Re: [Alsa-user] rate coverter after dmix
On 5/9/17, remu kellywrote: > > How this can be achieved, seeing that we can't have a plugin after dmix. Couldn't you use snd-aloop and have a plugin AFTER dmix? It basically creates a loopback interface who's output channel is the input channel. Although I've never used it (yet). But one way to pulse over jack without using the jack module part of pulse. -- 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
Re: [Alsa-user] Short pauses in playback depending on output volume
Does the playback file exist on a slow storage device? I have a few usb sticks that pause with cd quality wav files because of the slow I/O of the device. But the same file compressed to flac or mp3 on the same storage device will play without pauses. Or the cd quality file on any "faster" device, even sdhc cards plays without pauses. It could be a power management issue and hardware related. Swapping around the ports that things are plugged into or using a powered hub can help, sometimes. Try running something like nmon when you trigger the issue. With "L" for cpu, it can be a little more informative. With a blue "w" when it's waiting on something from the system. Which might indicate hardware type issues like bus speed or swap usage. Also check dmesg for indications of hardware issues (iffy connections). - James On 1/15/17, Fabian Kellerwrote: > Hi all, > > While developing an audio software based on PortAudio, I discovered a > surprising problem related to ALSA: I'm getting short pauses in the > audio playback (sounding like typical buffer underruns) depending on > the audio amplitude. > > As a test, I have generated two wave files containing pure white > noise. One of them with an amplitude of 0.5 the other one using a full > range amplitude of 1.0. I'm playing both files with aplay, but I'm > getting the same behavior with other players and also with the > software I'm developing: The 0.5 amplitude files plays without any > issues. But the 1.0 amplitude file plays with short breaks in the > audio stream. I'm getting about two of these breaks per minute, but > there does not seem to be a deterministic pattern. I would guess the > pauses are <100 ms in duration, which is why I was debugging in the > direction of buffer underruns for many days until I discovered this > amplitude effect. I have xruns logging enabled, so I'm pretty sure > this is not related to that. > > Do you have any idea what could be causing this? > > System specs: > - Ubuntu 14.04. with the default libportaudio2 (based on the last 2014 > release) > - Standard Intel onboard sound: PCH [HDA Intel PCH], device 0: ALC892 > Analog [ALC892 Analog] > > Another test: Amplitude 0.9 also has breaks, so it is not just the > full range amplitude. > > Thanks, > Fabian > > -- > Developer Access Program for Intel Xeon Phi Processors > Access to Intel Xeon Phi processor-based developer platforms. > With one year of Intel Parallel Studio XE. > Training and support from Colfax. > Order your platform today. http://sdm.link/xeonphi > ___ > Alsa-user mailing list > Alsa-user@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user > -- 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
Re: [Alsa-user] ice1712 recording
> You can mix all 4 inputs down into one stream and then record that, but do you really mean that you can record to 4 separate application threads concurrently without mixing? No, I mean you can record all four channels as input at the same time with the same app. Hence the -c 4 aka 4 channels. In audacity you just select 4 channels and press record. Once recorded you can break them out into 4 mono channels and save each individually (unmixed). There's no mixing involved until you configure it to do so. WAV files and other audio formats often contain multiple tracks. Individual unmixed tracks. They are mixed at the time of playback if configured to do so. By default sound is often configured for stereo with one track panned left and one track panned right. But it's two tracks, not mixed (until reproduced and analog-illy mixed in air). Baring cheap sound devices that have bleed over between tracks with unintentional mixing. In audacity you can separate the channels and unpan them for two true mono tracks. As well as a few CLI options for the same. Sox is good for that. With unix-isms you can send your output to stdout and pipe to stdin of another app. Like tee which can then save a file and redirect the same output to another file. I've sometimes gone this route to output the raw WAV and a compressed MP3 at the same time to two different storage devices / locations. I seem to recall an arecord option to output each channel as it's own mono file. Which you could tail -f on other terminals to pipe that to other things. Seems like --separate-channels is that option. $ arecord --help - James -- Developer Access Program for Intel Xeon Phi Processors Access to Intel Xeon Phi processor-based developer platforms. With one year of Intel Parallel Studio XE. Training and support from Colfax. Order your platform today.http://sdm.link/xeonphi ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ice1712 recording
It can record from all 4. Although many applications only care about left / channel 1 (defaults). I tend to run pulseaudio over jackd setup. As that was the only way to have your mic be an input other than 1 for apps like skype. Since channel 1 and 2 are typically stereo output. $ man arecord -c, --channels=# the number of channels. The default is one channel. . I suspect that adding -c 4 would overcome the issue. The -f dat has a default of -f S16_LE -c 2 -r 48000. So you might want to change that parameter as well. $ arecord -Dhw:1 -f S24_LE -t wav -c 4 -r 96000 output.wav I have that card, although I haven't booted that old PCI system in a while. What I recall of that card, that should be the highest sampling available for it. 24 bit, 96kHz, 4 channels (input). In theory it has a 10 channel output mixer. It's a nice card, a shame most newer things don't have PCI in favor of PCIe. You might want to simplify your .asoundrc, it's more likely to get in the way than help these days. #- defaults.ctl.card 1 defaults.pcm.card 1 defaults.pcm.device 0 #- Assuming that it didn't get index 0 in /proc/asound/cards. But did get 1. In alsa speak that's equivalent to -Dhw:1,0 . Although you might want to omit the ,0 since that's typcially playback, not capture, so -Dhw:1 - James On 12/4/16, Ralf Mardorfwrote: > On Sun, 4 Dec 2016 20:18:24 +, zcx wrote: >>I have a Delta 44 sound card here that uses the ice1712 chipset. >> >>Am I right in thinking that although the card has 4 mono inputs, it >>can only capture one stream at a time? arecord seems to think so... > > Only one app can grab the device, if you run two instances of the same > app, only one instance can grab the device. > > If several apps should be able to use the device at the same time, you > need a workaround, e.g. dmix or e.g. a sound server, such as e.g. jackd. > > -- > 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 > -- 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
Re: [Alsa-user] sans-pulseaudio Firefox? was: a strange thing
I've used alsa and firefox. By default java (in debian) is configured for pulseaudio. FILE: /etc/java-7-openjdk/sound.properties But both alsa and pulseaudio configs are in there (in debian). Just comment out pulse and uncomment alsa. I switch between a lot, depending on if I am home or using my laptop as a laptop. Which also affects icedtea-web, the java plugin for the browser IIRC. FILE: .asoundrc ### for pulseaudio #ctl.pulse { type pulse fallback sysdefault } #pcm.pulse { type pulse fallback sysdefault } #ctl.!default { type pulse fallback sysdefault } #pcm.!default { type pulse fallback sysdefault } ### for alsa defaults.ctl.card 0 defaults.pcm.card 0 defaults.pcm.device 0 ###---end--- Comment swap there too. As well as comment modify/swap .config/pulse/client.conf since I pulseaudio over the network. The 30 band calf eq chews up a lot of the CPU so I offloaded that to another laptop. A little high end boost to keep the ancient speakers sounding normal-ish. Depending on my lazy level I'll sometimes use two users, one configured for alsa, one for pulseaudio. About the only issue is that adobe's flash uses pulseaudio, so if you're still using that you'll have "issues" with flash content. In days of old there's a compat thing you could install and it can be made to work. I'm not sure of the current methodology. But at least aoss can be avoided in most cases now. Most of my flash stuff these days is the freshplayer plugin and googles chrome pepperflash plugin (in firefox). Freshplayer from sources in debian stable, and pepperflash extracted and manually maneuvered. Recently moved out of the chrome.deb (version 54+) and put somewhere else. But it respects the .asound the client.conf config settings. https://get.adobe.com/flashplayer/otherversions/ The ppapi one is the pepperflash download. My manual method puts them in the ~/. settings area so I never had to be root and only that one user gets to use it. YMMV, depending on distro. I tend towards debian stable from a minimal install via debootstrap. It's faster for me on my slow internet, and I can get extras like network drivers while still on the network with the host linux install. Much like an arch-chroot install. - James -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] usb audio. should this not work?
As previously said pavucontrol to configure pulseaudio. BITD the default sound card was index 0. Which could not be overridden by some things. So re-indexing was the desired way to override things. These days most things respect the .asoundrc. And you can have a pretty short one to change your default index #. (if not using pulseaudio) FILE: ~/.asoundrc defaults.ctl.card 2 defaults.pcm.card 2 defaults.pcm.device 0 You really only need the defaults.pcm.card # one though. The others are nice for things that change mixer levels in app or for hdmi audio out which might be device 3, not 0. Where card # is what is listed in /proc/asound/cards. YMMV. $ cat /proc/asound/cards Many apps let you override the default by parameter as well. Such as -D hw:2 for aplay. Or --ao=alsa:device=hw,2 for mpv which is a fork of mplayer(2?). With the above .asoundrc it's simpler with just --ao=alsa, or completely omit the option. Things like audacity let you select the available card under preferences. If you wish to use something other than the system defaults. - James -- Attend Shape: An AT Tech Expo July 15-16. Meet us at AT Park in San Francisco, CA to explore cutting-edge tech and listen to tech luminaries present their vision of the future. This family event has something for everyone, including kids. Get more information and register today. http://sdm.link/attshape ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] turn off system beep
You might also blacklist snd-pcsp Or maybe purge and re-install the alsa-base stuffs. # dpkg --purge --force-all alsa-base # apt-get install alsa-base Make sure those needed blacklist items are there and add them if need be. blacklist pcspkr blacklist snd-pcspkr blacklist pcsp blacklist snd-pcsp My overkill list added to my blacklist items. You might also check /etc/rc*.d/ for anything that might be playing sound(s) on shutdown. If that other thing doesn't work. $ find /etc/rc?.d/ -name 'K*' - James On 8/17/11, Julien Claassen jul...@c-lab.de wrote: Hello Xenia! the only idea I'd have, is to re-enable the pcspkr module and in alsamixer 0 or whatever you use to setup your audio ardware -, mute it. Otherwise the real pc-speaker as such is no ALSA device, as far as I'm aware. The pc-speaker module is, I believe, intended to use it as a low-quality playback-device. Kind regards Julien =-=-=-=-=-=-=-=-=-=-=-=- Such Is Life: Very Intensely Adorable; Frightening Absence Just Arriving, Reigns Disappeared, Ornate - flowers! == Find my music at == http://juliencoder.de/nama/music.html . If you live to be 100, I hope I live to be 100 minus 1 day, so I never have to live without you. (Winnie the Pooh) -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] plughw versus hw
plughw is probably better at sharing a device than hw would be. And plughw probably allows for some conversion of content. Otherwise they are functionally the same IMO. Not that I'd know since I haven't really delved that deep into things. You might check lsof or fuser to see if something is using the sound device and keeping you from using hw instead of plughw. Or just use plughw since it works. If it's a content conversion issue, you might try to create a converted version of the media and see if that fixes the issue when using hw. Sox can do a lot of conversions. Ffmpeg as well. Many means to an end. Not that it fixes the issue, but it can help to better understand the issue. You might also try renaming your .asoundrc to see if that frees up hw to be used in the way that you are trying to use it. If that works, then there's something in your .asoundrc that's getting in the way. - James On 6/21/11, Pierre Habraken pierre.habra...@free.fr 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 Mosgalinmosga...@vm10124.spb.edu 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 ? Pierre -- 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 -- 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] First post
Well many source packages default to /usr/local/ Many distros default to /usr/ And the distros IGNORE /usr/local/ unless otherwise told. It's not a compile thing, it's a runtime thing. Of course you could always run things with the full paths /usr/local/bin/alsamixer and such. But if you add the location in $PATH, it'll find it. But if you use the default /usr/local/ it might look for and load the distros version from /usr/ first. So I generally overwrite the distro's versions. i.e. make install and --prefix=/usr. Versus building debs and installing it that way which is the preferred way to do thing. But mainly because I can never recall the fakeroot debian/binary stuff to make debs off the top of my head. And I don't always have networking setup to google it at the stage that I'm installing alsa. But I don't do enough from source stuff to really consider my setup a different distro. Just customized per say. If only for the optimization of not having to look through 1,000 drivers for the 1 that is actually used. And media players with CPU specific optimizations are always nice. As a side note, alsa is in the 2.6 kernel tree. Are we on 2.8 yet? So if you compile a recent kernel, you automagically get a recent alsa version with it. Or if your distro offers a recent kernel. It's done for you. No need to re-invent the wheel as previously said. But sometimes your distro doesn't package things in a way that you want to use them. i.e. Timidity with sequencer support. Jackd with sequencer support. Alsa with OSS emulation. And other fine tuning type needs. Or your distro is on such an ancient kernel, that stuff just doesn't work at all given the lack of age of your hardware versus the copious amounts of age in your kernel version. - James On 6/20/11, David Henderson dhender...@digital-pipe.com wrote: Thanks again for the continued help James. I knew '--prefix' was a 'configure' option, but thought one would use it when permanently installing the software to a non-standard directory on the system. Since this software is being compiled on a temp system and installing to a staging directory, wouldn't the 'DESTDIR' be a better option to use while compiling the software so it can be packaged and installed on the custom distro? Thanks for the tips on the kernel headers and configure parameters. :) Dave On 06/19/2011 07:06 PM, James Shatto wrote: --prefix is a ./configure option. If you're going to apply the new alsa to an existing distro kernel and not a custom from source one. You'll likely need to install the kernel-headers package for that kernel and distro. And may need to manually move the old version of alsa (or remove). Plus that whole depmod thing. $ dpkg -l '*kernel*headers*' Which resolves to linux-kernel-headers in debian. Which is a psuedo package for: linux-libc-dev 2.6.26-26lenny3 and of course 2.6.26-26lenny3 resolves to linux-tree-2.6.26lenny3 so: # apt-get install linux-libc-dev linux-tree-2.6.26-26lenny3 (in debian 5.0 / lenny) If it's a custom one, just don't make clean after making the kernel. It should reside in /lib/modules/`uname -r`/build/ or something like that. BITD, this would just be a symlink to/from /usr/src/linux and was what early alsa assumed by default. Depending on what multimedia features you need. You might want --with-sequencer=yes and --with-oss=yes and a --driver=your card options on your alsa-driver compile. Without those =no might be assumed. And you might compile ALL drivers which could take a really long time. Less so these days, but BITD, the better part of a day it seemed. It really depends on what you want interacting with your sound card. Timidity and other synth like software requires the --with-sequencer=yes if your card doesn't have native midi abilities (most don't these days). And various pulse-audio and browsers and other things that just need --with-oss=yes or things might not work as expected, if at all. Little things that you'll find out one way or another as you learn your way around. HTH, - James On 6/19/11, David Hendersondhender...@digital-pipe.com wrote: Hi James, thanks for your help too. :) I'll provide replies in the same fashion given. A) I don't want to overwrite the Kubuntu installation files as I'm compiling this version of alsa for my own distro. I would prefer to use Kubuntu's pre-packaged software within itself. So since the compiled version of alsa will be going into /opt/staging/alsa, should I include --prefix=/opt/staging/alsa as the parameter to configure? B) I'll assume at this point, that no matter what version of the Linux kernel is being used, it's still required to install the alsa-driver package. That being said, I'm going to run into the same problem as A above since the version of Kubuntu I'm using to build the custom distro isn't using the same kernel version. So what configure option do I have to pass in order
Re: [Alsa-user] First post
If you're really into going it on your own. There's gentoo, and there's LFS aka linux from scratch. Both of which impose a lot of source compilation. The inherent problem with sources is that you run into maintenance issues. i.e. If you use the same install for a long enough time, it'll eventually become unusable due to remnants of old versions and not enough hours in a lifetime to figure out what/where those are and manually correct. Ultimately you'll be doing fresh installs long before your hardware's expiration date. Not that I don't do regular installs myself. But I swap out hard drives every two years to be pro-active against that type of failure. And I do a lot of media editing, so I probably abuse my drives more than most. A distro is just a good ideal. There's configuration files that you really can't generate by hand without a pretty hefty understanding of what you are doing. Distros have done all this legwork for you and provide you with a sane default configuration file where you just need to uncomment a line to enable something or comment it to disable it. Lots of sanity saving things in a distro that you'll be scouring sources to figure out on your own in LFS land. And probably installing a distro anyway to cp their config. There's a lot to learn. But really you don't need to learn that stuff. There's no bread and butter / money in it. Sure you'll have a greater understanding. And should some do or die worst case scenario happen you'll know how to resolve it, where most other folks wont know where to begin. But really most IT jobs these days are installing and uninstalling and configuration gigs. We don't need to write a word processor, as one (several actually) already exist. And some of them aren't too shabby. As far as build systems. The configure + make + make install is the OLD way. Not all sources use that one. There's scons, mercurial, and various *make incarnations. And of course distro specific ways that are compatible with their package manager(s). Plus the typical development role of 1001 ways to do one thing. Fortunately alsa is still a bit old school. Or unfortunately depending on your POV. - James -- 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] First post
