "Models" >>> file with all the other Intel HDA ALCxxx's. IAC, there's no sound. >>> >>> I went ahead and ran the shell script alsa/debian provided and >>> you have the output at: >>> >>> Your ALSA information is located at http://www.alsa- >>> project.org/db/?f=5c517fe25e2c5693dbee5674e0f715bd43e8b958 >>> >>> Please inform the person helping you. >>> >>> I hope this helps. Regards, daveA >> >> The mentioned output shows that no HDA hardware is initialized with the >> HDA driver. If you do commands bellow as root, do you see some messages >> related to sound in dmesg ? >> >> rmmod snd-hda-intel >> modprobe snd-hda-intel > > hydra:/# rmmod snd-hda-intel > ERROR: Module snd_hda_intel does not exist in /proc/modules > hydra:/# modprobe snd-hda-intel > WARNING: All config files need .conf: /etc/modprobe.d/nvidia-kernel-nkc, it > will > be ignored in a future release. > WARNING: Could not open '/lib/modules/2.6.32-trunk-686- > bigmem/kernel/sound/core/snd-page-alloc.ko': No such file or directory > WARNING: Could not open '/lib/modules/2.6.32-trunk-686- > bigmem/kernel/sound/core/snd.ko': No such file or directory > WARNING: Could not open '/lib/modules/2.6.32-trunk-686- > bigmem/kernel/sound/core/snd-timer.ko': No such file or directory > WARNING: Could not open '/lib/modules/2.6.32-trunk-686- > bigmem/kernel/sound/core/snd-pcm.ko': No such file or directory > WARNING: Could not open '/lib/modules/2.6.32-trunk-686- > bigmem/kernel/sound/core/snd-hwdep.ko': No such file or directory > WARNING: Could not open '/lib/modules/2.6.32-trunk-686- > bigmem/kernel/sound/pci/hda/snd-hda-codec.ko': No such file or directory > FATAL: Could not open '/lib/modules/2.6.32-trunk-686- > bigmem/kernel/sound/pci/hda/snd-hda-intel.ko': No such file or directory You should install these kernel modules. If you don't know how, please, post this question to support for your Linux distribution. It's not an ALSA problem. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- SOLARIS 10 is the OS for Data Centers - provides features such as DTrace, Predictive Self Healing and Award Winning ZFS. Get Solaris 10 NOW http://p.sf.net/sfu/solaris-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound from Intel Realtek ALC887
On Thu, 11 Feb 2010, David Raleigh Arnold wrote:
> For some reason there is no alsa support at present for this onboard card,
> as of the most recent alsa sources which I could figure out how
> to look at. I see that there was some work on it several versions ago,
> but for a long time the ALC887 has not been present in the "Models"
> file with all the other Intel HDA ALCxxx's. IAC, there's no sound.
>
> I went ahead and ran the shell script alsa/debian provided and
> you have the output at:
>
> Your ALSA information is located at http://www.alsa-
> project.org/db/?f=5c517fe25e2c5693dbee5674e0f715bd43e8b958
>
> Please inform the person helping you.
>
> I hope this helps. Regards, daveA
The mentioned output shows that no HDA hardware is initialized with the
HDA driver. If you do commands bellow as root, do you see some messages
related to sound in dmesg ?
rmmod snd-hda-intel
modprobe snd-hda-intel
Jaroslav
-
Jaroslav Kysela
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.
--
SOLARIS 10 is the OS for Data Centers - provides features such as DTrace,
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Re: [Alsa-user] dual pdaudio-cf cards
On Tue, 9 Feb 2010, Burris T. Ewell wrote:
> I'm trying to capture 4 tracks using a pair of pdaudio-cf cards. The
> drivers load and the cards are shown in /proc/asound on IRQs 17 and
> 18. I have a .asoundrc setup as in the example on TwoCardsAsOne on
> the wiki. The cards are being fed from a pair of clock-synced ADCs.
>
> Everything seems OK until I try to capture from the route device (or
> multi device.) The red lights on the cards will come on for a moment,
> then the machine will lock up spewing "PDAUDIOCF SRAM buffer overrun
> detected!" to the console. Looking in the pdaudiocf driver, the
> interrupt handler says that this condition "should never happen."
>
> stat = inw(chip->port + PDAUDIOCF_REG_ISR);
> if (stat & (PDAUDIOCF_IRQLVL|PDAUDIOCF_IRQOVR)) {
> if (stat & PDAUDIOCF_IRQOVR)/* should never happen */
> snd_printk(KERN_ERR "PDAUDIOCF SRAM buffer overrun
> detected!\n");
> if (chip->pcm_substream)
> tasklet_schedule(&chip->tq);
> if (!(stat & PDAUDIOCF_IRQAKM))
> stat |= PDAUDIOCF_IRQAKM; /* check rate */
> }
>
>
>
> Any ideas on what is wrong or how to fix this? Perhaps the cards need
> to be reset or the buffers drained before starting capture? I'm happy
> to hack the driver and submit a patch you tell me what I need to do to
> fix it. Any pointers would be appreciated
No idea. Does one card work? The pdaudiocf card is reset after each
operation to default state, so the SRAM buffer should be empty before
the record starts. The interrupt should be triggered when 1/8 of SRAM
buffer is filled, so there should be plenty of time to move samples from
it.
The reason might be a poor I/O performance through the PCMCIA bus.
Just put some printk lines to the interrupt handler showing timestamps
(for example use jiffies variable) in the pdacf_interrupt() and
pdacf_tasklet() routines and you'll see the timing. The tasklet routine
reads and removes samples from the SRAM buffer.
Jaroslav
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Re: [Alsa-user] how do you edit hda pins and nodes?
On Mon, 8 Feb 2010, Robert Persson wrote:
> Hello,
>
> I have a Thinkpad R61i with the Conexant CX20549 HDA codec. A lot of the
> controls are wrongly wired, with showstopping consequences, such as not
> being able to change the mic capture level, resulting in intolerable
> distortion.
>
> To get round this I rolled my own kernel with hwdep enabled and used
> hda-analyzer to tweak what had to be tweaked. Unfortunately hda-analyzer
> would always download its own latest version from git, and right now the
> version in git is broken. I have tried a snapshot of an older version, but
> that doesn't work either. Therefore I am going to need to do this manually.
Would be better to send me error messages from hda-analyzer to get things
fixed? Also send your 'alsa-info.sh' information.
Thanks,
Jaroslav
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Re: [Alsa-user] snd-hda-intel / Ibex Peak chipset sound device not found...
On Thu, 4 Feb 2010, Mark Knecht wrote:
> Are any Alsa devs reading this list or do I need to post this elsewhere?
>
> I've got a new Intel motherboard that was just recently released which
> uses the new H55 chipset. The spec sheet says it's Intel HDA but I
> don't think Alsa or the kernel is recognizing it, at least not
> automatically. The kernel is 2.6.32-gentoo-r2 for now.