Ummm. I'm not sure if I follow you. $ make will build the objects and stuff in the current path of your source tree. $ make install copies the executables to the system usable locations. /usr/bin/ /lib/modules/. /usr/share/doc/. (which is why you need to be root in a lot of cases to run make install, but not to run make) A package maintainer will likely use stuff like what's in debian/rules debian/binary and such to build a package manager package, instead of using make install to place the important components (results) where they need to be. A package manager package lets you keep track of what got installed and where and the package provides additional features useful for long term maintenance and/or large scale deployment. If you want to build a package (i.e. .deb) you'd use those tools for that to do that. Otherwise you have to at least mimic make install. Which is a bit futile IMO, given that you could just run make install. i.e. how exactly would you create your tarball? From a diff of an entire backup before and after make install? By doing everything done by make install manually? That's fine for relatively small things like alsa. But for X, KDE, ???, and other more bulky entities. You'd need a couple lifetimes of spare time to re-invent make install. And scons install and and ... and ... Also bear in mind that if you're building something on a system other than the one where it will be deployed. You will run into some version compatibility issues. Just a minor difference in the API between version 1.0.24 and 1.0.25 could make things unusable. And as previously mentioned, alsa comes with the 2.6 kernel, so you'll have an existing version already in place that you will need to deal with, one way or another. When there's multiple versions of things, at runtime things like to load in alphabetical order or ascii order at least. Which generally means the that OLDer version takes priority. So even if you install your newer version, it's probably going to be ignored unless you remove or replace the older version. The manual approach to dependency hell I guess, of sorts. Lots of little things that will keep you from succeeding. It's probably time better spent learning an existing package management system IMO. Than to create your own. Especially if you're on your own and not part of team. But it's almost all open source so if you can read the source, everything that you need to know is there in one form or another. - James On 6/20/11, David Henderson dhender...@digital-pipe.com wrote: On 06/20/2011 11:52 AM, Pierre Lorenzon wrote: Hi, From: David Hendersondhender...@digital-pipe.com Subject: Re: [Alsa-user] First post Date: Sun, 19 Jun 2011 15:28:48 -0400 Thanks for the reply Pierre. I checked into the blfs book, but it merely says these five chapters will cover alsa and then gives you a basic type configure make. This is obviously not going to answer the questions below. :) Any other thoughts? Dave On 06/19/2011 11:22 PM, Pierre Lorenzon wrote: Hi, It looks like to me such questions are well answered in the blfs book. I personnaly think that the latter is a very good tool to build his own custom distro. Bests Pierre From: David Hendersondhender...@digital-pipe.com Subject: [Alsa-user] First post Date: Sun, 19 Jun 2011 14:41:08 -0400 Hi everyone! I'm currently expanding my knowledge of GNU/Linux to include building packages from scratch towards an overall goal of a custom distro. So far, I have a nice base for a command line OS, but want to expand into the multimedia aspect. Alsa was my first (only?) choice for the audio portion, but I'm running into problems. The alsa site is somewhat overwhelming to newbies and is easy to get lost. I have a few questions below from which I hope I can find help. All contributions are greatly appreciated. :) Thanks, Dave 1) Currently I have downloaded alsa-driver, alsa-lib, and alsa-utils packages. Is there an order in which these packages need to be compiled and installed? This question is answered by the blfs book. First alsa-lib and after alsa-utils. 2) I'm currently running the relatively new Linux kernel 2.6.33 so do I need the alsa-driver package? No ! I am running a 2.6.32 kernel and never installed alsa-driver. Anyway if the sound system is something very exotic it might be necessary ... Great one less thing to compile! :) 3) I've been able to successfully compile the alsa-lib package and install it in the custom distro. When I try to compile the alsa-utils package, I constantly get the error: checking for libasound headers version= 1.0.16... not present. configure: error: Sufficiently new version of libasound not found. I'm actually using an existing Kubuntu installation to build the packages for my custom distro. As a result, after I compiled the newer alsa-lib, I didn't install the package into
Re: [Alsa-user] First post
This is part of the reason that I use --prefix=/usr because the /usr/includes/ are also affected by the --prefix option (i.e. /usr/local/includes / which is empty). And I've never really gotten into the changing $PATH part of things. But there's a whole slew of -I and -L options (with a different case / case sensitive) for gcc to bypass / customize a lot of that. A real PITB IMO. But just my opinion. i.e. Use what is already there, not re-invent it in your image. And yes a bit OT at this point. - James On 6/20/11, David Henderson dhender...@digital-pipe.com wrote: I think your statement here i.e. how exactly would you create your tarball? From a diff of an entire backup before and after make install? best sums it up. Without a staging directory to install to, you would have to parse the entire FS in order to find what the make install step did. By using a staging directory, you still run make install, it just installs everything in it's retained hierarchy within that staging directory. That's why I said /opt/staging/alsa/bin in Kubuntu (build OS) becomes /bin in the custom distro. That's what the DESTDIR parameter does, it allows you to retain whatever directory hierarchy to use, but during the make install phase, instead of using / as the root, it uses whatever you include (e.g. DESTDIR=/opt/staging/alsa) as the value pre-pended for root. Honestly, at this point, we've gotten way off topic. lol These are all issues for me to work out, but appreciate you guys efforts. :) Presently, I'm thinking that alsa-utils (as we've determined alsa-driver probably doesn't have to be installed) is failing to compile because it's looking under /... for the header files and not /opt/staging/alsa/... Is there a way to make the configure script look into that directory for the header files during the configure phase? Thanks again for everyone's continued efforts in getting this matter resolved. Dave On 06/20/2011 04:06 PM, James Shatto wrote: Ummm. I'm not sure if I follow you. $ make will build the objects and stuff in the current path of your source tree. $ make install copies the executables to the system usable locations. /usr/bin/ /lib/modules/. /usr/share/doc/. (which is why you need to be root in a lot of cases to run make install, but not to run make) A package maintainer will likely use stuff like what's in debian/rules debian/binary and such to build a package manager package, instead of using make install to place the important components (results) where they need to be. A package manager package lets you keep track of what got installed and where and the package provides additional features useful for long term maintenance and/or large scale deployment. If you want to build a package (i.e. .deb) you'd use those tools for that to do that. Otherwise you have to at least mimic make install. Which is a bit futile IMO, given that you could just run make install. i.e. how exactly would you create your tarball? From a diff of an entire backup before and after make install? By doing everything done by make install manually? That's fine for relatively small things like alsa. But for X, KDE, ???, and other more bulky entities. You'd need a couple lifetimes of spare time to re-invent make install. And scons install and and ... and ... Also bear in mind that if you're building something on a system other than the one where it will be deployed. You will run into some version compatibility issues. Just a minor difference in the API between version 1.0.24 and 1.0.25 could make things unusable. And as previously mentioned, alsa comes with the 2.6 kernel, so you'll have an existing version already in place that you will need to deal with, one way or another. When there's multiple versions of things, at runtime things like to load in alphabetical order or ascii order at least. Which generally means the that OLDer version takes priority. So even if you install your newer version, it's probably going to be ignored unless you remove or replace the older version. The manual approach to dependency hell I guess, of sorts. Lots of little things that will keep you from succeeding. It's probably time better spent learning an existing package management system IMO. Than to create your own. Especially if you're on your own and not part of team. But it's almost all open source so if you can read the source, everything that you need to know is there in one form or another. - James On 6/20/11, David Hendersondhender...@digital-pipe.com wrote: On 06/20/2011 11:52 AM, Pierre Lorenzon wrote: Hi, From: David Hendersondhender...@digital-pipe.com Subject: Re: [Alsa-user] First post Date: Sun, 19 Jun 2011 15:28:48 -0400 Thanks for the reply Pierre. I checked into the blfs book, but it merely says these five chapters will cover alsa and then gives you a basic type configure make. This is obviously
Re: [Alsa-user] First post
A) If you want to overwrite your existing distro's versions, you probably want the --prefix=/usr option on your ./configure commands. If not, be sure to change your $PATH to look at /usr/local FIRST. B) Compile alsa-lib first, alsa-driver second. Most compile options only need --prefix=/usr if you want to override the default of /usr/local. But alsa-driver requires extra parms depending on what you want. Some packages are only tool sets, so make -f Makefile? And use them from where you made them, or copy/move them to more common $PATH's. C) You might have versioning conflicts depending on what you're trying to mix and match. libc and other things might not work well together unless you're running the latest and greatest of every component. And even that is problematic some of the time. D) unless you have a lot of time to waste, or just need the learning, I'd recommend going with existing distros. There's enough of them that one might suit your current needs. www.distrowatch.com HTH, - James On 6/19/11, David Henderson dhender...@digital-pipe.com wrote: Thanks for the reply Pierre. I checked into the blfs book, but it merely says these five chapters will cover alsa and then gives you a basic type configure make. This is obviously not going to answer the questions below. :) Any other thoughts? Dave On 06/19/2011 11:22 PM, Pierre Lorenzon wrote: Hi, It looks like to me such questions are well answered in the blfs book. I personnaly think that the latter is a very good tool to build his own custom distro. Bests Pierre From: David Hendersondhender...@digital-pipe.com Subject: [Alsa-user] First post Date: Sun, 19 Jun 2011 14:41:08 -0400 Hi everyone! I'm currently expanding my knowledge of GNU/Linux to include building packages from scratch towards an overall goal of a custom distro. So far, I have a nice base for a command line OS, but want to expand into the multimedia aspect. Alsa was my first (only?) choice for the audio portion, but I'm running into problems. The alsa site is somewhat overwhelming to newbies and is easy to get lost. I have a few questions below from which I hope I can find help. All contributions are greatly appreciated. :) Thanks, Dave 1) Currently I have downloaded alsa-driver, alsa-lib, and alsa-utils packages. Is there an order in which these packages need to be compiled and installed? 2) I'm currently running the relatively new Linux kernel 2.6.33 so do I need the alsa-driver package? 3) I've been able to successfully compile the alsa-lib package and install it in the custom distro. When I try to compile the alsa-utils package, I constantly get the error: checking for libasound headers version= 1.0.16... not present. configure: error: Sufficiently new version of libasound not found. I'm actually using an existing Kubuntu installation to build the packages for my custom distro. As a result, after I compiled the newer alsa-lib, I didn't install the package into the Kubuntu OS, but rather a staging directory (/opt/staging/alsa). I'm sure the reason this is failing is because it's probably looking for /usr/lib/... or some other default location. How do I tell the configure script for the alsa-utils to look in the staging directory for the header files it needs? -- 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 -- 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 -- 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] First post
--prefix is a ./configure option. If you're going to apply the new alsa to an existing distro kernel and not a custom from source one. You'll likely need to install the kernel-headers package for that kernel and distro. And may need to manually move the old version of alsa (or remove). Plus that whole depmod thing. $ dpkg -l '*kernel*headers*' Which resolves to linux-kernel-headers in debian. Which is a psuedo package for: linux-libc-dev 2.6.26-26lenny3 and of course 2.6.26-26lenny3 resolves to linux-tree-2.6.26lenny3 so: # apt-get install linux-libc-dev linux-tree-2.6.26-26lenny3 (in debian 5.0 / lenny) If it's a custom one, just don't make clean after making the kernel. It should reside in /lib/modules/`uname -r`/build/ or something like that. BITD, this would just be a symlink to/from /usr/src/linux and was what early alsa assumed by default. Depending on what multimedia features you need. You might want --with-sequencer=yes and --with-oss=yes and a --driver=your card options on your alsa-driver compile. Without those =no might be assumed. And you might compile ALL drivers which could take a really long time. Less so these days, but BITD, the better part of a day it seemed. It really depends on what you want interacting with your sound card. Timidity and other synth like software requires the --with-sequencer=yes if your card doesn't have native midi abilities (most don't these days). And various pulse-audio and browsers and other things that just need --with-oss=yes or things might not work as expected, if at all. Little things that you'll find out one way or another as you learn your way around. HTH, - James On 6/19/11, David Henderson dhender...@digital-pipe.com wrote: Hi James, thanks for your help too. :) I'll provide replies in the same fashion given. A) I don't want to overwrite the Kubuntu installation files as I'm compiling this version of alsa for my own distro. I would prefer to use Kubuntu's pre-packaged software within itself. So since the compiled version of alsa will be going into /opt/staging/alsa, should I include --prefix=/opt/staging/alsa as the parameter to configure? B) I'll assume at this point, that no matter what version of the Linux kernel is being used, it's still required to install the alsa-driver package. That being said, I'm going to run into the same problem as A above since the version of Kubuntu I'm using to build the custom distro isn't using the same kernel version. So what configure option do I have to pass in order for alsa to see the source code of the custom distro's kernel version? C) So far, so good, but I'll keep that in mind. :) D) Thanks for the URL, but this is a project that I've wanted to do for the last 5-7 years and now I have the ability to do so. Not only that, but knowing details at this level of building an OS can also help with my job - so I get a two fold benefit. :) Otherwise, I'd definitely follow your advice! lol Thanks again for your help, I look forward to hearing back from you. Dave On 06/19/2011 04:36 PM, James Shatto wrote: A) If you want to overwrite your existing distro's versions, you probably want the --prefix=/usr option on your ./configure commands. If not, be sure to change your $PATH to look at /usr/local FIRST. B) Compile alsa-lib first, alsa-driver second. Most compile options only need --prefix=/usr if you want to override the default of /usr/local. But alsa-driver requires extra parms depending on what you want. Some packages are only tool sets, so make -f Makefile? And use them from where you made them, or copy/move them to more common $PATH's. C) You might have versioning conflicts depending on what you're trying to mix and match. libc and other things might not work well together unless you're running the latest and greatest of every component. And even that is problematic some of the time. D) unless you have a lot of time to waste, or just need the learning, I'd recommend going with existing distros. There's enough of them that one might suit your current needs. www.distrowatch.com HTH, - James On 6/19/11, David Hendersondhender...@digital-pipe.com wrote: Thanks for the reply Pierre. I checked into the blfs book, but it merely says these five chapters will cover alsa and then gives you a basic type configure make. This is obviously not going to answer the questions below. :) Any other thoughts? Dave On 06/19/2011 11:22 PM, Pierre Lorenzon wrote: Hi, It looks like to me such questions are well answered in the blfs book. I personnaly think that the latter is a very good tool to build his own custom distro. Bests Pierre From: David Hendersondhender...@digital-pipe.com Subject: [Alsa-user] First post Date: Sun, 19 Jun 2011 14:41:08 -0400 Hi everyone! I'm currently expanding my knowledge of GNU/Linux to include building packages from scratch towards an overall goal of a custom distro. So far, I have
Re: [Alsa-user] Help with ALSA on new computer
It depends on the HDMI device. For my video card, the specification of the sound that travels over that wire is pretty strict. ONLY AC3, only 44.1kHz, only stereo / 2 channels, only... And it does work if all criteria is met. But I much prefer to use the analog audio (lossless / PCM). But a single wire does have it's uses. I imagine that it's in your configuration somewhere to make what you're wanting happen. .asoundrc? pauvcontrol? Just be aware of the specifications of your gear. You have two of them technically, sending and receiving, and they may not match, but on the lowest of lowest common denominators. At least change the indexing so that your HDMI card is card 0, that way even the stupid apps try to use it, versus some other card. - James On 6/16/11, Jerry Geis ge...@pagestation.com wrote: I got a new computer (Zotac HD-ID40) ION2 I have installed alsa 1.0.24 on Centos 5.6 x86_64. Not sure why it registers 2 audio devices either. here is info: lspci | grep Audio 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) 03:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) lspci -n 00:1b.0 0403: 8086:27d8 (rev 02) 03:00.0 0300: 10de:0a64 (rev a2) 03:00.1 0403: 10de:0be3 (rev a1) lspci -v 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Subsystem: ZOTAC International (MCO) Ltd. Device a140 Flags: bus master, fast devsel, latency 0, IRQ 10 Memory at fe9fc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel 03:00.0 VGA compatible controller: nVidia Corporation GT218 [ION] (rev a2) (prog-if 00 [VGA controller]) Subsystem: ZOTAC International (MCO) Ltd. Device 3100 Flags: bus master, fast devsel, latency 0, IRQ 5 Memory at fd00 (32-bit, non-prefetchable) [size=16M] Memory at d000 (64-bit, prefetchable) [size=256M] Memory at ce00 (64-bit, prefetchable) [size=32M] I/O ports at ec00 [size=128] Expansion ROM at fcf8 [disabled] [size=512K] Capabilities: [60] Power Management version 3 Capabilities: [68] MSI: Enable- Count=1/1 Maskable- 64bit+ Capabilities: [78] Express Endpoint, MSI 00 Capabilities: [b4] Vendor Specific Information: Len=14 ? Kernel driver in use: nvidia Kernel modules: nvidiafb, nvidia 03:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) Subsystem: ZOTAC International (MCO) Ltd. Device 3100 Flags: bus master, fast devsel, latency 0, IRQ 5 Memory at fcf7c000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] Power Management version 3 Capabilities: [68] MSI: Enable- Count=1/1 Maskable- 64bit+ Capabilities: [78] Express Endpoint, MSI 00 Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel How can I get sound over HDMI working on this computer? Analog sound out works fine. I have edited the /etc/asound.conf to point to the device 1,3 which is the HDMI audio. Thanks, Jerry -- 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 -- 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] [SPAM] No Sound in Debian 6
Finding out what's different with more detain than Yes and No would help. boot into knoppix $ lsmod | grep -i snd | sort 21 | tee alsa_knoppix.log boot into debian $ lsmod | grep -i snd | sort 21 | tee alsa_debian.log save these files on a common medium (flash drive) of course. $ diff -a -U 3 alsa_knoppix.log alsa_debian.log And of course noting /proc/asound/cards plus you might want to modinfo the module of importance for each, to see what differs there (versioning / parms). And various other things generally covered in the alsa-info.sh script. HTH, - James On 5/23/11, s.keup...@arcor.de s.keup...@arcor.de wrote: Hello! I'm afraid the problem is totally not linked to the Debian Squeeze distro, reaosn: there are similar problems for some chipsets, especially the famous Intel HDA Audio chipset in Ubuntu: ... Note: I'm not telling that the problem happens for all chipsets, but tons of persons should probably encouter such a problem when using laptops and netbooks fitted with Intel hda chipsets. It may not be a problem of distributions, however, everything worked with the alsa packages and configuration that come with Ubuntu, but not with those which come with Debian. At this moment I overbridge the problem on a absurd virtual way: external speakers, there is a vol-button you can rise up. External speakers are not working for me. @all: I think I've found the root of my problem: I booted the computer with a Knoppix Flash drive, with which all sound output and input was fine. While checking the /proc/asound/cards of the Knoppix system, I noticed an Sigmatel Audio Card, additional to the HDA Intel entry. For the same Sigmatel Chip there werde Channels displayed in Alsa mixer, which seemed to actually control my volume. This Sigmatel entry is missing on my Debian System. Does anyone know how to add this card to my alsa? Best Regards Savio -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] [SPAM] No Sound in Debian 6
A little overkill from my description. And so forget that versioning would alter the module sizes. But alsa-info has the needed info. Knoppix - Kernel 2.6.37 - alsa 1.0.23 Debian - Kernel 2.6.32 - alsa 1.0.24 Is that the way your debian came, or did you try to fix things manually? Just an odd versioning combo AFAIK. The the older debian kernel would have the newer alsa version. Other notes. Aside from that WTF modinfo null. stuff for debian. knoppix - Codec: SigmaTel STAC9205 debian - Codec: Conexant ID 2c06 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1b.0 0403: 8086:284b (rev 02) Subsystem: 1028:01f1 So the knoppix one is the one that works? If so: modprobe snd-hda-intel id=SigmaTel ??? Or add snd-hda-intel.id=SigmaTel to the kernels boot line in grub (or lilo). Or something like that, it's been a while. Syntax might vary. And it might need to be in the kernel's .config if it's not already. Unless you compile alsa / install the old 2.4.x way. Beyond that your guess is as good as mine. Probably some alias / options line you can add to /etc/modprobe.d/*alsa* as well, or instead of those two other ways, that functionally do the same thing (with quirks). Hopefully something in there rings a bell for you. Or else the old alsa-project.org and doc stuff might hint towards a solution. On the surface it looks like debian is defaulting to a conexant codec and failing and knoppix is defaulting to a sigmatel codec and NOT failing. Which is the same module / driver for all intents, so something configuration is awry. Or I could be wrong. - James On 5/23/11, s.keup...@arcor.de s.keup...@arcor.de wrote: Hello, Thank you. These are the differences of the two logs: http://pastebin.com/nAtnhMvc As I am not a really experienced user, I do not know how to add and configure the modules properly. It would be nice if you would explain this in more detail. This is the modinfo on the three sound modules installed on Knoppix: http://pastebin.com/ADknBwRt And this is the output of the alsa-info skript you mentioned - On Knoppix: http://pastebin.com/fHCguzr6 , On Debian: http://pastebin.com/0rAPiyEN . Best Regards Savio PS: I hope the Pastebin Links are ok. I just wanted the mail not get too messy. PPS: As seen in the files, I was wrong saying the there would be a Sigmatel card on Knoppix in /proc/asound/cards. However a Sigmatel codec is used on Knoppix. Finding out what's different with more detain than Yes and No would help. boot into knoppix $ lsmod | grep -i snd | sort 21 | tee alsa_knoppix.log boot into debian $ lsmod | grep -i snd | sort 21 | tee alsa_debian.log save these files on a common medium (flash drive) of course. $ diff -a -U 3 alsa_knoppix.log alsa_debian.log And of course noting /proc/asound/cards plus you might want to modinfo the module of importance for each, to see what differs there (versioning / parms). And various other things generally covered in the alsa-info.sh script. HTH, - James -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Spikes when recording
That seems pretty regular at 8 to 10 minute intervals. Do you live near a subway line? Or other electric mass transit option? Is the computer on a UPS or power conditioner type supply line? I get a spike like that when I use a battery box to power an electret mic. If I turn it on after pressing record. Beyond that, your guess is as good as mine. - James On 5/20/11, Peter Hoffmann p...@peter-hoffmann.com wrote: Hello, I'm recording audio 24/7 with a delta 1010 sound card and have a strange problem: Every night at 2:30 I get spikes and some inaccurancy within some seconds in a one hour length recording. I've upload a screenshot to illustrate the problem: http://img88.imageshack.us/img88/4562/spikesh.png My .asoundrc pcm.capt { type dsnoop ipc_key 223456 slave { pcm hw:0,0 rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } } pcm.c1 { type plug ttable.0.0 1 slave.pcm capt } pcm.c2 { type plug ttable.0.1 1 slave.pcm capt } pcm.c3 { type plug ttable.0.2 1 slave.pcm capt } pcm.c4 { type plug ttable.0.3 1 slave.pcm capt } pcm.c5 { type plug ttable.0.4 1 slave.pcm capt } pcm.c6 { type plug ttable.0.5 1 slave.pcm capt } pcm.c7 { type plug ttable.0.6 1 slave.pcm capt } pcm.c8 { type plug ttable.0.7 1 slave.pcm capt } I'm recording with arecord -q -f cd -t wav -d 3600 -c 1 -D c1 out.wav. Any hints where the problem might be? Kind Regards, Peter Hoffmann -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] very low-level volume in both Debian and Ubuntu
I agree that sound should just work. And I'm still kind of surprised that a) we have to configure it with a text file. And b) twenty years later, that's still the case for the most part. Not that I think that we should give up the command line even in part. Alsamixer isn't intuitive, but it is semi-user friendly IMO. It's far from perfect. But you launch it with alsamixer and exit with the escape key (aka boss key). Cursor up is up in volume, cursor down is down in volume. The M key for mute and unmute. The tab key to switch between playback and capture is not that intuitive. It should probably default to ALL IMO. But it does have the typical F1 help screen. Although I'm not sure of the accessibility options at this time. There does appear to be a bug report filed on it. https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/430937 Although that dates back to 2009 and karmic. And the fix seems to be to install gnome-alsamixer and turn up master F. I still think that alsamixer is the route to fix it and it's just a level setting. http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=utils/alsa-info.sh $ sh alsa-info.sh And the output (in /tmp) for that one should give you information on your card. Take note of the mixer part for Master. I've used aumix in the past. But if you have more than one card or other things, it can be wrong / useless more often than not. Although it's still the only way to tell some soundcards to record from PCM out IME. I've never found a way to affect that setting in any other way, in the manner needed. Even though I can see the effect of that change in the output of amixer. HTH, - James On 5/20/11, Y P yellowpeng...@edpnet.be wrote: Hello James: On Thu, May 19, 2011 at 09:59:01AM -0500, James Shatto wrote: The first step would be to see if it's even an ALSA issue. I had to say first of all I'm not using Flash nor graphical mixers since I'm a VIP - vision impaired person; sometimes I use the Orca screen reader but my current/daily usage of Gnu/Linux OSes is command-line. Sorry if I forgot to precise this before. With flash video (youtube) there's a speaker icon and a slider which affects the volume. I recently noticed hulu had my levels way low with such an icon. With mplayer there's a softvol option which might differ from the levels set in alsa. Yes but you can't go louder than the maximum of volume-level, so the problem remains. And of course alsamixer to actually set your levels. I'm not using any alsamixer, it is not user-friendly so I prefer adding oss-compat, libsox-fmt-all and aumix, and adjust then the volumes -v -w -W -s at the commandline. I'm afraid it is really an ALSA/Pulse problem: I just googled with the keywords ALSA+Pulse+netbook+very+low+volume+output+problem and Google gave me about 7 screens of results. The reasons why I believe the problem is really an ALSA+Pulse issue are: - one of the Google results talks about an upgrade from an Ubuntu 9.10; I've got a volume-problem since upgrading my Ubuntu Intrepid just a few weeks ago, but before the release of the newest / latest Natty (11.04); so if some other people encounter the problem with the same Intel hda chip in Ubuntu 9.10 (Maverick) I had probably upgraded to the problem while I hadn't it before ? - from the Google results I see that the problem is not bound to one specific distribution, that explains why the problem also occurs in the Debian Squeeze; there is also a low level on headset, so the problem is really bound to sound output, not to the distribution. technically nothing is broke and nothing needs fixing. If the levels are set and maxed out and the problem persists, then it could be an alsa issue. I will probably try to install a fresh Squeeze on my Hercules eCAFE; at this moment the output-volume of my 11.04 is normal, loud is loud, maxed is quite too loud! The major difference with the freshly installed Debian on my EEE where I've got Ubuntu Intrepid before is, that maxed the volumes are stil too low, impossible to stream radio and puting your machine as background-radio. Although I'd extract the audio content being played and look at it in audacity to see if it's not just the content to verify the potential source. I'm not using audacity at all since that tool is graphical. The difference I noticed you can hear it at boot time: before the problem I was able to hear clearly the Ubuntu tamtam at gdm login, at this moment I hear the Espeak voice in Debian's gdm login very very far away, it's unusable ! IMHO there is a very important crucial bug happend a few versions ago and that causes a volume difference of 32 dB or probably much more. Just a power user here and nothing really current version wise on my end to have that issue or know much about it myself. But it'd be nice to know how to fix it, if I do run into it. I will continue to surf and have a look around to fix it, but I'm
Re: [Alsa-user] vert low-level volume in both Debian and Ubuntu
The first step would be to see if it's even an ALSA issue. With flash video (youtube) there's a speaker icon and a slider which affects the volume. I recently noticed hulu had my levels way low with such an icon. With mplayer there's a softvol option which might differ from the levels set in alsa. And of course alsamixer to actually set your levels. Make sure those are appropriate for what you're trying to do. If they aren't, then technically nothing is broke and nothing needs fixing. If the levels are set and maxed out and the problem persists, then it could be an alsa issue. Although I'd extract the audio content being played and look at it in audacity to see if it's not just the content to verify the potential source. Just a power user here and nothing really current version wise on my end to have that issue or know much about it myself. But it'd be nice to know how to fix it, if I do run into it. HTH, - James On 5/19/11, Y P yellowpeng...@edpnet.be wrote: Hello, escuse me but I'm asking myself if the problem I encounter regarding very low level of volumes is due to Alsa/Pulse : a few weeks/maybe a month ago, I upgraded my EEE netbook's Ubuntu O S; the result is : no longer a normal level volume; last weekend I did an installation of Debian on a EEE netbook, same result : very low level output volume from the internal EEE speakers. Finally today a good friend told me he did an installation of an Ubntu on a PowerMac, idem same problem regarding sound : very low level. How can I test the origins of this problem to be sure which is the source of the problem ? I did also had a problem in using an eCAFE EC-900 as DJ due to a very annoying bug: plugging a minijack in it won't help to cut the output to the internal speakers. Can someone tell me if one of these problems or both are being fixed soon ? sound is very important, appology for all these remarks! please fix them if possible! Grtnx, Y)ellow P)enguin -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Newbie query about interfaces
It depends on what you need. A majority of the cheap USB interfaces are USB 1.x and only do 2 channels. USB 2.x only recently got an audio standard ( 5 years) and devices that are starting to use that. With USB 3.x already being out in the wild of sorts. IMO, if you need more than 2 channels, you're better off with a firewire device (for now). And yes, they cost a good chunk of money as many of them include microphone preamps. At $$$ per channel. Unless you already have gear than can deliver 5.1 input over optical cable, you're probably going to have to chunk out some change. Even at $50 a channel, 4x channels is $200-ish. If you have line level inputs and don't need microphone preamps, you might have a few options. A used Delta 44 (PCI) runs about $100 USD on craigslist and is fairly well supported driver wise. Although pulse audio still kind of sucks at a default configuration for it. And various versions of alsamixer seem to disagree with the hardware specs more often than not. Bit it's 4 line level inputs and 4 line level outputs (24/96) on a budget and works fine under linux. But it depends on the budget. It's 1/4 connectors ONLY. Insert external microphone preamp(s), and external headphone preamp(s) to use it like most OTHER interfaces, plus cables and adapters and whatnot. If you don't already have a lot of that stuff. You're going to be looking at some $$$. Or squiggly LLL in your case. Not that I see how any of this is an alsa issue. Until you have questions on a specific device. There's other websites with forums that discuss various interfaces and whatnot. I have an M-Audio Mobile Pre (USB 1.x, 2 channels in, 2 channels out) and it works fine under linux with alsa (usb class compliant). I also have a Delta 44 and it works fine, with a little extra configuration in some cases. HTH, - James On 5/1/11, Graham Dicker graham.dic...@antecor.com wrote: Dominique Michel wrote: Le Fri, 29 Apr 2011 16:28:47 +0100, Graham Dicker graham.dic...@antecor.com a écrit : I have been recording for many years with a Yamaha digital 4 track recorder. I would now like to switch to using my Suse Linux minitower using Ardour. I am not sure though what kind of audio interface I need to buy. Ideally I would like to have similar capability to what I am used to with my 4 track i.e. up to four instruments being recorded simultaneously, each on to it's own track. I understand that an interface like the Yamaha Audiogram 6 will work but it's not clear that I will be able to route each instrument on to it's own track. Is that what it does? Or do I need something else? If the audiogram 6 is recognized by alsa, it will work. Beside ardour, you will need jack-audio-connection-kit (jack) and some alsa mixer like the alsamixer. jack have several GUI like qjackctl. with it, you can route the audio channels as you want to. Ciao, Dominique Thank you for responding. I already have all the software working using the motherboard audio interface and have been using it to record stuff for a few months. But I can only do two mono tracks or one stereo track at a time - a painfully slow process. I have looked around for an interface with more inputs and find they vary from around £80 to £3000 or more. The Audiogram 6 is within my budget, I don't want to spend more than that. The descriptions on the vendor websites for this and similar units in this price bracket don't mention simultaneous recording on multiple tracks and as far as I can tell they are just a kind of mixer, and can only record one track at a time (on any OS). Is that true? If so, what kind of interface do I need? Thank you Graham Dicker -- WhatsUp Gold - Download Free Network Management Software The most intuitive, comprehensive, and cost-effective network management toolset available today. Delivers lowest initial acquisition cost and overall TCO of any competing solution. http://p.sf.net/sfu/whatsupgold-sd ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- WhatsUp Gold - Download Free Network Management Software The most intuitive, comprehensive, and cost-effective network management toolset available today. Delivers lowest initial acquisition cost and overall TCO of any competing solution. http://p.sf.net/sfu/whatsupgold-sd ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] overrun with 'arecord' - why ?