> So I suspect the PCI ID is 8086:3b56.
We have already this PCI ID in the snd-hda-intel driver:
+ /* PCH */
+ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH },
But it was added recently, so I guess, that your kernel does not support
it.
Jaroslav
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Re: [Alsa-user] playing / recording mono on a stereo-only hardware
On Sat, 30 Jan 2010, Guennadi Liakhovetski wrote:
> On Sat, 30 Jan 2010, Jaroslav Kysela wrote:
>
>> Could you also try this sequence?
>>
>> # export LIBASOUND_COMPAT=1
>> # aplay -v -Dplughw:0 /home/lyakh/c1.wav
>
> Same:
Ok. The next try might be to enable the CHOOSE_DEBUG debug output in
alsa-lib/src/pcm/pcm_params.c . Please, put this output to
http://pastebin.ca .
Thanks,
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Re: [Alsa-user] playing / recording mono on a stereo-only hardware
On Sat, 30 Jan 2010, Guennadi Liakhovetski wrote: > On Sat, 30 Jan 2010, Jaroslav Kysela wrote: > >> On Sat, 30 Jan 2010, Guennadi Liakhovetski wrote: >> >>> On Sat, 30 Jan 2010, Giuliano Pochini wrote: >>> >>>> On Sat, 30 Jan 2010 16:59:48 +0100 (CET) >>>> Guennadi Liakhovetski wrote: >>>> >>>>> On Sat, 30 Jan 2010, Giuliano Pochini wrote: >>>>> >>>>>> On Fri, 29 Jan 2010 11:44:43 +0100 (CET) >>>>>> Guennadi Liakhovetski wrote: >>>>>> >>>>>>>> Try something simpler: >>>>>>>> >>>>>>>> arecord -v -Dplughw:0 -c1 -r44100 -fS16_LE out.wav >>>>>>>> >>>>>>>> It prints the given parameters and, if the format is not supported >>>>>>>> by the >>>>>>>> card, it also prints the choosen settings. In that case audio data >>>>>>>> is >>>>>>>> transparently converted to the format you requested. >>>>>>>> You shouldn't have to specify plughw: because it's the default. >>>>>>> >>>>>>> Unfortunately, didn't work: >>>>>> >>>>>> Did you try with "-Dplughw:0" ? >>>>> >>>>> Yes, I did, doesn't help. >>>> >>>> Does the driver uses rules to set hw params constraints ? If so, there >>>> may >>>> be an error there. If -for example- rule_format_given_channels() and >>>> rule_channels_given_format() are not perfectly symmetrical then the plughw >>>> plugin does not work. >>> >>> This is an ASoC driver (sound/soc), here's the code: >> >> I don't see any issues here. Post 'aplay -v -Dplughw:0' log. > > I presume, with a mono wav file. Without any .asoundrs or > /etc/asound.conf: > > aplay -v -Dplughw:0 /home/lyakh/c1.wav > Playing WAVE '/home/lyakh/c1.wav' : Signed 16 bit Little Endian, Rate 8000 > Hz, Mono Could you also try this sequence? # export LIBASOUND_COMPAT=1 # aplay -v -Dplughw:0 /home/lyakh/c1.wav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- The Planet: dedicated and managed hosting, cloud storage, colocation Stay online with enterprise data centers and the best network in the business Choose flexible plans and management services without long-term contracts Personal 24x7 support from experience hosting pros just a phone call away. http://p.sf.net/sfu/theplanet-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] playing / recording mono on a stereo-only hardware
On Sat, 30 Jan 2010, Guennadi Liakhovetski wrote: > On Sat, 30 Jan 2010, Giuliano Pochini wrote: > >> On Sat, 30 Jan 2010 16:59:48 +0100 (CET) >> Guennadi Liakhovetski wrote: >> >>> On Sat, 30 Jan 2010, Giuliano Pochini wrote: >>> >>>> On Fri, 29 Jan 2010 11:44:43 +0100 (CET) >>>> Guennadi Liakhovetski wrote: >>>> >>>>>> Try something simpler: >>>>>> >>>>>> arecord -v -Dplughw:0 -c1 -r44100 -fS16_LE out.wav >>>>>> >>>>>> It prints the given parameters and, if the format is not supported by the >>>>>> card, it also prints the choosen settings. In that case audio data is >>>>>> transparently converted to the format you requested. >>>>>> You shouldn't have to specify plughw: because it's the default. >>>>> >>>>> Unfortunately, didn't work: >>>> >>>> Did you try with "-Dplughw:0" ? >>> >>> Yes, I did, doesn't help. >> >> Does the driver uses rules to set hw params constraints ? If so, there may >> be an error there. If -for example- rule_format_given_channels() and >> rule_channels_given_format() are not perfectly symmetrical then the plughw >> plugin does not work. > > This is an ASoC driver (sound/soc), here's the code: I don't see any issues here. Post 'aplay -v -Dplughw:0' log. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- The Planet: dedicated and managed hosting, cloud storage, colocation Stay online with enterprise data centers and the best network in the business Choose flexible plans and management services without long-term contracts Personal 24x7 support from experience hosting pros just a phone call away. http://p.sf.net/sfu/theplanet-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Identical USB cards to fixed card numbers
On Thu, 21 Jan 2010, Diego Tognola wrote:
> Thanks Clemens,
>
> I tried /dev/snd as well but maybe my 'test' is invalid ?
>
> My expectations was that if I have rules for USB1-> card1, USB2->card2
> and USB3->card3 setup, then reboot with the audio card on USB2
> unplugged, then I should not see a card2 in /dev/snd or cat
> /proc/asound/cards ? What I see, however, is a card1 and card2, no
> card3.
You should base your rules on USB serial numbers (hopefully the hardware
vendor created unique serial numbers, otherwise you have no proper way to
determine hw). You may also use the full hardware path (USB bus numbers
and USB port numbers).
Jaroslav
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Re: [Alsa-user] envy24control peaks error
On Sun, 27 Dec 2009, Lincoln A. Baxter wrote:
> On Nov 21, Aaron Brick wrote the following to this list:
>
> https://sourceforge.net/mailarchive/message.php?msg_name=bc54d50f0911201706q464e9275t25b02fce8e989...@mail.gmail.com
>
> This weekend I upgraded from an AMD dualcore motherboard to an Intel
> Core2 Quad...
>
> And I upgraded my kernel from 2.6.27 to 2.6.32.
>
> And now I am getting exactly the same problem. My strace show the
> following 4 relevant lines:
>
> ioctl(4, USBDEVFS_IOCTL, 0x8571b80) = -1 ENOENT (No such file or
> directory)
> fstat64(1, {st_mode=S_IFREG|0644, st_size=95720, ...}) = 0
> mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
> 0xb693f000
> write(1, "Unable to read peaks: No such fil"..., 48) = 48
>
> Tracing through the envy24control program, it is clear the problem is
> not in envy24, but ... it seems, in alsa itself, or the ice1712 driver
> (probably the later)... However, I suspect a kernel configuration might
> fix the problem but I have not been able to find it.