Low latency kernel? xruns are basically a resource issue. Web browsers have flash and java and javascript scripts that loop for infinity and other things that strip you of your resources. Basically I'd start by closing your browsers while recording. If you're trying to capture content from the browsers, there's other ways to accomplish that IMO. Without having to sample the output from a soundcard on a soundcard. Otherwise when browsers access sound, it's generally the old OSS way (/dev/dsp). A few extras like java and flash have gotten smart about more modern ways, but not all sounds from a web browser are triggered by those methods. And not all versions of those things are smart about it. Alternatively give your audio priority consideration in /etc/security/limits.conf. Check /proc/asound/ for information while recording. You might need to tweak period size or other things. Audio might need priority in other ways. Which could mean changing the nice level of the recording application, or the nice level of most everything else. I've gotten in the habit of running povray and ffmpeg conversions with nice -n 19, just so I can still do other things while they run. Otherwise they all run at the same nice level and do battle over who's more important. Which is not an environment you want your realtime recording application(s) to be in. Basically one thing at a time. If arecord is your recording method, and what you're recording doesn't need a gui, you might try running without X and see if you still have xruns. If you must, slowly add things back like X, until it breaks to know where your limits are. AFAIK, arecord is a single threaded application. So if it's not being run on the not used CPU(s), and the one that it is on is maxed... Not that I'd know how or IF you can choose CPU per task for non SMP aware applications. - James On 4/8/11, Sergei Steshenko steshenko_ser...@list.ru wrote: Hello, I've tried to run 'arecord' as part of simultaneous playback + capture rig (for acoustic measurements) and noticed overruns. So, even plain single 'record' occasionally produces overruns: arecord -D hw:0,2,0 -c 2 -r 96000 -d 6 -f S32_LE recorded.wav Recording WAVE 'recorded.wav' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo overrun!!! (at least 855507586.521 ms long) . So, my question is: Why ?. It's a 2.6Ghz machine with SATA disks. Two cores, web browsers are the most active tasks (nothing fancy, no sound activity on the side of the web browsers). Effectively one core is free. Any ideas ? The gear: Card: HDA NVidia Chip: Realtek ALC883 . Thanks, Sergei. -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] overrun with 'arecord' - why ?
No, I'm not trying to capture content from browsers; the browsers have no relationship to what I'm doing. If it's running on the same computer at the same time, there is a relationship. i.e. Fewer resources. An xrun is a lack of resources. (or a bug) arecord -D hw:0,2,0 -c 2 -r 96000 -d 6 -f S32_LE recorded.wav So the sox variant you're using is? rec -s -4 -L -c 2 -r 96000 recorded.wav trim 00:00:00 00:00:06 Have you tried arecord without -D ? And/or with -t wav - James -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] overrun with 'arecord' - why ?
Please reread my message in this thread on 'sox' - it contains the complete command line I've used. So apples to oranges? since your sox only does 4 seconds (trim 1 5) and your arecord does 6 seconds -d 6. Statistically that's 50% more opportunity for failure in arecord. - Did omitting -D help? did adding -t wav help, so it doesn't have to assume stuff (or not) based only on file extension? milliseconds are what? 1/1000 of a second. So 855507586.521 is about 85,550 seconds. Or roughly 1,426 minutes or roughly 23 hours and 46 minutes. Kind of odd for a 6 second capture don't you think? Or is the result and the example unrelated? - James -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] 50 year old male with no sound coming out of his speakers
http://ubuntuforums.org/showthread.php?t=286016 does that one help? Appears common to need to do a card reset for some reason. If that doesn't work, you might try the snd-hda-intel driver, versus the snd-intel8x0 that it says you're using. I don't know which of those drivers go with that card. And google hits are varied. Just a user as far as alsa goes. And don't have that particular card on anything of mine. - James On 4/5/11, jida...@jidanni.org jida...@jidanni.org wrote: Gentlemen, I cranked everything up but still not an ounce of sound. My ALSA information is located at http://www.alsa-project.org/db/?f=823f89190858a6673ec0075c004db6de86c7495b Yes I connected headphones to the green jack and ran speaker-test(1). Doing the same on a different computer one hears static, but on this computer -- silence. You know what would be really neat, if there was something in /proc that could show that yes, there really is something plugged into that 3.5mm jack socket, as there is a change in resistance ohms, showing that the what looks like it is soldered to the board really is. -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Troubleshooting M-Audio Delta 44 (ICE1712)
Try: speaker-test -c 2 -D hw:0 where hw:# is the number of your card as it shows in /proc/asound/cards. snd-ice1712 is the driver. With a Delta 44 myself. Pulse-audio doesn't play nice with it, so disable that if reasonable. Most apps I use interface with alsa or jackd directly, so I just leave pulse audio as it came with ubuntu. Semi working, but mostly annoying. Complete with re-indexing the alsa drivers so that ice1712 is card 0 to get it working initially. Even if pauvcontrol says my only output option is dummy out or HDMI out from my video card (snd-hda-intel). But probably not your issue, so I'll stop rambling. - James On 4/2/11, James P. Early earl...@gmail.com wrote: Greetings everyone, I'm writing to request assistance with getting my M-Audio Delta 44 (ICE1712) functioning under a fresh Gentoo installation. At first, I thought I had an issue with jackd, because I could not get it to start. I tried many configuration options, but typically got ALSA poll time out messages, then jackd would die. So, I decided to examine my ALSA installation more carefully. Today, I decided to see if I could just get a sound to play using aplay (from the command line without starting X11). What I notice is that aplay just freezes at some point while processing the input wav file. I get no sound, and I have to Ctlr-C to stop it. Here's an example: === $ aplay -vv /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 - 0 1 - 0 2 - 0 3 - 0 4 - 0 5 - 0 6 - 0 7 - 0 8 - 0 9 - 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat: STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 6144 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min: 1024 period_event : 0 start_threshold : 6144 stop_threshold : 6144 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat: STD channels : 10 rate : 48000 exact rate : 48000 (48000/1) msbits : 24 buffer_size : 6144 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min: 1024 period_event : 0 start_threshold : 6144 stop_threshold : 6144 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Hardware PCM card 0 'M Audio Delta 44' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat: STD channels : 10 rate : 48000 exact rate : 48000 (48000/1) msbits : 24 buffer_size : 6553 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min: 1024 period_event : 0 start_threshold : 1 stop_threshold : 7378022089539715072 silence_threshold: 0 silence_size : 7378022089539715072 boundary : 7378022089539715072 appl_ptr : 0 hw_ptr : 0 + | 46%^C Aborted by signal Interrupt... == You'll notice in this example that it has frozen at the 46% mark. Other files may stop at different points. All indications (aplay, lspci, /proc/asound, alsamixer, etc.) are that the card is being recognized, and that the associated kernel module is loaded. I've tried a number of sites for answers including: http://alsa.opensrc.org/TroubleShooting http://www.alsa-project.org/main/index.php/Matrix:Module-ice1712 http://www.gentoo.org/doc/en/alsa-guide.xml ... and countless forum posts I have run alsa-info, and the result can be seen here: http://www.alsa-project.org/db/?f=0703a7d0bd8da7a7b4bc387e107900acd0674a7b I would greatly appreciate hearing from anyone with information about further testing or configuration. Thanks, ~Jim -- 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 -- Create and publish websites with WebMatrix Use the most
Re: [Alsa-user] MobilePre USB support
I have the mobile pre (old one, but not the oldest one). It just works. USB compliant, at least for USB 1.x standards. i.e. 2 channels input, 16 bit, 48kHz max. The gray one with buttons on front, and pretty much any analog connection type known to man. Although the line input (3.5mm) does not provide the plug in power for cheap-ish / camcorder type mics. And the phantom power is known to be a bit under volt, but good enough for most mics IMO. Worked out of the box for me. But there is some wonkyness in Debian Lenny(5.0) with freezes (wasn't an issue in sarge). But then again mine is old enough that the blue LED light comes and goes. And I'm running the distro supplied kernel 2.6.26. As some of the changes in the current version of things makes it a lot more difficult to run a custom kernel. At least with old school ways and a working config from a previous install. I may upgrade to 6.0 as soon as some of my current projects are wrapped up. But for $80 off of craigslist, I'm not complaining. - James On 3/15/11, Jonathan Wilkes jancs...@yahoo.com wrote: Hi, I'm looking for a relatively inexpensive USB audio interface that will work (painlessly) on a laptop running Gnewsense, which would basically mean either Hardy or Squeeze. After some looking and reading I'm thinking about something like Maudio Fast Track (one with the knobs on the top) Fast Track Pro or MobilePre (one with the knobs on the top). Can anyone tell me which of these works the most reliably with ALSA? Or if there's some other interface in this price range that has an XLR input and works as well as the USB Transit seems to do? I basically want to record into Ardour or Pure Data, and monitor with headphones. Thank you, Jonathan -- Colocation vs. Managed Hosting A question and answer guide to determining the best fit for your organization - today and in the future. http://p.sf.net/sfu/internap-sfd2d ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Colocation vs. Managed Hosting A question and answer guide to determining the best fit for your organization - today and in the future. http://p.sf.net/sfu/internap-sfd2d ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] MobilePre USB support
AFAIK, the old old one is white-ish and looks like a fallback to the 1950's. And AFAIK, that is the only difference. The Fast Track Pro is probably the more liked modern one (USB 2.x?). Although I don't know of it's linux status. Should be fine IMO, if it's class compliant. It took a good long while, but there is a 2.0 audio standard now. When recording you want 24 bit IMO. 16 bit is more of a delivery format. 24 bit gives you more dynamic range, which better suits recording IMO. About all I use my Mobile Pre for is laptop sound. Or if I need to archive an odd format like reel to reel tapes or cassettes (judges tapes). I have a Delta 44 (24/96) on the desktop and a Korg MR-1000 (24/192 or DSD) for anything more serious. Except for the DSD part, all linux compatible as well. Although the Korg only functions as a usb storage device as far as a computer is concerned. M-Audio tends to use the same ADC/DAC chips in most of their gear, so it's a fairly safe bet IMO. Safe-er than some other options anyway. Although not that configured by default in things like pulse audio and such. But the driver(s) work, always have IMO. Some of the mixer stuff can be a little off. But I'm not exactly running the latest and greatest of everything. Most of my hardware is sufficiently old that I don't need to in most cases. - James On 3/15/11, Jonathan Wilkes jancs...@yahoo.com wrote: Thanks, James. From a quick google search it looks like one can still get these. But just to make sure I'm talking about the same one-- what's the difference between the oldest one you referred to and the one you've got? And has anyone had success with the new shiny little one with top knobs? Thanks, Jonathan --- On Tue, 3/15/11, James Shatto wwwshad...@gmail.com wrote: From: James Shatto wwwshad...@gmail.com Subject: Re: [Alsa-user] MobilePre USB support To: alsa-user@lists.sourceforge.net Date: Tuesday, March 15, 2011, 2:04 PM I have the mobile pre (old one, but not the oldest one). It just works. USB compliant, at least for USB 1.x standards. i.e. 2 channels input, 16 bit, 48kHz max. The gray one with buttons on front, and pretty much any analog connection type known to man. Although the line input (3.5mm) does not provide the plug in power for cheap-ish / camcorder type mics. And the phantom power is known to be a bit under volt, but good enough for most mics IMO. Worked out of the box for me. But there is some wonkyness in Debian Lenny(5.0) with freezes (wasn't an issue in sarge). But then again mine is old enough that the blue LED light comes and goes. And I'm running the distro supplied kernel 2.6.26. As some of the changes in the current version of things makes it a lot more difficult to run a custom kernel. At least with old school ways and a working config from a previous install. I may upgrade to 6.0 as soon as some of my current projects are wrapped up. But for $80 off of craigslist, I'm not complaining. - James On 3/15/11, Jonathan Wilkes jancs...@yahoo.com wrote: Hi, I'm looking for a relatively inexpensive USB audio interface that will work (painlessly) on a laptop running Gnewsense, which would basically mean either Hardy or Squeeze. After some looking and reading I'm thinking about something like Maudio Fast Track (one with the knobs on the top) Fast Track Pro or MobilePre (one with the knobs on the top). Can anyone tell me which of these works the most reliably with ALSA? Or if there's some other interface in this price range that has an XLR input and works as well as the USB Transit seems to do? I basically want to record into Ardour or Pure Data, and monitor with headphones. Thank you, Jonathan -- Colocation vs. Managed Hosting A question and answer guide to determining the best fit for your organization - today and in the future. http://p.sf.net/sfu/internap-sfd2d ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Colocation vs. Managed Hosting A question and answer guide to determining the best fit for your organization - today and in the future. http://p.sf.net/sfu/internap-sfd2d ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Colocation vs. Managed Hosting A question and answer guide to determining the best fit for your organization - today and in the future. http://p.sf.net/sfu/internap-sfd2d ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net
Re: [Alsa-user] Help configuring HDSP9632
In theory you don't need the dmix thing anymore. If your applications use ALSA natively, it will automagically mix sound from several applications (in software). If the applications use OSS, you can force it to use alsa with aoss. BITD you'd run esddsp or artsdsp -m app to do this sort of thing. Depending on the sound daemon of choice you happened to be using. These days those daemons just get in the way, chew up resources and cause XRUNs or other woes. If your applications are configured to use ALSA, this should be a non-issue. Assuming that you're running something current and not RH 5.1 from some book or something. In the few times that I tried to use dmix BITD, it was generally the cause of problems, not the solution. If your applications use OSS and you don't launch them with aoss, then they will lock the device (per days of old). I'm not sure if that's addressed with oss emulation or not. And aoss isn't perfect as something like a browser will launch pop ups that are NOT launched with aoss and break the very thing you were trying to avoid. Mostly problematic with internet gaming where the games are pop ups. But for most other application you can select the audio system of choice. alsa, oss, jackd, artsd, esd, pulse-audio, and probably others. Alsa, having the least overhead IMO, if you're coming up short on system resources. check your .asoundrc and whatever system defaults were created for you or by you in /etc/. I'm not sure of that locations default naming convention as it probably varies between distros. alsa.conf? asound.conf? +/- an /etc/ or /etc/alsa/ or /etc/sound/ or ??? And various tricks of old to delete the asound.state file to force new defaults. Located at /var/lib/alsa/asound.state on my system. YMMV HTH, - James On 2/26/11, Bill Unruh un...@physics.ubc.ca wrote: On Sat, 26 Feb 2011, Friedrich Ewaldt wrote: Hi Matt, I didn't use a RME HDSP9632 for quite a long time (also I never used it with the dmix plugin). However, the dmesg message sounds like a clock source problem. All I can suggest is to check for the correct rate settings, e.g. compare what hdspconf is showing to the output of cat /proc/asound/card0/hdsp --fe Matthew Robbetts schrieb am 26.02.2011 15:18: Hi guys, I've been trying off and on for weeks now, but I can't get my RME HDSP9632 configured under ALSA properly. (Is it me or does ALSA really not make this an easy process? I can't find anywhere to get any feedback from the system on configuration errors. Hell, if you make a typo in the config file, you only find out because of some scary-looking output from aplay. The docs on the website seem to be quite old and often conflict with each other and I can't find any relevant man pages.) Anyhow, the card works out of the box, insofar as it lets one application play sound through it at a time. So I'm trying to do the OK. It works. That finishes alsa. common thing of configuring dmix to let multiple applications output sound at once. Nothing fancy, really! At least, at this point. That in general has nothing to do with the card or the driver of the card. Most cards do not allow multiple inputs to all play at once. It is software. It is often pulseaudio or jack could be used as well. . My /etc/asound.conf file is as follows (pieced together from tuts and the like): pcm.!default { type plug slave.pcm hdsp9632_dmix hint { show on description Default device: Plugs into hdsp9632_dmix. } } ctl.hdsp9632_dmix { type hw card 0 } pcm.hdsp9632_dmix { type dmix ipc_perm 0660 ipc_key 1025 ipc_key_add_uid false slave { pcm hw:0,0 rate 44100 channels 2 period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 } hint { show on description hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. } } Using this file, I get # aplay -L null Discard all samples (playback) or generate zero samples (capture) default Default device: Plugs into hdsp9632_dmix. hdsp9632_dmix hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. which is what I hope for. But, if I try and play something with vlc, I get an error message and No AutoSync source for requested rate comes up in dmesg. The card is currently set to clock master at the same sample rate as the audio (44.1kHz). If anyone can shed any light on what I'm doing wrong (and, ideally, some methodology on configuring ALSA which doesn't require scrabbling around in the dark!), I will be grateful until the end of time. The alsa users documentation has long long long been its greatest shortfall. And noone seems to be stepping up to the plate to write the docs. One of the problems with the open software movement. In a
Re: [Alsa-user] Record 8 separate Line IN Channels from M-Audio Delta 1010 Card
-f cd is a shortcut for a STEREO track. AFAIK, the output for arecord is ONE file, with many channels in it. i.e. -f cd == -f S16_LE -t wav -c 2 -r 44100 and i.e. -f cdr == -f S16_BE -t wav -c 2 -r 44100 (what it gets converted to before burning a disc) or something like that... $ arecord -t wav -f S16_LE -c 8 -r 48000 -D ice1712 All_8_Tracks.wav (would that be ice1724? Dont know, just asking.) sndfile-deinterleave sox audacity ffmpeg and probably others to chunk out each individual track and convert them to mono. Plus/minus on the syntax's, it's been a while and using gray matter only. Aften to create 5.1 ac3 audio. ffmpeg is limited to creating 5.0 iirc. And other quirks for pretty much all of the options. - James On 2/21/11, Sergei Steshenko steshenko_ser...@list.ru wrote: On Mon, 21 Feb 2011 17:37:24 +0100 Peter Hoffmann p...@peter-hoffmann.com wrote: wa (BTW can anyone point me to a tool to split a multi channel wav into a file per channel?) SoX (sox.sf.net), 'ecasound'; 'audacity'. Regards, Sergei. -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound, no /proc/asound/
Well depending on HOW it was obtained. The short answer is that the kernel primarily installs to only TWO locations. /lib/modules/`uname -r`/ and /boot/ So check for the #.##.## of your kernel version in those locations. Also note a few symlinks /boot/config / boot/system /boot/kernel that might link to the #.##.## of your kernel. Not to worry those are handled at your re-install. But there might be an initrd image in there that could linger and not update if you do anything manual-ish that could be a trouble maker. As in could be formed from another version you're not actually using, but the boot loader tries to use it anyway. That one is the primary difference between a custom kernel and a distro kernel in a lot of cases. So the basic procedure might be... $ sudo dpkg --purge --force-all kernel-*version* (might also be some header, image, modules, or other things for that image depending on how the distro packages it. Purge them all. Make sure only for the version in question. And keep your OLD kernels / ALTERNATE kernels around because you'll have to boot to them to re-install. And/or just to do this step.) $ sudo rm -rf /lib/modules/linux-*version* (tab completion is your friend) $ sudo rm -rf /boot/*version* (make sure you're not grabbing anything important. As long as your alternates don't share the same version number, you should be safe-ish) Perhaps a good ideal to do a full backup before these steps, just in case. $ sudo apt-get install kernel-*version* (plus any related packages) It's not unheard of for a distro to botch a particular kernel for a particular purpose. Depending on your distro and version there of. Most times they will be updated or replaced with the latest and greatest at the next update. At least a couple times a year, so 30 days to six months and your issue might automagically disappear. Otherwise try those steps above. Perhaps an ls *version* beforehand to ensure that you're not grabbing anything not intended to be grabbed. You can also mv the stuff versus rm if you want a recovery option, but a bit more tedious and no real need to hang on to it for all intents with alternate options. Potentially dangerous commands there so be weary of fat fingers. And backup first if you don't trust yourself. And backup if you DO trust yourself. Not to clutter the issue, but sometimes /boot/ is on a different partition / device and unmounted after boot. In that instance you might need to do some trickery to have it be there to uninstall from. Basically don't assume anything, verify verify verify. With certain permission schemes(acl/selinux) it's entirely possible that the process is not that simple. But it could be. - James On 2/15/11, Marcin Szyniszewski mszyn...@gmail.com wrote: Hello, I checked if alsa and stuff is working on other kernels - it seems it is working brilliantly! Mic and sound works fine! So the problem would be with the latest kernel. I removed it while being on previous one and installed it again, but the problem is still present. Then I tried to do all the stuff that was suggested here again, nothing worked. So it looks like it's the fault of kernel but reinstalling it somehow doesn't work! Do you have some suggestions? Maybe it's not kernel after all? Or maybe there's some different way to remove the kernel completely and reinstall it? My impression that that at some earlier stage audio *was* working, so the current lack of working is due to something like an attempt to do something like 'upgrade' the kernel. If so, my recommendation is to *always* do a backup of your system before doing anything that might furtle things up. I use 'clonezilla' for this every now and then to try to protect myself from my own idiocy. Put the backup on a removable USB HD. But there are various other ways you may prefer. Yeah, thats a good idea. Fortunately there are previous kernels available! Best, *mszynisz* -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound, no /proc/asound/
As I suspected, the modules aren't loaded so alsa isn't even running. Hence your original open error(s). How did you install alsa? Not that I think it is your issue, but it could be. If you boot with lilo, you need to re-install lilo after creating a new kernel. Even if it's technically the same version of your old kernel. Although most distros default to grub these days. So not likely. If you compiled from source at least for some modules, you'll need to reboot to use the new kernel and the new modules. Not really applicable to sound as you probably didn't change any PCIe or other internals to gain the functionality. In the old days if you compiled from source you could insmod (modprobe) the modules in alsa-driver-???/modules/ until you got the right order and all of the modules loaded. This is representative of the errors that you're seeing. You can't load a certain module because another module wasn't loaded before it. That has those symbols (functions) that it needs. Which brings things full circle to alsa isn't properly installed. $ sudo dpkg -l '*alsa*' Only pay attention to the ones that start alsa or alsa-. On my debian setup (similar to ubuntu) I have alsa, alsa-base, alsa-firmware-loaders, alsa-headers, alsa-source, alsa-tools, alsa-tools-gui, and alsa-utils. On my system all of those are installed, except alsa-firmware-loaders, alsa-headers (needed to compile other things from source against it), and alsa-tools-gui. IMO you are probably missing alsa-base. This should have entries in /etc/modprobe.d/alsa* for autoloading your modules (without concerning yourself about the order of insertion). It could also be that you haven't run depmod -a, or your distro didn't. Which updates a sort of list of what modules are related so they can also load when the other is loaded. IME, alsa is independent of this list and relies on other things (/etc/modprobe.d/). If you haven't solved your issue by now, I guess you're stuck with the old school ways. Meaning you'll likely have to create a /etc/modprobe.d/ entry for alsa so it can auto load at boot. Which might look something like: #--- START - /etc/modprobe.d/alsa_custom.conf ---# alias char-major-116 snd alias char-major-14 soundcore options snd major=116 cards_limit=3 # duplicate this following sequence for each soundcard you have # and bump (or omit) the index=# depending on the order / priority # that you desire. And adjust the first # in the sound- aliases to # match the index number. # your specific module NEXT LINE (and the next one) options snd-hda-intel index=0 alias snd-card-0 snd-hda-intel # this one assumes OSS emulation, you might need to # reference alsa-project.org to find a different one if you # opted out on that option. --with-oss=yes ? # (been a while) alias sound-slot-0snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss #--- END ---# And 20 years after linux started, we're still configuring sound from the command line. Be sure to reboot OR try to use the soundcard to get the modules to auto magically load. They generally load at boot because your distro will likely try to restore mixer settings. And therefor try to use your soundcard. (which is or was failing for you) - James On 2/12/11, Marcin Szyniszewski mszyn...@gmail.com wrote: Thank you all for the replies! Very appreciated! :) $ sudo modprobe [module] FATAL: Error inserting snd (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error running install command for snd WARNING: Error inserting snd_pcm (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hda_codec (/lib/modules/2.6.35-25-generic/kernel/sound/pci/hda/snd-hda-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_hda_intel (/lib/modules/2.6.35-25-generic/kernel/sound/pci/hda/snd-hda-intel.ko): Unknown symbol in module, or unknown parameter (see dmesg) This doesn't look good. What do you think is wrong?? Note that * is a wildcard. So /dev/dsp* is any devices that start with /dev/dsp. Yes, of course. I did ll in this folder and went through the whole list. Nothing's there. # modprobe snd-hda-intel Gives me permission errors. $ sudo modprobe snd-hda-intel Gives the result above. $ sudo pavucontrol sudo: pavucontrol: command not found $ lsmod | grep -i snd snd_page_alloc 7120 0 $ cat /proc/asound/cards cat: /proc/asound/cards: No such file
Re: [Alsa-user] No sound, no /proc/asound/
$ sudo dpkg -l '*alsa*' Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend |/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad) ||/ Name Version Description +++--- un alsa none (no description available) ii alsa-base1.0.23+dfsg-1ubuntu4 ALSA driver configuration files ii alsa-firmware-loaders1.0.23-3ubuntu1 ALSA software loaders for specific hardware ii alsa-oss 1.0.17-4 ALSA wrapper for OSS applications ii alsa-source 1.0.23+dfsg-1ubuntu4 ALSA driver sources ii alsa-tools 1.0.23-3ubuntu1 Console based ALSA utilities for specific hardware ii alsa-tools-gui 1.0.23-3ubuntu1 GUI based ALSA utilities for specific hardware ii alsa-utils 1.0.23-2ubuntu3.4 Utilities for configuring and using ALSA ii alsamixergui 0.9.0rc2-1-9 graphical soundcard mixer for ALSA soundcard driver ii bluez-alsa 4.69-0ubuntu2 Bluetooth audio support ii gnome-alsamixer 0.9.7~cvs.20060916.ds.1-2 ALSA sound mixer for GNOME ii gstreamer0.10-alsa 0.10.30-2 GStreamer plugin for ALSA un libsdl1.2debian-alsa none (no description available) Looks like there's some problem with alsa :( How to fix this? Well there's the old school ways. When all else fails, re-install. Fortunately in linux that's not as dreaded as it sounds $ sudo dpkg --purge --force-all alsa alsa-base alsa-firmware-loaders alsa-oss alsa-source alsa-tools alsa-tools-gui alsa-utils alsamixergui (removes the packages) $ sudo apt-get install alsa alsa-base alsa-firmware-loaders alsa-oss alsa-source alsa-tools alsa-tools-gui alsa-utils alsamixergui (puts them back) HTH, - James -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound, no /proc/asound/