Use latest envy24control from alsa-tools package.
Jaroslav
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Re: [Alsa-user] ESI Juli@ - 24/96 - SPDIF
On Fri, 11 Dec 2009, Peter Vrabec wrote: > Hi all, > > please can you help me. I can't play 24bit samples on ESI Juli@ thru optical > out. > > driver: snd_ice1724 > alsa version: I need to find out when I get home. But something in Fedora11. I > guess 1.0.21. > > $ aplay -D plughw:1,1 01.Desafinado.wav > Playing WAVE '01.Desafinado.wav' : Signed 24 bit Little Endian in 3bytes, > Rate 96000 Hz, Stereo > > $cat /proc/asound/card1/pcm1p/sub0/hw_params > access: MMAP_INTERLEAVED > format: S32_LE > subformat: STD > channels: 2 > rate: 96000 (96000/1) > period_size: 2048 > buffer_size: 8192 > > result: no sound > Note I didn't get any error/warning. > > same results with > speaker-test -c2 --device plughw:1,1 --rate 96000 --format S24_LE > > but I can hear some noise with: > speaker-test -c2 --device plughw:1,1 --rate 44100 --format S16_LE > speaker-test -c2 --device plughw:1,1 --rate 96000 --format S32_LE > I guess noise is expected (OK), isn't? > > My goal is to play bit perfect 16/44.1 and 24/96 via optical out into my DAC > Styleaudio Topaz. You hit probably two bugs. I was able to reproduce the speaker-test problem - the S24_LE format is not supported in this tool (so you got silence). This tool didn't check the supported formats. I added appropriate checks to the ALSA GIT repository: http://git.alsa-project.org/?p=alsa-utils.git;a=commitdiff;h=075becdb1ace743149eae9a32bc48f22d027092f Unfortunately, I don't know the reason, why aplay does not play your wav file. I have one S24_LE file here and it is played on 96kHz/24-bit hardware without problems. Could I download your .wav file somewhere for tests? Or you may try to test another player. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Return on Information: Google Enterprise Search pays you back Get the facts. http://p.sf.net/sfu/google-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] pcm name
On Thu, 10 Dec 2009, Dino Puller wrote:
> Hi all,
> is it possible to show my pcm into list shown by aplay -L ?
>
> Ex.
>
> pcm.ear{
>type plug
>slave.pcm "hw:1,1"
> }
pcm.ear {
type plug
slave.pcm "hw:1,1"
hint {
description "Earphones"
device 1
}
}
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Re: [Alsa-user] Delta66 xruns and asound.conf buffer size
On Tue, 24 Nov 2009, Chuck Hallenbeck wrote:
> Hi list,
>
> My colleague Curtis and I are still unable to capture audio from a Delta66
> for streaming through icecast without unacceptable dropouts of samples,
> resulting in choppy sound and speech difficult to follow. Clemens Lavisch
> called our attention to a too small buffer size in our asound.conf file,
> where the capture devices are defined, but every effort to increase it
> reveals that it is capped at 5461 bytes. We can lower it, but cannot
> increase it. There are examples on the net everywhere of higher values,
> but our efforts cannot increase it beyond 5461. If anyone can suggest
> where we might look for a solution to this problem, We would be grateful.
The value 5461 comes from this formula:
262144 (256KB audio buffer) / (12 (channels) * 4 (bytes per sample))
The hardware you're using can handle only 256KB ring buffer giving approx.
247msec audio buffer at 22050Hz rate. This time should be enough.
I think that your system is not tuned in respect of real-time
responses. Check process-scheduler settings, disk I/O usage and
run all audio tasks with highest realtime priority. Ideally, the machine
should not do any other tasks.
Jaroslav
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Re: [Alsa-user] mixing dmix with plug
On Sat, 10 Oct 2009, Mikhail Ramendik wrote:
> Hello,
>
> I am trying to make an .asoundrc that would give me iec958 output on
> my system while also supporting dmix.
>
> Without dmix, I get iec958 output with the following .asoundrc :
>
> pcm.!default {
>type plug
>slave.pcm iec958
> }
>
> But I do need dmix, and I can't make it work. Try 1:
>
> pcm.!default {
>type dmix
>slave.pcm iec958
> }
Try:
pcm.!default {
type plug
slave.pcm "dmix:iec958"
}
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Re: [Alsa-user] hda-analyzer_run.py error : `GDK_IS_SCREEN (screen)' failed
On Tue, 8 Sep 2009, TsanChung Wong wrote: I have errors running the python script on kbuntu 8.04, at http://git.alsa-project.org/?p=alsa.git ... py;hb=HEAD Is the python script run only on GNOME? Is a KDE version available? Where? $ python Python 2.5.2 (r252:60911, Jul 31 2008, 17:28:52) [GCC 4.2.3 (Ubuntu 4.2.3-2ubuntu7)] on linux2 # hda-analyzer_run.py /var/lib/python-support/python2.5/gtk-2.0/gtk/__init__.py:72: GtkWarning: could not open display Your DISPLAY variable probably does not exist for your root session or you have an X authentication problem. It seems that all necessary libraries are installed. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Higher quality dmix resampling
On Tue, 16 Jun 2009, Grant wrote:
I get a lot of static-like noise when dmix resamples audio with a USB
DAC I'm trying out. When I have mpd resample internally with
libsamplerate, it sounds perfect. Can I get dmix to use libsamplerate
or disable dmix? One of the main programs I use with audio is miro
which doesn't allow you to specify an audio ouput device.
I tried this in /etc/asound.conf to disable dmix:
pcm.!default {
type hw
card 0
}
ctl.!default {
type hw
card 0
}
and tested with mplayer but that yielded no sound at all. I also tried this:
defaults.pcm.rate_converter "samplerate"
but that also yielded no sound from mplayer. Can I get around this static?
- Grant
The following patch completely fixes this problem.
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4577
Does anyone know when this patch might show up in alsa-lib?
I applied this patch. It will be in next alsa-lib release.
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Re: [Alsa-user] threshold not applied
On Wed, 3 Jun 2009, Dennis Borgmann wrote:
> Hello list-users!
>
> I am trying to set a software-parameter with this command:
>
> snd_pcm_sw_params_set_start_threshold
>
> This is working fine with a machine, that I got standing in my own flat.
> But as soon as I use it with a PC from a friend, this parameter is not
> applied (I can see this from time measurements while doing snd_pcm_readi).
For capture, start_threshold just starts capture when you request given
amount of samples. Nothing else, so you may expect to read less samples
than in start_threshold. You probably want to set avail_min parameter.
Jaroslav
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Re: [Alsa-user] query alsa for supported sample rates and formats?