Basically the same way. To redo your kernel. dpkg --purge --force-all apt-get install. Just make sure that you DO NOT do it to the kernel that you are currently running. Which might mean installing an older kernel as a safe recovery and boot to that before recovering the kernel you want to run. This week anyway. Definitely a module mismatch. But is it because the kernel you are running is mangled with old modules, old initrd images, old ??? Because the boot loader isn't using the NEW kernel version? Depmod and any number of things depending on how you came about your current config. Assuming the old school insmod route can't be made to work at all. Which requires some functional knowledge of your system. i.e. lsmod (from a working version / live CD). But requires no configuration to load the modules, outside of the right sequence. And full paths if you use insmod, and not modprobe. $ sudio find /lib/modules/`uname -r`/ -iname '*snd*.*o' I've been assuming that you've been running a distro supplied kernel. I guess the question should be asked, how did you come by your current kernel? Supplied by the distro or did you do something different? In either case you might want to try a distro supplied kernel. Preferably one that differs from the version (name) that you are currently using. At least in terms of simple fixes. Beyond that you might rm ~/.asoundrc and the /etc/modprobe.d/alsa_custom.conf when you reinstall alsa. Or at least mv to ~/ with different names so you can easily recover them. Otherwise it appears that you might have installed alsa from source, and an update to the same kernel version might have overwritten in part your changes. The rm step to happen between dpkg --purge and apt-get install. For the kernel you might want to rm the /lib/modules/2.6.35.???/ for the kernel in question, just in case something lingered. Between purge and install of course. While running a differently named kernel. Otherwise a fresh FULL reinstall should fix your issue. Assuming that your card is supported in the first place, which it appears to be or it would have never worked. Otherwise we could troubleshoot for days without more information about how you got to your current state of affairs. Not that you'd have that standard M$ answer. I installed AOL and now XXX doesn't work anymore... - James On 2/12/11, Bill Unruh un...@physics.ubc.ca wrote: On Sat, 12 Feb 2011, Marcin Szyniszewski wrote: On Sat, Feb 12, 2011 at 16:26, James Shatto wwwshad...@gmail.com wrote: $ sudo depmod -a $ sudo modprobe snd-hda-intel WARNING: Error inserting snd_timer (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) This usually means that you have a module mismatch-- the modules you loaded are not the up to date modules for your kernel. You may well have neglected to uninstall previous modules before puttin in the new ones. I would remove the current kernel and then reinstall the kernel forcing it to reinstall everything (I have no idea how debian does this-- I use a rpm based system). None of the alsa modules are being installed so it is not surprizing you are getting no sound. WARNING: Error inserting snd_pcm (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hda_codec (/lib/modules/2.6.35-25-generic/kernel/sound/pci/hda/snd-hda-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_hda_intel (/lib/modules/2.6.35-25-generic/kernel/sound/pci/hda/snd-hda-intel.ko): Unknown symbol in module, or unknown parameter (see dmesg) Looks like module loader is not willing to cooperate :/ Do you know what's going on? Thank you all for the replies! Please help! Best, *mszynisz* -- William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273 PhysicsAstronomy | Advanced Research | Fax: +1(604)822-5324 UBC, Vancouver,BC | Program in Cosmology | un...@physics.ubc.ca Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/ -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound, no /proc/asound/
Most times when I get something like that it has to do with the /dev/'s not being present. Could be that udev isn't running on your box. Or isn't configured for alsa. It could also be something else like snd-pcm-oss not auto loading. And it's friends, snd-mixer-oss snd-seq-oss. Basically cannot open means some sort of missing something or bad permissions. Is the user in the audio group? Do the /dev/audio* and /dev/dsp* stuff exist? In the old days we'd run ./snddevices from the alsa-driver source tree. But that's probably not the solution of choice these days. # /etc/init.d/alsa-utils restart # /etc/init.d/udev restart # groups user # grep -i audio /etc/group lsmod, dmesg, and all of the other stuff that's probably covered by that alsa-info.sh script thing. - James On 2/11/11, Jim Lesurf j...@audiomisc.co.uk wrote: In article AANLkTikA=hHDEy3pCsamVvgye7u9=_4pqqw_pscjb...@mail.gmail.com, Marcin Szyniszewski mszyn...@gmail.com wrote: Is the file /usr/bin/alsamixer present, or /sbin/alsa ? Or the /usr/share/alsa directory? You should have these or equivalents IIUC. /usr/bin/alsamixer is present and gives: cannot open mixer: No such file or directory Did you issue alsamixer as the command or the full pathname? If the former, maybe something is wrong with your path/environment setup. Afraid I don't know what the problem is, so I can only suggest some ideas and diagnostics to check. I am wondering if your OS install hasn't actually loaded the modules correctly for your hardware. Try the command 'lsmod' to list the modules that are loaded. If the list is too long use 'lsmod | grep snd' to just list the ones that have 'snd' in their names. You can then use modinfo module name to check details of each module. Or modprobe (with care!) to alter what is loaded. Do you have another sound system like Pulse active? if so, that may be interfering with the direct use of ALSA. You could also put a simple redefinition of the ALSA default into an .asoundrc file and see if that can be made to work with aplay. But from what you have said I have doubts about that. You might also consider trying to install the latest version of ALSA in case what you have isn't suitable for your hardware or is furtled in some way. Sorry I can't be more help. But I hope the above may be useful. Slainte, Jim -- Electronics http://www.st-and.ac.uk/~www_pa/Scots_Guide/intro/electron.htm Audio Misc http://www.audiomisc.co.uk/index.html Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound, no /proc/asound/
Note that * is a wildcard. So /dev/dsp* is any devices that start with /dev/dsp. It looks like you don't have the modules loaded. # modprobe snd-hda-intel $ sudo modprobe snd-hda-intel (depending on your distro / $ is user / # is root) It might be /etc/init.d/alsasound or other named thing depending on your version and distro. It might not even be in /etc/init.d/ depending on your distro. It looks like you have pulse audio running, so you might try the pavucontrol application. Should be accessible through the speaker icon in the taskbar in ubuntu. Or just run it from a terminal. $ sudo pavucontrol You appear to be installed and with permissions, but if you don't have /dev/dsp and friends, then you don't have alsa running. Probably didn't load up the modules at boot. Not completely uncommon on a new install. Someplace to start looking anyway. $ lsmod | grep -i snd $ cat /proc/asound/cards - James On 2/11/11, Torsten Schenk torsten.sch...@zoho.com wrote: I also use ubuntu (10.04) and it came to happen that the system didn't load the modules automatically any more. I don't know why that happened or where this loading is prohibited. Just try to load the module manually and see if that works. If so, you could also post this on a ubuntu mailing list. $ sudo modprobe [module] You need to replace [module] with the module that fits your card, eventually snd-hda-intel or snd-usb-audio, these are very common cards. Greets, Torsten On Fri, 11 Feb 2011 20:54:49 +0100 Marcin Szyniszewski wrote Did you issue alsamixer as the command or the full pathname? If the former, maybe something is wrong with your path/environment setup. I used it as both. Nothing works :/ I am wondering if your OS install hasn't actually loaded the modules correctly for your hardware. Everything worked before. I tried to make my mic work and sound stopped to work. Now nothing works :P Try the command 'lsmod' to list the modules that are loaded. If the list is too long use 'lsmod | grep snd' to just list the ones that have 'snd' in their names. $ lsmod | grep snd snd_page_alloc 7120 0 But I don't know what that means :P You can then use modinfo to check details of each module. Or modprobe (with care!) to alter what is loaded. Ok, and what modules should I check? Do you have another sound system like Pulse active? if so, that may be interfering with the direct use of ALSA. Stopping pulse and reinstalling ALSA didn't work. :( You might also consider trying to install the latest version of ALSA in case what you have isn't suitable for your hardware or is furtled in some way. I think I have the latest version. Sorry I can't be more help. But I hope the above may be useful. Thanks for help :) Most times when I get something like that it has to do with the /dev/'s not being present. Could be that udev isn't running on your box. Or isn't configured for alsa. It could also be something else like snd-pcm-oss not auto loading. And it's friends, snd-mixer-oss snd-seq-oss. Basically cannot open means some sort of missing something or bad permissions. Is the user in the audio group? Do the /dev/audio* and /dev/dsp* stuff exist? In the old days we'd run ./snddevices from the alsa-driver source tree. But that's probably not the solution of choice these days. /dev/audio* doesn't exist, as well as /dev/dsp* Should I do something about that?? # /etc/init.d/alsa-utils restart bash: /etc/init.d/alsa-utils: No such file or directory # /etc/init.d/udev restart Rather than invoking init scripts through /etc/init.d, use the service(8) utility, e.g. service udev restart Since the script you are attempting to invoke has been converted to an Upstart job, you may also use the restart(8) utility, e.g. restart udev restart: Rejected send message, 1 matched rules; type=method_call, sender=:1.45 (uid=1000 pid=9806 comm=restart) interface=com.ubuntu.Upstart0_6.Job member=Restart error name=(unset) requested_reply=0 destination=com.ubuntu.Upstart (uid=0 pid=1 comm=/sbin/init)) # groups mszynisz : mszynisz adm dialout fax cdrom floppy tape audio dip video plugdev fuse netdev lpadmin admin sambashare # grep -i audio /etc/group audio:x:29:pulse,mszynisz lsmod, dmesg, and all of the other stuff that's probably covered by that alsa-info.sh script thing. My output of alsa-info.sh script is attached. Please help, I really need my sound :( Best, mszynisz -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] No sound, no /proc/asound/
Except that your web browser likely defaults the OSS, not ALSA. And OSS emulation IS part of alsa. Even if you have to launch an application with aoss to use the alsa sound drivers. It's probably not native alsa, but it is coded as part of alsa's drivers, and therefor part of alsa. But yeah, check /dev/snd* for items as well. It does vary depending on version of alsa, version of the kernel, and other things. $ ls -l /dev/* /dev/*/* | grep -i audio In either case your audio group will likely be assigned to the audio devices available to you. By all means nitpick that I used grep -i audio, versus awk '{ print $4 $9 }' | grep -i audio or something. - James On 2/11/11, Bill Unruh un...@physics.ubc.ca wrote: On Fri, 11 Feb 2011, James Shatto wrote: Note that * is a wildcard. So /dev/dsp* is any devices that start with /dev/dsp. It looks like you don't have the modules loaded. /dev/dsp and /dev/audio are the oss sound drivers, not alsa. alsa has an oss emulation module, which will create those but they are NOT needed for using alsa. What you have under alsa is a buch of entry points under /dev/snd Now if your program uses the oss sound system, then you must load the alsa-oss emulators as well (snd_seq_oss, snd_pcm_oss snd_mixer_oss) This will generate the various /dev/dsp entry points. # modprobe snd-hda-intel $ sudo modprobe snd-hda-intel (depending on your distro / $ is user / # is root) It might be /etc/init.d/alsasound or other named thing depending on your version and distro. It might not even be in /etc/init.d/ depending on your distro. It looks like you have pulse audio running, so you might try the pavucontrol application. Should be accessible through the speaker icon in the taskbar in ubuntu. Or just run it from a terminal. $ sudo pavucontrol You appear to be installed and with permissions, but if you don't have /dev/dsp and friends, then you don't have alsa running. Probably Totally false. /dev/dsp is NOT part of alsa. didn't load up the modules at boot. Not completely uncommon on a new install. Someplace to start looking anyway. $ lsmod | grep -i snd That is a good starting point. $ cat /proc/asound/cards - James On 2/11/11, Torsten Schenk torsten.sch...@zoho.com wrote: I also use ubuntu (10.04) and it came to happen that the system didn't load the modules automatically any more. I don't know why that happened or where this loading is prohibited. Just try to load the module manually and see if that works. If so, you could also post this on a ubuntu mailing list. $ sudo modprobe [module] You need to replace [module] with the module that fits your card, eventually snd-hda-intel or snd-usb-audio, these are very common cards. Greets, Torsten On Fri, 11 Feb 2011 20:54:49 +0100 Marcin Szyniszewski wrote Did you issue alsamixer as the command or the full pathname? If the former, maybe something is wrong with your path/environment setup. I used it as both. Nothing works :/ I am wondering if your OS install hasn't actually loaded the modules correctly for your hardware. Everything worked before. I tried to make my mic work and sound stopped to work. Now nothing works :P Try the command 'lsmod' to list the modules that are loaded. If the list is too long use 'lsmod | grep snd' to just list the ones that have 'snd' in their names. $ lsmod | grep snd snd_page_alloc 7120 0 But I don't know what that means :P You can then use modinfo to check details of each module. Or modprobe (with care!) to alter what is loaded. Ok, and what modules should I check? Do you have another sound system like Pulse active? if so, that may be interfering with the direct use of ALSA. Stopping pulse and reinstalling ALSA didn't work. :( You might also consider trying to install the latest version of ALSA in case what you have isn't suitable for your hardware or is furtled in some way. I think I have the latest version. Sorry I can't be more help. But I hope the above may be useful. Thanks for help :) Most times when I get something like that it has to do with the /dev/'s not being present. Could be that udev isn't running on your box. Or isn't configured for alsa. It could also be something else like snd-pcm-oss not auto loading. And it's friends, snd-mixer-oss snd-seq-oss. Basically cannot open means some sort of missing something or bad permissions. Is the user in the audio group? Do the /dev/audio* and /dev/dsp* stuff exist? In the old days we'd run ./snddevices from the alsa-driver source tree. But that's probably not the solution of choice these days. /dev/audio* doesn't exist, as well as /dev/dsp* Should I do something about that?? # /etc/init.d/alsa-utils restart bash: /etc/init.d/alsa-utils: No such file or directory # /etc/init.d/udev restart Rather than invoking init scripts through /etc/init.d, use the service(8) utility, e.g. service udev restart Since the script you
Re: [Alsa-user] ftp.alsa-project.org down?
It doesn't appear to be NAT. At least not anything that I have control over. Same error(s) on the router box with or without firewall. FTP to my other ISP's base web space works fine. $ curl ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2 curl: (56) FTP response reading failed $ wget ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2 Error in server response, closing control connection. $ ftp open ftp.alsa-project.org 421 Service not available, remote server has closed connection and so on and so on. traceroute to alsa0.alsa-project.org (212.20.107.51), 30 hops max, 40 byte packets 1 192.168.2.1 (192.168.2.1) 1.566 ms 1.975 ms 2.980 ms ... 5 user45.embarqnow.net (64.45.249.45) 21.211 ms 22.044 ms 22.377 ms 6 ge-6-14.car2.Houston1.Level3.net (4.78.10.17) 30.828 ms 18.662 ms 17.253 ms 7 ae-2-5.bar2.Houston1.Level3.net (4.69.132.238) 19.686 ms 24.716 ms 25.090 ms 8 ae-7-7.ebr1.Atlanta2.Level3.net (4.69.137.142) 49.183 ms 49.600 ms 48.246 ms 9 ae-63-60.ebr3.Atlanta2.Level3.net (4.69.138.4) 47.535 ms 47.883 ms 47.829 ms 10 ae-2-2.ebr1.Washington1.Level3.net (4.69.132.86) 57.109 ms 55.129 ms 57.400 ms 11 ae-61-61.csw1.Washington1.Level3.net (4.69.134.130) 54.671 ms ae-91-91.csw4.Washington1.Level3.net (4.69.134.142) 45.316 ms 44.817 ms 12 ae-82-82.ebr2.Washington1.Level3.net (4.69.134.153) 50.233 ms 49.554 ms ae-72-72.ebr2.Washington1.Level3.net (4.69.134.149) 50.113 ms 13 ae-44-44.ebr2.Frankfurt1.Level3.net (4.69.137.61) 139.208 ms ae-43-43.ebr2.Frankfurt1.Level3.net (4.69.137.57) 134.281 ms ae-44-44.ebr2.Frankfurt1.Level3.net (4.69.137.61) 166.472 ms 14 ae-5-5.car2.Prague1.Level3.net (4.69.135.50) 179.133 ms 179.015 ms 179.089 ms 15 ae-11-11.car1.Prague1.Level3.net (4.69.135.41) 175.002 ms 175.047 ms 175.025 ms 16 212.162.8.14 (212.162.8.14) 156.870 ms 181.953 ms 182.332 ms 17 perexsoft.customer.vol.cz (212.20.107.218) 164.298 ms 184.698 ms 159.328 ms 18 * * * 19 * * * 20 * * * 21 * * * 22 * * * 23 * * * 24 * * * 25 * * * 26 * * * 27 * * * 28 * * * 29 * * * 30 * * * - James On 5/30/10, Jaroslav Kysela pe...@perex.cz wrote: 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 - Jaroslav Kysela pe...@perex.cz Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] ftp.alsa-project.org down?
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. I'm interested in doing this because jackd requires 1.0.18 or better version(s) of alsa for alsa support (oss might actually work) if compiled from sources. And my current version of mplayer wont compile with jack support against my current version of jackd. aka dependency hell in source mode. Thanks, - James -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Microphone + AudioOut to HDMI
On Tue, 5 Jan 2010 01:07:21 +0300 An St vit@gmail.com wrote: Hello! Please help. I can't get working microphone at HDMI output. HDMI audio normally has some sort of limit in place. For my ATI HD4550 video card, the audio has to be transmitted in an AC3 codec(5.1 surround). AKA compressed, it will not work with PCM audio which might be where your making a connection is giving you trouble. That capability might depend on your graphics card, but that's the quirk of mine. Assuming that your HDMI audio is provided via a graphics card. Maybe alsa can handle the AC3 conversion transparently / internally, or NOT. It's seems a bit destined for problems IMO, so I just avoid the issue with an RCA cable from a dedicated soundcard. Fortunately my HDTV has a channel with HDMI/DVI input and RCA audio input so running that machine on a 42 display is possible. If my graphics card does handle PCM audio over HDMI, it's probably limited to 2 channels, 48kHz, 16 bit, and all that jazz. Maybe even 44.1kHz. I really haven't checked the specs that recently on it. But I only have one receiving device for HDMI audio (HDTV) so it's not a priority to explore for me. Which is kind of ironic since the audio device registers and an hda-intel device. AKA high definition audio. But the limits are listed in the manual. Not that I've looked at it in the past year+. HTH, James -- This SF.Net email is sponsored by the Verizon Developer Community Take advantage of Verizon's best-in-class app development support A streamlined, 14 day to market process makes app distribution fast and easy Join now and get one step closer to millions of Verizon customers http://p.sf.net/sfu/verizon-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Redirecting the output audio to the microphone input
On Wed, 9 Dec 2009 09:23:44 -0200 Kazuo Teramoto kaz@gmail.com wrote: On Wed, Dec 9, 2009 at 5:16 AM, James Shatto shado...@earthlink.net wrote: You can set the record device to PCM (aumix term, never been able to find the equivalent alsamixer way). Although you'll likely need to adjust your volume levels to get a good level (which might be below audible levels). I cant find the setting to set the record device to PCM can you give me a amixer command line for it? Like I said aumix seems to be the only command line one I've found with that option/feature. Not that I've tried to do it other ways. $ aumix -q $ aumix -v R $ aumix -q The options for -p and -w seem to set it to mic for microphone input. In the gui mode it's the square with the green center square, click the one for master volume (first one) to set it to the pcm output as your default recording channel. It should turn/be red in the gui, when it is. There might be an amixer option for that. Alsamixer seems to only let me toggle mic capture ON, but NOT off, which undoes the change aumix made. I mainly use this feature to make a WAV of festival output for text to speech. Seems like aumix says Capture to on and Mic to off (relative to capture), when aumix changes things. With alsamixer I seem to only be able to set Mic to ON. Bear in mind that this is completely dependent on your hardware supporting this feature. Roughly %50 volume on my laptop. And it requires a soundcard that supports that. Otherwise use a cable to connection line out to line in. Which could be on that machine or another one. This is not a solution in my case, I don't have a line in jack connection. A somewhat related idea is how I can read audio from a file and pipe it to microphone input. If you have a file, you don't really need to, outside of some sort of realtime performance setup with effects. But if you have a file, just how real time is it? Sox, ffmpeg, audacity, and a few other applications can convert file formats to other formats. Or play them back, you don't really need to record them, if they're already in file format. And if your soundcard isn't full duplex, you might already be getting some bleed through accidentally. I need what I asked for, because I like to emulate a microphone. I like to play sounds in a program that only accept mic input, but cant take files an input e.g. Skype and other voice programs, with games (Counter Strike Source), etc. I not doing this to convert files (if Í needed to convert I had searched for a converting solution, I'm not that stupid =] ) Then don't call them files. Use sources or other more appropriate terms. -- «Dans la vie, rien n'est à craindre, tout est à comprendre» Marie Sklodowska Curie. For the resulting changes after $ aumix -v R $ amixer get Mic Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 27 [87%] [6.00dB] [off] Front Left: Capture [off] Front Right: Capture [off] $ amixer get Capture Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 8 [53%] [12.00dB] [on] Front Right: Capture 8 [53%] [12.00dB] [on] I haven't been able to get this result by anything other than aumix. And I'm not familiar with the amixer equivalent. But here's what it's like BEFORE I change the capture device AWAY from Mic with aumix. $ amixer get Mic Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 27 [87%] [6.00dB] [off] Front Left: Capture [on] Front Right: Capture [on] $ amixer get Capture Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 8 [53%] [12.00dB] [on] Front Right: Capture 8 [53%] [12.00dB] [on] Your hardware may vary. My hardware is an ATI IXP SB400 on my compaq presario laptop. 1002:4370 Other options might be to use jackd and qjackctl to make associations, or some form of pulse audio. There's many means to an end. Bear in mind that piping line out to mic in, WILL result in feedback if there's any sort of playthrough / relation between the two channels. - James -- 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
Re: [Alsa-user] Redirecting the output audio to the microphone input
On Tue, 8 Dec 2009 21:26:14 -0200 Kazuo Teramoto kaz@gmail.com wrote: Hello. I like to redirect the sound I hear in the speakers to microphone, so it can be recorded with e.g. arecord. You can set the record device to PCM (aumix term, never been able to find the equivalent alsamixer way). Although you'll likely need to adjust your volume levels to get a good level (which might be below audible levels). Roughly %50 volume on my laptop. And it requires a soundcard that supports that. Otherwise use a cable to connection line out to line in. Which could be on that machine or another one. A somewhat related idea is how I can read audio from a file and pipe it to microphone input. If you have a file, you don't really need to, outside of some sort of realtime performance setup with effects. But if you have a file, just how real time is it? Sox, ffmpeg, audacity, and a few other applications can convert file formats to other formats. Or play them back, you don't really need to record them, if they're already in file format. And if your soundcard isn't full duplex, you might already be getting some bleed through accidentally. I think alsa can do this with some sort of combination of plugin file, dsnoop and some asoundrc-fu but I cant get all the concepts to create a solution by myself. Someone can help me please? Thanks, Kazuo Teramoto -- 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] USB soundcard advice
On Mon, 2 Nov 2009 20:42:48 +0100 Y.A. Bolawy bol...@gmail.com wrote: Hi all, I'd like some advice on a USB soundcard. The reason for getting one is that I'd like to have good quality sound on all the computers I use or will use. The quality should be good enough to allow speech recognition. Of course, that is possible to some extend with any soundcard, but if the quality is low it has a big impact. Unfortunately, the better the quality of the cards, the less standard compliance they seem to be. At least that seems to be the underlying message of everything I've read so far. USB Audio is a standard. As long as the box says class compliant, it should work out of the box in linux. Only one caveat though as it wont default to card 0 and be your default card. Since you probably have a motherboard with onboard sound. Configure accordingly. I have a USB M-Audio Mobile Pre and it works fine. Although web browsers don't seem to use it properly even though I have it configured to card 0. I've never had a problem recording from it though. Not really the best audio option, but loads better than most stock soundcards. HTH, James -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Multiple cards, alphanumeric names?
On Tue, 03 Nov 2009 18:25:24 -0600 Jonathan E. Brickman j...@joshuacorps.org wrote: OK. I now find myself happily educated in card names (HD2 in my case), devices as being items on cards (HD2,0 et cetera), and subdevices whose names appear to be used in rather different locations. My next question: What if I had two cards of this type? Do I have to use numeric names, or is there an alphanumeric rule built in somewhere which gives me HD2(0) or some such? J.E.B. I think that you're getting grub device names confused with alsa names. Normally you can address them by hw:0 or hw:1 or hw:2. Basically hw:0,1 for capture device and hw:0,0 for playback. You can give more meaningful names in your .asoundrc configuration. But generally NOT HD#, that's a grub thing for Hard Drive. Although most alsa apps reference them by -c # where the # matches their designation in /proc/asound/cards. Many apps that use alsa use something like -D hw:2 or -ao alsa:device=hw:2 and that is assuming that you don't want to just use the defaults. HTH, - James -- 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] audio loopback in linux
On Fri, 28 Aug 2009 10:38:16 +0200 (CEST) Julien Claassen jul...@c-lab.de wrote: Hi! I'm not sure, if alsa does it, still. But you can do it with jackd (Jack Audio Connection Kit). It's a low latecny audio server and a lot of Linux Audio software support it. You can find packages in your distro. Then you simply do: jack_connect system:capture_1 system:playback_1 jack_connect system:capture_2 system:playback_2 Or install some GUI connection tool, e.g. qjackctl. Hope that helps Julien jack + qjackctl does this (or the CLI alternative). If your hardware supports it, it'll be listed in alsamixer. My delta 44 is anyway. On that card it shows as H/W H/W 1 H/W 2 H/W 3 and you just change it from PCM Out to HW In 0 or whatever source you want. Bear in mind that if input is a mic and the speaker is loud enough you'll get feedback. And if you're doing it for some sort of TV Capture card that audio has less latency than video, so you'll hear them talk before their lips move (just slightly) which can/will drive you nuts. HTH, James -- 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] M-Audio Delta 66 Not working and cannot load modules after compile
(/lib/modules/2.6.24-19-generic/ubuntu/sound/alsa-driver/acore/seq/snd-seq-device.ko): there's your problem Alsa from source will likely install to: /lib/modules/`uname -r`/kernel/sound/ Which means you likely have two versions: /lib/modules/`uname -r`/ubuntu/sound/ find /lib/modules/`uname -r`/ -iname '*snd-*.*o' depmod -ae will pick up both version. You probably need to remove one version to get it to work. Then rerun depmod. I'm not sure if soundcore.ko is alsa, or the kernel. In days of old it was part of OSS, or so I thought. It might just be easier to do a custom kernel. With alsa compiled over it. That way there's not multiple versions / locations. As the path assumptions would match. HTH, James - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] M-Audio Delta 66 Not working and cannot load modules after compile
On Mon, 29 Sep 2008 23:22:27 -0500 John Beavers [EMAIL PROTECTED] wrote: Hello all, I have a few problems. My main problem is that I have installed an M-audio Delta 66, and it will output no sound, and does not recognize input from sound sources, either. But before we get to that, I have a more pressing issue. I tried compiling the latest build of Alsa, and when I get to the step of inserting the driver, it gives me all sorts of errors: sudo modprobe snd-ice1712 FATAL: Error inserting snd (/lib/modules/2.6.24-19-rt/ubuntu/sound/alsa-driver/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error running install command for snd WARNING: Error inserting snd_seq_device (/lib/modules/2.6.24-19-rt/ubuntu/sound/alsa-driver/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) I have a Delta 44 and it works fine. The inputs are picky in that they need to be fed with a line level source. Which for me means using a microphone preamp. Even on some sources that may not need one in other circumstances / other cards. That being said, the unknown symbol is common. If insmod-ing without deps, you can only do this in a specific order. If you boot using lilo and didn't rerun lilo to install the newer kernel, then it may be having version conflicts. If you didn't run depmod -ae after installing the newer alsa you might also have trouble. snd-ice1712 should be the right module(s). For me I modify the /etc/modules.conf configuration or modprobe.d / modutil.d modern equivalents. While disabling distro supplied defaults. In my case it looks something like this. # /etc/modprobe.d/alsa_custom alias char-major-116 snd alias char-major-14 soundcore options snd major=116 cards_limit=1 options snd-ice1712 index=0 alias snd-card-0 snd-ice1712 alias sound-slot-0snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss # END With this configuration, anytime you try to use the sound device the modules are automatically loaded. Also when you modprobe snd-ice1712, it picks up the dependants in whatever order they were supposed to be used in. Reindexing it to 0 makes it the default soundcard. There's other ways to do that, but this simplifies things for OSS type apps. Like festival / mozilla / . There are other issues if you're not using udev and/or didn't run the snddevices script to create the /dev devices (not to be run if you ARE using udev). But that doesn't appear to be your issue. And other ways to implement the above custom configuration with alsaconf and other utilities. I just never got them to work for me back in the day, and never adapted to letting current tools try to do it for me. - James - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA
I need to make multitrack recordings; I' m looking for a sound card usb2 model of at least 4/6/8 balanced inputs, XLR with phantom power to 48V and audio resolution 24-bit/96kHz and with many analog audio outputs maybe XLR balanced, SPDIF in / out and MIDI in / out / trough. For those specs you need a firewire device. USB has limited bandwidth. The best I've come across that work, are 2x 16 bit 48kHz input with simultaneous 2x 16 bit 48kHz output. The best I've seen is 2x 24 bit 96kHz, and the reviews on them are not great. Buggy, not full duplex at that rate, and other driver-ish issues. Even in windows. The USB bus is very limited and at a very minimum has latency issues if you want to multitrack. Even at 16 bit 48 kHz. If PCI is a possibility, then an Echo Layla 3G might be to your liking. But I don't know about it's linux support status. I just don't know if you're gonna find a device like that, that works in either windows or linux. Unless it's a firewire device like a Presonus Firepod / Firebox / FP10 / Whatever the marketing name of the year is. Or a PCI device. Go PCI or Firewire, you'll have many more options, and not as many headaches. HTH - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA
Hello everyone! I now resigned to not being able to use the m-audio fast track ULTRA usb soundcard with my LINUX-DAWs, someone can recommend another card usb I can afford multitrack audio recordings of quality, which is working with Linux? As said before, my M-Audio Mobile Pre seems class compliant. And otherwise works. My Delta 44 (pci) works too. The Delta worked for several months before they finally came up with Windows Vista drivers. But it doesn't sound like you want another M-Audio. So probably the Lexicon Omega / Alpha type cards might work for you. What type of inputs are you needing? And how many inputs? TRS / XLR / 3.5mm stereo? There's several out there depending on your needs. What's your budget? If you need a lot more inputs at higher rates, then firewire might be better suited. freebob.sf.net HTH, - James - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] OT: multimedia repositories for Debian...