On Mon, 11 May 2009, Matt Garman wrote: > On Fri, Apr 17, 2009 at 10:14:16AM +0200, Clemens Ladisch wrote: >> Matt Garman wrote: >>> Is there a way to query alsa to see what sample rates and >>> formats the sound hardware natively supports? >> >> Try the attached program. > > I get the following output when I run this for my M-Audio Audiophile > 24/96. No surprises: what led me to asking this question was my > observation that I couldn't play CD audio (S16_LE) using hw, and > have to use dmix or plughw. > > Device: hw (type: HW) > Access types: MMAP_INTERLEAVED RW_INTERLEAVED > Formats: S32_LE > Channels: 10 > Sample rates: 8000 11025 16000 22050 32000 44100 48000 64000 88200 > 96000 > Interrupt interval: 20-409500 us > Buffer size: 20-819125 us > > Now, before Clemens was nice enough to post that program, I posed > the same question on the M-Audio.com forums: > >http://forums.m-audio.com/showthread.php?p=38192 > > The response seems to suggest that the card should support more > formats. Or perhaps I'm misreading the responses (or the tech that > answered the question doesn't really understand my question). > > Thoughts? The hw supports the 32-bit format natively only. All other formats must be converted to this format in applications or alsa-lib (using plughw / plug / linear plugins). Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your production scanning environment may not be a perfect world - but thanks to Kodak, there's a perfect scanner to get the job done! With the NEW KODAK i700 Series Scanner you'll get full speed at 300 dpi even with all image processing features enabled. http://p.sf.net/sfu/kodak-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] How to reorder channels?
On Fri, 13 Feb 2009, Roman Odaisky wrote: > I merely need to reorder channels. dshare does that but disables mixing. How > can I just tell ALSA to route channels in a way slightly different from the > default one? Use dmix instead of dshare if you like also mix multiple sources. Jaroslav - Jaroslav Kysela Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA and JavaSound
On Mon, 17 Nov 2008, Lars Schnoor wrote: > I have to admit that I am a bit surprised about ALSA. I have a good > sound card (M-Audio Delta 1010LT) which should do hardware mixing, but It's not true. Envy based cards do not support hardware mixing on PCM streams from applications. They support only digital matrix mixer for all inputs / outputs. You may try to use ALSA dmix devices (software mixing in alsa-lib). Jaroslav ----- Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] ALSA 1.0.18 final packages are available
Hi all, ALSA 1.0.18 final packages (driver, lib, utils, tools, plugins) are available for download at http://www.alsa-project.org . The list of changes is on these links: http://www.alsa-project.org/main/index.php/Changes_v1.0.18rc3_v1.0.18 http://www.alsa-project.org/main/index.php/Changes_v1.0.17_v1.0.18 Have fun, Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] BugTracker
On Sat, 16 Aug 2008, Wesley Johnson wrote:
> Can the ALSA web page managers please indicate on their pages which
> ones are for development team members only and which are for general users.
>
> I followed a link that said "I found a bug" and it led to the
> BugTracker page that wanted me to login. It had a button to create a
> new account, so I tried to create an account. I would never send me any
> confirmation of the account nor would it explicitly refuse.
The log from server:
Aug 13 01:30:01 alsa0 postfix/pipe[6443]: 4776A2451E:
to=<[EMAIL PROTECTED]>, relay=spamfilter, delay=0.58,
delays=0.11/0/0/0.47, dsn=2.0.0, status=sent (delivered via spamfilter
service)
Aug 13 01:30:11 alsa0 postfix/smtp[6523]: D4DA12451F:
to=<[EMAIL PROTECTED]>, relay=mx2.usfamily.net[64.131.63.4]:25,
delay=10, delays=0.41/0/8.8/1.1, dsn=2.0.0, status=sent (250 Ok)
Aug 13 01:44:14 alsa0 postfix/smtp[6729]: F1DE524525:
to=<[EMAIL PROTECTED]>, relay=mx1.usfamily.net[64.131.63.3]:25,
delay=3.9, delays=0.1/0/3/0.8, dsn=2.0.0, status=sent (250 Ok)
Aug 14 22:00:04 alsa0 postfix/pipe[8393]: C25BD10385B:
to=<[EMAIL PROTECTED]>, relay=spamfilter, delay=2.7,
delays=0.21/0/0/2.5, dsn=2.0.0, status=sent (delivered via spamfilter
service)
Aug 14 22:00:08 alsa0 postfix/smtp[8803]: 72500103863:
to=<[EMAIL PROTECTED]>, relay=mx2.usfamily.net[64.131.63.4]:25,
delay=6.6, delays=2.5/0/3.5/0.56, dsn=4.0.0, status=deferred (host
mx2.usfamily.net[64.131.63.4] said: 451 Resources temporarily unavailable
(in reply to DATA command))
Aug 14 22:17:14 alsa0 postfix/smtp[9875]: 72500103863:
to=<[EMAIL PROTECTED]>, relay=mx1.usfamily.net[64.131.63.3]:25,
delay=1032, delays=1030/0.17/0.71/0.96, dsn=2.0.0, status=sent (250 Ok)
Blame your e-mail provider. Our server sent e-mails correctly to
SMTP host for your domain. If you have some spam filter, check if the
e-mail from our server is not in your spam folder.
Jaroslav
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Re: [Alsa-user] Difference between index=2, and index=-2?
On Thu, 22 May 2008, Nigel Henry wrote:
> When I set index options if cards are ordered incorrectly, I set these as:
> index=0
> index=1
> index=2 , etc
>
> On my Debian installs, including Kubuntu, /etc/modprobe.d/alsa-base, at the
> bottom of the file are options for abnormal drivers that try to grab card0.
>
> All of these drivers are set as index=-2
>
> I looked in the man page for modprobe.conf, but there is no info about the
> values for index options.
>
> What exactly is the difference between index=2, and index=-2?
>
> Just an academic question helping me to be a bit more clued up.
-2 means bitmask which "slots" (indexes) can be used. It's 0xfffe in
hexa, thus only first bit is zero (disable). It means, allocate any free
slot (index) except from slot (index) 0.
Jaroslav
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Re: [Alsa-user] One process per audio channel
On Mon, 10 Mar 2008, John Sigler wrote:
> Hello everyone,
>
> I have an RME AES-32 PCI board which provides 4 stereo input channels
> and 4 stereo output channels.
>
> (I'm using the hsdpm driver at this time.)
>
> I want to use one process per channel, i.e. process A handles stereo
> input #1 (on the XLR connector #1), process B handles stereo input #2,
> etc.
>
> The processes are independent, in that process A might be started, and
> only several hours later, process B is started, then a few hours later
> process A is killed and restarted.
>
> Is it possible to do that with the ALSA library?
Yes, look for the dsnoop plugin configuration in alsa-lib.
Jaroslav
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Re: [Alsa-user] ALSA (linux2.6) support for ARM platform?
On Fri, 4 Jan 2008, Sarick Jiang wrote:
> Hi folks,
>
> >From the technical report "What's new in Linux 2.6",
> http://free-electrons.com/doc/linux26.pdf
>
> It says:
> ALSA replaced OSS.
> However, ALSA still incompatible with the ARM platform. Made
> x86 architecture related assumption that conflicts with ARM caching
> mechanisms. Plain audio playback can still be achieved though.
>
> So will there be a problem if we are going to use ALSA on ARM platform?
> And if this is a problem, are there any solutions for the ARM platform to
> run Linux ALSA?
The ARM notice is not technically very correct. The ARM platform is a bit
different, of course. But the current ALSA interface can handle it as well
and we know that ARM platforms are supported without major problems.
Jaroslav
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Re: [Alsa-user] Microphone input monitor
On Mon, 19 Nov 2007, Andy Brown wrote:
> Hi All,
> Bit of an odd one, so please bear with me here!
> What I'm trying to do is write a bash script that can read the current
> volume level coming from the microphone and then deal with certain
> values (silence or noise detection).
> Does anyone know of a way of 'grabbing' the current microphone level via
> the shell?
> I'm playing around with ecasignalview as that samples my alsa sound
> device and siaplys a peak/average peak, however this is a
> user-interactive program, can anyone suggest how I can query something
> similar?
>
> Thanks in advance, and apologies if this is not quite the right place to
> post this query!
Hi, you may try to play with output of arecord utility. Example is here:
arecord -vvv -d 2 /dev/shm/a.wav
Jaroslav
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Re: [Alsa-user] Addressing ALSA's Configuration Complexity
On Sun, 21 Oct 2007, T.P. Reitzel wrote: > How long has ALSA been in existence now? The documentation for ALSA is > still incomplete and excessively complex for the average user trying to > adapt it to his needs. Here's a rather common scenario. A user just > purchases a new computer. For the sake of argument, the driver for the > audio chipset on his computer is immature so the user sticks an older > audio card into his computer. ALSA sees the on-board chipset as card0 > and the PCI card as card1. Now, how does this chap configure ALSA to > FULLY utilize the myriad number of applications using either ALSA or > OSS? Common scenarios have some users searching for YEARS to find the > proper solution to their problems of configuring ALSA. Say, this chap > downloads the speech recognition program, Julius/Julian (OSS only), and > wants to input audio from a microphone attached to card0. How does he > do it? Next, say this chap has VMware and needs to ouput audio (OSS > again) on card1 instead of the default card, 0. How does he do it? > Moving alone, now this chap wants to play a DVD in 5.1 surround sound on > card1 due to the immaturity of ALSA's driver for card0. How does he do > it? Now, the chap is listening to CD audio and wants to route the > playback to all 5 speakers instead of the two in front? How does he do > it? Generally, this chap wants to playback audio on card1 and record on > card0, and do it all in this lifetime. If the sarcasm is showing a bit > strongly, maybe ALSA's developers at SUSE should strongly consider that > this chap isn't an engineer. The aforementioned scenario is COMMON, and > I see nothing to address this scenario on the WIKI pages or ALSA's > website. Serious consideration of these issues need to addressed at SUSE > immediately as this time consuming process of configuring ALSA has gone > on far too long already. Yes, let the flames begin, but buckets of > water have already been thrown by myself and numerous other users to > more than quench the fire of emotion. Personally, I like ALSA, but > hiding its complexity for the average user should be a priority. The > time for talk is over and has been so for a long time. The system configuration complexity should be handled inside distribution, not subsystem itself (we have already clear soundcard/device naming, but it's true that users must learn about it). So ask distributions to add such possibility to their device manager / configuration utilities (like yast for SUSE). Also, we added recently a function to alsa library to obtain a list of device names. So ask application developers to integrate this function to their application, so users can easy choose the right playback / record device. Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Using alsamixer _not_ interactively
On Mon, 13 Aug 2007, Zbigniew Baniewski wrote:
> Is it possible to set just the choosed "sliders" not using the usual screen,
> but with a command-line parameter, like f.e. "alsamixer -set surround 23/23"
> (for left/right)? I would to have alsamixer functionality available from
> a batch file.
>
> If it's possible - could be possible to read currently set levels in the
> same way?
Use 'amixer' utility.
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Re: [Alsa-user] hda-intel on Dell Inspiron 1300 has suddenly stopped working
On Sun, 15 Jul 2007, stan wrote:
> On Sat, 14 Jul 2007 10:50:22 -0500
> "Jake B" <[EMAIL PROTECTED]> wrote:
> > Front Left: Playback 31 [100%] [0.00dB] [on] <--- actual setting
> > Front Right: Playback 31 [100%] [0.00dB] [on] <--- actual setting
> > Front Left: Playback 16 [52%] [-22.50dB] [on]
> > Front Right: Playback 16 [52%] [-22.50dB] [on]
> Just looked at my amixer output and it has zero there as well. I must
Zero in dB brackets means zero gain and zero attenuation. Negative values
mean gain, positive attenuation.
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Re: [Alsa-user] BUGTRACK PROBLEM - compiling alsa-utils-1.0.14rc2 on suse 10.2 64 bits
On Fri, 9 Mar 2007, Sergei Steshenko wrote: > Secondly, if you can, to fix access to ALSA bug tracking system - I remember What problem do you have with Mantis? Jaroslav ----- Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA can only use default - Not hw:0,0 hw1,0 etc.
On Mon, 15 Jan 2007, Johan Spee wrote: > Frank Barknecht <[EMAIL PROTECTED]> wrote: > > > $ arecord -L > > gives a list of predefined PCM devices you could use. One of these > > should be named "spdif". > > 'should be'. But this is ALSA we are dealing with... > (No offence Frank, my cynicism does not concern you) > > # arecord -L > List of CAPTURE Hardware Devices > card 0: T22 [Terratec PHASE 22], device 0: ICE1724 [ICE1724] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: T22 [Terratec PHASE 22], device 1: IEC1724 IEC958 [IEC1724 IEC958] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] > Subdevices: 3/3 > Subdevice #0: subdevice #0 > Subdevice #1: subdevice #1 > Subdevice #2: subdevice #2 > > Why call it 'spdif' or 'digital' when you can call it 'IEC1724 IEC958 > [IEC1724 IEC958]' right? You can use 'iec958' device name. > But that is not the 'name' I could use. Neither > is 'T22'. The only thing that works is 'plughw:0,1'. 'plug:iec958:T22' or 'plug:iec958' or 'plug:spdif:T22' or 'plug:spdif' device does not work? The iec958 device is raw device (without any conversions). Applications expect to work with 16-bit samples usually. Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with multichannel LADSPA plugins
On Tue, 19 Dec 2006, Benjamin Eikel wrote: > Hello, > > I am trying to setup a lowpass filter for my subwoofer channel. I used > the configuration template which is given in the ALSA wiki > (http://alsa.opensrc.org/SurroundSound) as a template. > > My asound.conf works when I use it like it is given below (sound from > all speakers). But when I uncomment the second LADSPA plugin for the > lowpass (plugin number 1), then I will only hear sound from the > subwoofer (so only channel number 2 is processed lowpass_21to21). The > other channels seem to be muted. I also tried the same setup which is > given in the Wiki mentioned above, but this does not work at all (the > delay_0.01s plugin is called delay_0,01s on my machine; it think it's > because of the German locale; that's why I am using the ids for the > LADSPA plugins now). I tried to set the policy of my first LADSPA plugin > (number 0) to 'none', but this did not help. I was still hearing only > the sound from the subwoofer. I tried to use 3 plugins for the 3 > channels, each with policy none, but this did not work too. The ALSA > documentation I found (e.g. > http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html) is not > very helpful at all (I am not sure how to set these bindings inside the > plugins correctly). When I connect upmix_20to51 directly to > upmix_21to51, the sound is working fine (of course without lowpass). > > I would greatly appreciate any hint about what I am doing wrong. It was a bug (thanks for this report). The patch (or full file) which corrects "policy none" behaviour is here: http://hg.alsa-project.org/alsa-lib?fd=b838a4b481c2;file=src/pcm/pcm_ladspa.c Or, please, wait for the alsa-lib 1.0.14rc2 package which will contain this change. Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] symbol versioning issue in alsa-lib 1.0.13
On Fri, 13 Oct 2006, Yoann WALTHER wrote:
> Hi,
>
> I'm working on a set-top-box running mips, using linux 2.6.15, with alsa
> driver enabled v1.0.10rc3, and uclibc.
> I cross-compiled alsa-lib and alsa-utils, and got into issues with symbol
> versioning enabled :
>
> If I let symbol versioning enabled, aplay will fail to get hw/sw params
> correctly. But if I compile without symbol versioning, it works great.
> Does anyone know why such symbol versioning is used, and whether it is
> useful or not to keep it ?
> Maybe symbol versioning is not supported by uclibc, or is there a bug in the
> alsa-lib ?
If you compile all applications against one alsa-lib version I suggest to
disable versioning. It's only useful to run old (ALSA 0.9) binaries with
newer versions of alsa-lib. I guess, that your runtime linker (ld.so) does
not care about symbol versions.
Jaroslav
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Re: [Alsa-user] configuring ALSA driver to avoid invertion
On Thu, 5 Oct 2006, Andrew Gaydenko wrote:
> Jaroslav,
>
> Have tried the script. With given sample a speaker cone moves from a
> listener rather toward one. As 0x12345600 hasn't 0x8000 bit set, I
> suggest a cone must move toward a listener. Well, to avoid any mysticism
> I simply measured DAC output with multimeter :-) The output is negative,
> of course.
Okay, what about route plugin:
pcm.routetest {
type route
slave.pcm "hw:0,0";
ttable {
0 { 0 -1.0 }
1 { 1 -1.0 }
}
}
Replace -D hw:0,0 with -D routetest.
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Re: [Alsa-user] configuring ALSA driver to avoid invertion
On Thu, 5 Oct 2006, Andrew Gaydenko wrote:
> Jaroslav,
>
> I use (with JACK) 'hw:0,0' for analog output and 'hw:0,1' for SPDIF one.
Could you try a more "lowlevel" tool like aplay with a raw file?
You may try the python script bellow (modify values as you want to measure
level for different samples). Note that hw: devices are signed for
ICE1724, so zero is in the middle of range, minimum value is 0x8000
and maximum 0x7f00.
import struct
import os
FILE="file.raw"
SAMPLE=0x12345600
FORMAT="S32_LE"
fp = open(FILE, "w+")
x = struct.pack("I", SAMPLE)
for i in range(0, 10):
fp.write(x)
fp.close()
os.system("aplay -D hw:0,0 -f %s -c 2 -r 48000 %s" % (FORMAT, FILE))
os.remove(FILE)
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Re: [Alsa-user] configuring ALSA driver to avoid invertion
On Tue, 3 Oct 2006, Andrew Gaydenko wrote:
> I use ICE1724 ALSA driver for Terratec Aureon 7.1 Space card. Almost
> all is fine, big thanks to developers! Nevertheless, I have noticed
> both default (line out) and plug:spdif devices'es outputs are inverted.
> Is it possible to configure additional inversion and, as a result, to
> get "normal" output signal?
>
> I'm not sure it is the the driver bug - probably, it is manufacturer's bug :-)
Is not your problem related to signess? Did you tried to use 'U32_LE'
format?
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Re: [Alsa-user] Which Card for direct Audio-Recording
On Mon, 10 Jul 2006, Lee Revell wrote: > On Mon, 2006-07-10 at 12:25 -0700, Bill Unruh wrote: > > > > > >> Example: sox infile -t alsa -w -s /dev/snd/pcmC0D0p > > > > > > That's really from the sox man page? It's totally wrong. So is the > > > page: > > > > Yup that is the man page. > > > > .alsa ALSA /dev/snd/pcmCxDxp device driver > > This is a pseudo-file type and can be optionally > > compiled > > into SoX. Run sox -h to see if you have support for > > this > > file type. When this driver is used it allows you to > > open up > > the ALSA /dev/snd/pcmCxDxp file and configure it to use > > the > > same data format as passed in to SoX. It works for > > both > > playing and recording sound samples. When playing > > sound > > files it attempts to set up the ALSA driver to use the > > same > > format as the input file. It is suggested to always > > override > > the output values to use the highest quality samples > > your > > sound card can handle. Example: sox infile -t alsa -w > > -s > > /dev/snd/pcmC0D0p > > > > > > http://www.domenech.org/bt878a-adc/alsa-e.htm > > > > > > Does "sox infile -t alsa -w -s plughw:1,1" work? > > > > No it does not. However > > sox pluck/p6.wav -t alsa /dev/snd/pcmC0D0p > > DOES work. (or with the -w -s option as well) > > So it seems that sox does directly access the sound files. > > Why does sox abuse the ALSA API this way? Why can't it use the alsa-lib > interface like everyone else? Because it uses 'FILE */fread()/fwrite()' internally. Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Amixer - mute only one channel
On Tue, 4 Jul 2006, Vladimír Volčko wrote: > I see. Could you recommend sound HW which can do this feature? Almost all not AC97 based, for example HDA codecs, envy24... Jaroslav > Jaroslav Kysela napsal(a): > > On Tue, 4 Jul 2006, Vladimír Volčko wrote: > > > > > >> Thanks for quick reply, but when I try your recommended command, both > >> channels are muted ... see below: > >> > > > > Standard AC97 codecs cannot mute left and right channels separately (see > > pswitch-joined flag). It's the hardware constraint. > > > > Jaroslav > > > > > Using Tomcat but need to do more? Need to support web services, security? > Get stuff done quickly with pre-integrated technology to make your job easier > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo > http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 > ___ > Alsa-user mailing list > Alsa-user@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user > Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE LabsUsing Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Amixer - mute only one channel
On Tue, 4 Jul 2006, Vladimír Volčko wrote: > Thanks for quick reply, but when I try your recommended command, both > channels are muted ... see below: Standard AC97 codecs cannot mute left and right channels separately (see pswitch-joined flag). It's the hardware constraint. Jaroslav ----- Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE LabsUsing Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Amixer - mute only one channel
On Tue, 4 Jul 2006, Vladimír Volčko wrote:
> Hi!
>
> It is posible mute only one channel by amixer?
> For example I need mute only left 'line' channel. Setting 0% volume is
> not solution.
amixer sset Line frontleft,mute
Jaroslav
-
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Re: [Alsa-user] Multi-device and dmix plugin
On Wed, 28 Jun 2006, Peter Rödiger wrote:
> Alright, thanks. So, just to be clear, there is no way of having the
> behavior explained above AND dmix? That would be a pity since it is (at
> least in my view) a very normal thing that should work. And by the way,
> i tworks under Windows, so it is indeed possible. Any other suggestion
> of achieving this with asound.conf?
You may try the opposite way, use the multi plugin on top of the dmix
plugin. But it will probably not work well, because the cards have
different clocks and the multi plugin does not do any corrections. It's
better to buy a multichannel hardware which is quite cheap.
plug->multi->dmix->hardware
Jaroslav
-
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Get stuff done quickly with pre-integrated technology to make your job easier
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Re: [Alsa-user] Multi-device and dmix plugin
On Wed, 28 Jun 2006, Peter Rödiger wrote:
> - Support multi-card/device for direct plugins
>
>
> - Support multi-card/device for dmix/dsnoop/dshare plugins
> The unique ipc key is calculated based on card/device/sub index
>
> - Clean up and share the code among all d* plugins
>
>
> - Refer the defaults.pcm.* configuration
> The base ipc_key number, ipc_gid and ipc_perm are referred.
>
> I understand, that it is now possible to do this sort of thing I'm
> looking for, is that right?
Nope. It means that dmix will work well without touching of default
configuration files for all soundcards in the system but separately.
Jaroslav
-
Jaroslav Kysela <[EMAIL PROTECTED]>
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Get stuff done quickly with pre-integrated technology to make your job easier
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Re: [Alsa-user] Amixer: How to change volume only for one PCM line (left or right)?
On Fri, 2 Jun 2006, Vladimír Volčko wrote:
> index=0 is not solution. It sets both channel (left and right):
>
> amixer cset iface=MIXER,name='PCM Playback Volume',index=0 50%
amixer cset iface=MIXER,name='PCM Playback Volume',index=0 25%,50%
index is for the element indetification, not for the channel selection
Jaroslav
-
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Re: [Alsa-user] Amixer: How to change volume only for one PCM line (left or right)?
On Fri, 2 Jun 2006, Vladimír Volčko wrote: > Hi! > > I try managing subject, but I have no success. > > When I use following command (inspired by man page example): > amixer cset iface=MIXER,name='PCM Playback Volume',index=0 50% > > ... amixer set both lines (see ouput bellow): > > numid=23,iface=MIXER,name='PCM Playback Volume' > ; type=INTEGER,access=rw---,values=2,min=0,max=31,step=0 > : values=16,16 > > For command: > amixer cset iface=MIXER,name='PCM Playback Volume',index=1 90% > > ... amixer report error mesage: > amixer: Control default cinfo error: No such file or directory Try index=0. Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Multiple dmix instances on multiple cards
On Tue, 7 Feb 2006, EmIScA wrote: > ipc_key 5678292 > ipc_key 5678291 Wrong. The difference between these two values should be at least 2. One instance uses 2 ipc keys. Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs --- This SF.net email is sponsored by: Splunk Inc. Do you grep through log files for problems? Stop! Download the new AJAX search engine that makes searching your log files as easy as surfing the web. DOWNLOAD SPLUNK! http://sel.as-us.falkag.net/sel?cmd=lnk&kid=103432&bid=230486&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Dmix madness?
On Tue, 7 Feb 2006, Ismail Donmez wrote: > [EMAIL PROTECTED] ~ $ ls -1 /tmp/|grep alsa-dmix|wc -l > 39082 > > Alsa driver/lib/utils 1.0.11rc2 Could you trace the conditions when the files are left in /tmp directory? It appears that something kills the resource daemon before it can cleanup things. Jaroslav ----- Jaroslav Kysela <[EMAIL PROTECTED]> --- This SF.net email is sponsored by: Splunk Inc. Do you grep through log files for problems? Stop! Download the new AJAX search engine that makes searching your log files as easy as surfing the web. DOWNLOAD SPLUNK! http://sel.as-us.falkag.net/sel?cmd=lnk&kid=103432&bid=230486&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] stable APIs and ABIs
On Tue, 24 Jan 2006, Sergei Steshenko wrote:
> Takashi, as end user I want to know nothing about alsa-lib and kernel.
>
> I want to have a website with driver per card, i.e. I want to perform
> only intellectualy primitive lookup operation: read the file names in
> repository, find the file which matches my card name and install it.
>
> Like with 'xpdf' - I see the program name (moral equivalent of my card
> name) and version suffix. If the newest version doesn't work, I revert
> to an older one.
If the end user is not capable to do the standard *nix installation, then
nothing can help. The distribution makers are supposed to offer the user
friendly interface for installation and upgrade. It's out of scope of
standard projects to prepare such "one click" operations for end users.
I haven't found anything interesting for our project in this discussion,
so I vote to end it.
Jaroslav
-
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ALSA Project, SUSE Labs
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Re: [Alsa-user] very low framesize
On Wed, 4 Jan 2006, "Alexander Carôt" wrote:
> Hi to all,
>
> I have a general technical question :
>
> With low latency kernel and related stuff I typically run my soundcard with
> 48kHz and 128 Samples/frame.
>
> What surprises me is the fact that in case of using 8kHz only I still can't
> lower the framsize to values below 128. 64 or 32 don't work at all
> (dropouts,under/overruns). This surprises me because with 8kHz it should
> take much less computation power.
>
> Can anyone tell why this is the case ?
Almost all PCI hardware optimizes DMA transfers and uses FIFOs.
So, typically you cannot go bellow 64-128 bytes per period.
Allowing such values is the bug in driver - if you tell us
which hardware does this, we can disable these period sizes
in driver.
Jaroslav
-
Jaroslav Kysela <[EMAIL PROTECTED]>
Linux Kernel Sound Maintainer
ALSA Project, SUSE Labs
Re: [Alsa-user] Unable to find an audio port (1) for channel 1
On Mon, 2 Jan 2006, Jaroslav Kysela wrote:
> > Running "speaker-test -c2 -Dplug:test" only plays the "front right"
> > audio out of both front speakers.
>
> Confirmed. It seems that the problem has nothing with the ALSA LADSPA
> plugin code but with the latest mmap code optimizations. I'm still
> figuring where the real bug is.
The attached patch fixes this bug.
Jaroslav
-
Jaroslav Kysela <[EMAIL PROTECTED]>
Linux Kernel Sound Maintainer
ALSA Project, SUSE Labs? out.txt
Index: pcm_mmap.c
===
RCS file: /cvsroot/alsa/alsa-lib/src/pcm/pcm_mmap.c,v
retrieving revision 1.76
diff -u -r1.76 pcm_mmap.c
--- pcm_mmap.c 30 Nov 2005 11:39:21 - 1.76
+++ pcm_mmap.c 2 Jan 2006 12:11:10 -
@@ -326,122 +326,130 @@
for (c = 0; c < pcm->channels; ++c) {
snd_pcm_channel_info_t *i = &pcm->mmap_channels[c];
snd_pcm_channel_area_t *a = &pcm->running_areas[c];
+ char *ptr;
+ size_t size;
unsigned int c1;
- if (!i->addr) {
- char *ptr;
- size_t size = i->first + i->step * (pcm->buffer_size -
1) + pcm->sample_bits;
- for (c1 = c + 1; c1 < pcm->channels; ++c1) {
- snd_pcm_channel_info_t *i1 =
&pcm->mmap_channels[c1];
- size_t s;
- if (i1->type != i->type)
+ if (i->addr) {
+ a->addr = i->addr;
+ a->first = i->first;
+ a->step = i->step;
+ continue;
+}
+size = i->first + i->step * (pcm->buffer_size - 1) +
pcm->sample_bits;
+ for (c1 = c + 1; c1 < pcm->channels; ++c1) {
+ snd_pcm_channel_info_t *i1 = &pcm->mmap_channels[c1];
+ size_t s;
+ if (i1->type != i->type)
+ continue;
+ switch (i1->type) {
+ case SND_PCM_AREA_MMAP:
+ if (i1->u.mmap.fd != i->u.mmap.fd ||
+ i1->u.mmap.offset != i->u.mmap.offset)
continue;
- switch (i1->type) {
- case SND_PCM_AREA_MMAP:
- if (i1->u.mmap.fd != i->u.mmap.fd ||
- i1->u.mmap.offset !=
i->u.mmap.offset)
- continue;
- break;
- case SND_PCM_AREA_SHM:
- if (i1->u.shm.shmid != i->u.shm.shmid)
- continue;
- break;
- case SND_PCM_AREA_LOCAL:
- break;
- default:
- assert(0);
- }
- s = i1->first + i1->step * (pcm->buffer_size -
1) + pcm->sample_bits;
- if (s > size)
- size = s;
+ break;
+ case SND_PCM_AREA_SHM:
+ if (i1->u.shm.shmid != i->u.shm.shmid)
+ continue;
+ break;
+ case SND_PCM_AREA_LOCAL:
+ break;
+ default:
+ assert(0);
}
- size = (size + 7) / 8;
- size = page_align(size);
- switch (i->type) {
- case SND_PCM_AREA_MMAP:
- ptr = mmap(NULL, size, PROT_READ|PROT_WRITE,
MAP_FILE|MAP_SHARED, i->u.mmap.fd, i->u.mmap.offset);
- if (ptr == MAP_FAILED) {
- SYSERR("mmap failed");
+ s = i1->first + i1->step * (pcm->buffer_size - 1) +
pcm->sample_bits;
+ if (s > size)
+ size = s;
+ }
+ size = (size + 7) / 8;
+ size = page_align(size);
+ switch (i->type) {
+ case SND_PCM_AREA_MMAP:
+ ptr = mmap(NULL, size, PROT_READ|PROT_WRITE,
MAP_FILE|MAP_
Re: [Alsa-user] Unable to find an audio port (1) for channel 1
On Mon, 2 Jan 2006, Adam Nielsen wrote: > > > ALSA lib pcm_ladspa.c:1283:(snd_pcm_ladspa_parse_ioconfig) Unable to > > > find an audio port (1) for channel 1 > > > It means that the used LADSPA plugin has no second audio port. > > When you say "no second audio port" is that different to processing a > stereo signal? Because it was my understanding that the LADSPA plugin > only processed one channel of sound, and ALSA duplicated this so that if > you had a stereo sound there would be two instances of the LADSPA plugin > running (I think this is what "policy duplicate" is for.) Ok, it means that the LADSPA plugin has no second audio port in this contents. If you use a plugin with one input and one output, you cannot try to assign more inputs. And the duplicate policy works only for "mono" LADSPA plugins. For other plugins, you have to use the "none" policy. > Running "speaker-test -c2 -Dplug:test" only plays the "front right" > audio out of both front speakers. Confirmed. It seems that the problem has nothing with the ALSA LADSPA plugin code but with the latest mmap code optimizations. I'm still figuring where the real bug is. Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs --- This SF.net email is sponsored by: Splunk Inc. Do you grep through log files for problems? Stop! Download the new AJAX search engine that makes searching your log files as easy as surfing the web. DOWNLOAD SPLUNK! http://ads.osdn.com/?ad_id=7637&alloc_id=16865&op=click ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Unable to find an audio port (1) for channel 1
On Fri, 30 Dec 2005, Adam Nielsen wrote:
> ALSA lib pcm_ladspa.c:1283:(snd_pcm_ladspa_parse_ioconfig) Unable to find an
> audio port (1) for channel 1
> aplay: main:547: audio open error: Invalid argument
>
> So what does this mean? It looks like solving this problem will fix the
> mono-only LADSPA output problem.
It means that the used LADSPA plugin has no second audio port.
Jaroslav
-----
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ALSA Project, SUSE Labs
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Re: [Alsa-user] Assertion `snd_pcm_format_linear(slv->format)' failed
On Fri, 30 Dec 2005, Adam Nielsen wrote:
> > What are you trying to do?
>
> I'm still trying to get stereo output when passing the sound through a
> LADSPA plugin (which in the latest version of ALSA converts any incoming
> stereo source into mono.)
Could you enter a bug report if the current LADSPA code has some problems?
It's difficult to track your conversation if you deleted all "vital"
information from your previous e-mail.
Also 'aplay -v' should give you more usefull information about the actual
routing in the LADSPA chain and the ALSA plugin chain.
Jaroslav
-
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ALSA Project, SUSE Labs
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Re: [Alsa-user] Change volume of left or right channel only - how to do?
On Sun, 18 Apr 2004, Christian Anton wrote:
> Hi everybody!
>
> I want to change volume of only left or right channel of my Output of my
> soundcard by a script, and not touch the volume of the other channel.
> By example: I want to set volume on left side to 70% an leave the right
> side on 35%. I have multiple Soundcards in my Computer (3xens1371 and
> 1xintel8x0) I need a scriptable solution because i want to do so within
> a php-Script.
>
> Can anybody give me an idea how to do so? I already read the manuals of
> amixer, but it seems it can only change left AND right volumes at a time.
amixer set Speaker frontleft 25%
amixer set Speaker frontright 65%
Jaroslav
-
Jaroslav Kysela <[EMAIL PROTECTED]>
Linux Kernel Sound Maintainer
ALSA Project, SuSE Labs
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