I ended up recompiling an old version of xmms from source under Debian to get the real-time support I wanted, then compiled plugins from source as I needed them, using the checkinstall utility to create .deb files. Since this is the alsa-user list, does anyone know of current media players that use ALSA that support real-time priority? I want to play files of all audio formats, MIDI and streaming media without skipping on older hardware (-:. Arthur. For jackd it wasn't so much of the media players that needed realtime priority. Modifying /etc/security/limits.conf gave me realtime priority at the user level. Just three new entries in that file: @audio - rtprio 99 @audio - nice-10 @audio - memlock 65536 And no need for other repositories since I compiled from source. It was just one package of 2,000+ installed packages. Although there are others like audacity compiled with --port-audio=v19. Timidity with sequencer support and jackd. And various other annoyances. I've pretty much run on highly stripped, custom kernels since I was able to successfully create one. So recompiling from source is a no brainer. And a lot less annoying on todays faster cpus. HTH, - James - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] how to convert stereo- mono signal during record
On Thu, 28 Aug 2008 17:03:44 +0200 brunal [EMAIL PROTECTED] wrote: I have to precise that I'm using a M-Audio fatst track pro sound card, which works fine if I only record stereo files. Is there some reason the application doesn't do mono? Almost all I've seen allow you to record only one track. Even on my MCP61, or my Mobile Pre. Which are stereo devices. You could look at audacity to see how it does it. Although it might be difficult to find being a multiplatform application. Ardour lets you do it, but it's using jack. At a minimum you should be able to split the track out after the fact. $ sox -c 2 stereo.wav -c 1 mono.wav avg -l Maybe not useful for you if you need to do realtime continous recording. Or something like voice recognition. But arecord allows mono recording from a stereo device. Perhaps look at it's source for an example. My Korg MR-1000 is a stereo device that only records in stereo. But it's a stereo field recorder, that afaik doesn't run linux. HTH - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Donation Recommendation? - 24-bit sound card
I picked up a Delta 44 (pci / ice1712) off of craigslist for $100. I would check to see what they need first. The Delta is a fine card, but by the time you add in a microphone preamp, headphone preamp, and stuff just to record / playback stuff. Not to mention the costs of cables and adapters relative to a typical household existing items. Recording wise 24 bit gives you a greater dynamic sampling range. But for most people it's not needed for playback as your CDs and DVDs are already in a 16 bit format. And the benefit on playback is minimal. Aside from generally being a better soundcard, rather than from the media that it will be playing. Are you looking for a card that only does 24 bit? Most that I know of do 16 and 24 bit. And the only 24 bit media I have is that that I created myself. I'm going to donate a 24-bit capable sound card to the mpd project to add 24-bit support. Can anyone recommend an inexpensive one that is 24-bit capable and PCI, PCIe, or USB? - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA
It looks like you have all of the parts as far as kernel modules loaded. What are you trying to use the the card with? What does /proc/asound/ say about the card? cat /proc/asound/cards speakertest -c 2 -D hw:0 (change the 0 to match your cards index number, since usb probably isn't the primary unless all other soundcards are disabled somehow, or you re-indexed it in your alsa configuration. Or it's just not class compliant / supported) HTH, James On Fri, 22 Aug 2008 20:01:56 +0200 V Gabriele De Palo [EMAIL PROTECTED] wrote: this' s my # lsmod|grep hci command output: # lsmod|grep hci ahci 28804 10 libata160112 4 ata_piix,pata_acpi,ahci,ata_generic uhci_hcd 26896 0 ehci_hcd 37644 0 usbcore 147308 8 snd_usb_audio,snd_usb_lib,uvcvideo,ndiswrapper,usbhid,uhci_hcd,ehci_hcd Please, I need an help, if you can. sorry for my English too! Ciao e grazie. gabriele - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA
It looks like this might be your issue: La periferica di riproduzione è hw:0 I parametri dello stream sono 48000Hz, S16_LE, 2 canali Using 16 octaves of pink noise ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'defaults.namehint.extended' ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: Nessun file o directory ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: Nessun file o directory ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM hw:0 Playback open error: -2,Nessun file o directory [...], Not that I've had that issue. But you might only have some of your modules loaded. Or a mixmatch of versions loaded. Or some sort of udev or /dev/audio/ issues. Do you get sound on the main/default soundcard? Because it seems you got the same error on both cards. You might try hw:0,0 instead of just hw:0. Generally speaking hw:0,0 for playback and hw:0,1 for recording. I've seen issues like this (snd_ctl_open) when the /dev/ devices were not present. -- For reference here's my Mobile Pre setup. From memory and cut and paste for the most part so grain of salt. And I used to have 2 mobile pre's. You might try using a liveCD like knoppix and see if it auto detects/configures your primary soundcard. Just to verify that at least that works under linux. - There's many parts to an alsa setup / configuration. Listed below are some of mine. Most of these are taken care of by your distro. Or at least should be. - speaker-test -c 2 Playback device is default Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 2048 to 16384 Period size range from 1024 to 1024 Using max buffer size 16384 Periods = 4 was set period_size = 1024 was set buffer_size = 16384 0 - Front Left 1 - Front Right Time per period = 5.650262 0 - Front Left 1 - Front Right - cd alsa-driver-??? ./configure --prefix=/usr --with-cards=atiixp,atiixp-modem,usb-audio --with-sequencer=yes make make install - # FROM /etc/modprobe.d/alsa_custom # implies moving or disabling distro default entries. alias char-major-116 snd alias char-major-14 soundcore options snd major=116 cards_limit=3 options snd-atiixp index=0 options snd-atiixp-modemindex=1 options snd-usb-audio index=2 options snd-usb-audio index=3 alias snd-card-0 snd-atiixp alias sound-slot-0snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seqr-oss alias sound-service-0-12 snd-pcm-oss alias snd-card-1 snd-atiixp-modem alias sound-slot-1snd-card-1 alias sound-service-1-0 snd-mixer-oss alias sound-service-1-1 snd-seq-oss alias sound-service-1-3 snd-pcm-oss alias sound-service-1-8 snd-seqr-oss alias sound-service-1-12 snd-pcm-oss alias snd-card-2 snd-usb-audio alias sound-slot-2snd-card-2 alias sound-service-2-0 snd-mixer-oss alias sound-service-2-1 snd-seq-oss alias sound-service-2-3 snd-pcm-oss alias sound-service-2-8 snd-seqr-oss alias sound-service-2-12 snd-pcm-oss alias snd-card-3 snd-usb-audio alias sound-slot-3snd-card-3 alias sound-service-3-0 snd-mixer-oss alias sound-service-3-1 snd-seq-oss alias sound-service-3-3 snd-pcm-oss alias sound-service-3-8 snd-seqr-oss alias sound-service-3-12 snd-pcm-oss #alias usb-controller ohci-hcd #alias usb-controller1 ehci-hcd - # FROM /etc/modules snd-usb-audio snd-atiixp snd-seq-oss - # FROM /home/user/.asoundrc pcm.atiixp { type hw card 0 } ctl.atiixp { type hw card 0 } pcm.atiixp_modem { type hw card 1 } ctl.atiixp_modem { type hw card 1 } pcm.usb_audio2 { type hw card 2 } ctl.usb_audio2 { type hw card 2 } pcm.usb_audio3 { type hw card 3 } ctl.usb_audio3 { type hw card 3 } defaults.pcm.card 0 pcm.copy { type plug slave { pcm hw } route_policy copy } - I used udev to create the modules in /dev. I wasn't really given a choice with my current distros incarnation. I used to just run the ./snddevices script to mknod the audio devices. - find /lib/modules/`uname -r`/kernel/sound/ -iname '*' /lib/modules/2.6.25.9/kernel/sound/ /lib/modules/2.6.25.9/kernel/sound/ac97_bus.ko /lib/modules/2.6.25.9/kernel/sound/soundcore.ko /lib/modules/2.6.25.9/kernel/sound/pci /lib/modules/2.6.25.9/kernel/sound/pci/snd-atiixp-modem.ko /lib/modules/2.6.25.9/kernel/sound/pci/ac97 /lib/modules/2.6.25.9/kernel/sound/pci/ac97/snd-ac97-codec.ko /lib/modules/2.6.25.9/kernel/sound/pci/snd-atiixp.ko /lib/modules/2.6.25.9/kernel/sound/usb /lib/modules/2.6.25.9/kernel/sound/usb/caiaq /lib/modules/2.6.25.9/kernel/sound/usb/caiaq/snd-usb-caiaq.ko
Re: [Alsa-user] Donation Recommendation? - 24-bit sound card
Those are the two I have here. mplayer was giving me endless grief actually ripping the tracks from the DVD so I haven't yet done that in fact, but: mplayer -dumpaudio -dumpfile sound_track.wav ./source.vob Granted that my default distro supplied version of mplayer didn't work for this. I had to compile a custom version of mplayer from source. At least that's how it was for debian. At least it's not as bad as RH and others that strip out all mp3 support and such. Although I had to compile lame from source as well on debian. Mainly to get Layer III, versus Layer II type mp3's. Since my mp3 player is picky. It also solved the Audacity issue of exporting mp3's and not finding some lib. I have a number of lower grade historical archives of visual arts programs with insufficient audio. i.e. Normalized to the audio of the announcer, not the group. So I've had to edit the audio and rejoin it to the video to make it viewable (by humans) for all intents and purposes. That and their DVD audio format is 2.1 at best. HTH - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Donation Recommendation? - 24-bit sound card
mplayer -dumpaudio -dumpfile sound_track.wav ./source.vob I'd have figured that out :-) Didn't work with the 96/24 audio... Well, you could probably do the arecord method. arecord -D copy -t wav -c 2 -f S24_BE -r 96000 audio_track.wav (unverified syntax) Set record to the PCM / VOL device (which I can only do in aumix for some reason). Set the gain / lever for PCM to 50% / tastes. And play the track. Of course it assumes your soundcard is capable, you have copy in .asoundrc, and your soundcard can play 24/96kHz. Although that dvd-audio.sf.net tool might be a better option. Or you might just need the mencoder option of -oac copy -ovc null for mplayer. I don't have said media, so I can't play / test it out. - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA
On Tue, 19 Aug 2008 01:28:46 +0200 V Gabriele De Palo [EMAIL PROTECTED] wrote: Has someone got the Fast Track Ultra from M-Audio working with linux? I have an M-Audio Mobile Pre working. I just used it this morning to digitize some judging tapes. modprobe snd-usb-audio If it's USB class compliant it should work. But usb bandwidth is sort of limited relative to firewire / pci. You may also need ehci_hcd for usb 2.0. Otherwise you'll be stuck in usb 1.1 land with a much slower bus speed. modprobe ehci_hcd NOTE: the module name may differ depending on kernel version. And it implies other usb parts are in place. UHCI or OHCI depending on your computer. - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Which 96kHz card ?
I started to experimenting with Software Defined Radio so I need some higher quality sound card. I could buy Creative SB Audigy SE which has 24bit stereo sampling @96kHz but it is not yet supported. Can somebody here recommend me some other 24b/96kHz card which is supported by ALSA and has at least good SNR ? My M-Audio Delta 44 is supported(ice1712) and fairly nice. 4 ins and 4 outs. Unfortunately all TRS connections. I've spent as much on cables, adapters, microphone/headphone preamps, and other things, as I did for the card. Certainly higher end than most of your typical SB cards. But relative to more studio grade cards, a little lacking. The sound to noise ratio is good. My laptop hooked up to my stereo system has noticeable static, very noticeable. The Delta to the stereo only has a little noise, if you crank it and put your ear in contact with the speaker grill. I'm drooling a little over an Echo Layla 3G, but I have no idea of it's alsa/linux support level. In the meantime my mobility and battery life needs has me using a Korg MR-1000. A stand alone stereo recorder that doubles as a usb-storage device. Super Duper nice, and the converters on it(DSD), shows that the Delta isn't top of the line, but still very good. Unfortunately there's no known way to deal with DSD files in Linux. At least not known to me. Not that I've tried running audiogate in wine yet. HTH - James - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] record PCM output / no PCM capture function
I used to record streamed audio using this: arecord -D copy -f cd -t wav out.wav -d 10 Same here. My card has some bleed into the mic port so I do get something regardless. I am able to set PCM record from aumix. No ideal how to do it in alsamixer, it doesn't seem to be an option. I've only been able to set it from the gui of aumix. On Vol for some reason, and I need to set the PCM out level to 50% for the proper gain (ATIIXP). HTH - 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=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] problem updating to alsa 1.0.16
I reconfigured the alsa-driver package like this ./configure --with-cards=hda-intel,usb-caiaq --with-sequencer=yes --with-moddir=/lib/modules/2.6.22-14-rt/updates/alsa try adding a --prefix=/usr in there. Also since you're specifying a non default modules location, you'll probably need to run depmod -ae. And when inserting the modules you might even need to use insmod /full/path/to/modules. I used to get by with that if I insmod'd the modules from alsa-driver/modules/. Although autoloading doesn't quite work that way, so you have to insmod them in a specific order. To avoid those headaches, just use --prefix=/usr and let it over write whatever got installed with the kernel/distro by default. Much simpler. Although potentially screwing things up if the old works better than the new. But it's not like it's not going to boot if you screwed up the audio. Possible perhaps, likely, probably not. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] recording from /dev/dsp
record -D copy -f cd -t wav outfile.wav ecasound -i:/dev/dsp -o outfile.wav I found a way around this. Sort of. Since the files do get cached by the web browser, even if they don't finish downloading. I extracted their URLs from about:cache for the disk cache. about:cache - Disk cache device - List Cache Entries File - save page as. grep -i .mp3 ./saved_as_file.html tempfile1 grep -o http://[^]*\ ./tempfile1 | sed 's/\//' tempfile2 grep -i studioauditions.com ./tempfile2 tempfile3 wget -c -i ./tempfile3 There's probably a simpler way, but this one kept me from having to go to every individual file to redownload them or save as from about:cache. Anyway it works. And for some reason the flash player varies the volume level for various samples, which doesn't seem to be noticeable when playing the downloaded .mp3's from mplayer. Anyway enough rambling I guess. Couldn't get arecord to work good enough for either of my soundcards(usb_audio atiixp). It did sort of work for atiixp, but the resultant wav wasn't anything close to what actually made it through the speakers the first time around. - James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] recording from /dev/dsp
record -D copy -f cd -t wav outfile.wav ecasound -i:/dev/dsp -o outfile.wav One question. Is there an alsa dummy driver/package that might capture this through the above methods in it's intended form? I realize I wont hear anything locally. But it'd be nice to capture it without distortions, so I can play it on other soundcards and machines with better soundcards. And not loose any quality in the process. Thanks, James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Metallic recording in cs46xx on Thinkpad T22
Each time you start to record you have a 10% chance of having the recording completely distorted and having a metallic sound. I know that sound. And it is quite ugly. On my snd-hda-intel board(nVidia MCP61), I have to increase the number of periods to overcome this sound. default of 2 increased to 3 and all was fine, in jackd -n 3. For arecord you might look at different than default buffer/period sizes. It's basically a latency issue. Other considerations are to give the audio group realtime permissions so things like ethernet traffic doesn't cause clicks and other distortions in the sound. /etc/security/limits.conf HTH, James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Metallic recording in cs46xx on Thinkpad T22
Sadly, I've already tried all of these to no avail. What recording application are you using? I've had issues where audacity would give me that metalic sound and ardour+jackd would not. And vice versa. Depending on versions and whatnot. Beyond that I really can't offer any more insight without additional info. Like alsa version, kernel version, contents of /proc/asound/cards, .asoundrc, and whatever else might apply. Aside from upgrading to the latest kernel and latest version of alsa. It might be a known and already fixed issue. For audacity, most times I end up compiling it from source with the --with-portaudio=v19 parameter to work around various issues. Although it looks like debian caught on and now supplies a version with that option. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] recording from /dev/dsp
record -D copy -f cd -t wav outfile.wav This does not seem to capture any of the sounds from /dev/dsp. ecasound -i:/dev/dsp -o outfile.wav Nor does this. It's been a while since I've done this, what am I missing? Or is there something about usb-audio the prevents this from working? Or some ./configure option I missed at compile time? I know I've recorded what was coming out the speakers directly from the /dev/ device before. I just can't remember how. I'm trying to capture some streaming audio. .asoundrc below - pcm.atiixp { type hw card 2 } ctl.atiixp { type hw card 2 } pcm.atiixp_modem { type hw card 3 } ctl.atiixp_modem { type hw card 3 } pcm.usb_audio2 { type hw card 0 } ctl.usb_audio2 { type hw card 0 } pcm.usb_audio3 { type hw card 1 } ctl.usb_audio3 { type hw card 1 } defaults.pcm.card 0 pcm.copy { type plug slave { pcm hw } route_policy copy } - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] recording from /dev/dsp
Whether this works will depend on your hardware. Some devices support capturing the audio output, some don't. Well, I swapped it around so that the onboard sound was card 0 and it works that way. But the quality of what gets recorded is bad, actually hideous is more appropriate. From studio quality to sounding like there's a waterfall in close proximity. After normalizing to bring it to audible levels. Not quite what I want. Perhaps there's some other way to get flash audio extracted that isn't as hideous. arecord -D hw:0,0 -t wav -f dat tempfile.wav Works but hideous. As I notice a button for downloading an .mp3 of the clip in the flash. Which doesn't seem to work in linux. As thoughts of booting vista makes my skin crawl... http://www.studioauditions.com/jamroomsessions_home.php - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA utility aplay/arecord error
/home/saurav # aplay -D plughw:0,0 Vincent.wav /home/saurav # speaker-test /home/saurav # arecord -D plughw:0,0 -t wav file.wav have you tried to be a little more specific on your usage? $ speaker-test -c 2 -D hw:0 $ aplay -D hw:0 -t wav -f S16_LE -r 48000 Vincent.wav $ arecord -D hw:0 -t wav -f S16_LE -r 48000 file.wav IME those apps don't have very good psychic abilities. And the defaults they assume may or may not be supported by your soundcard. And/or may not match the soundfile you're trying to play. Kind of neat to make a screetching trumpet sound like a tuba at -r 22050. Or at the 8000 default. Although probably not the intention. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Newbie question - cannot get sound to work in Ubuntu
Anyway, I tried your suggestion and it recreated asound.state. Exactly the same as the previous one (diff gives no changes), and alsamixer gives the same error message. By the way, the USB sound card is disconnected, so that's not what's messing things up. Some distros do the alsactl save as part of the shutdown sequence(/etc/rc?.d/K*). If it doesn't then once you deleted asound.state, it should not be recreated. Not until you run alsactl save. At least that's how it works in debian. I know this because if asound.state exists, then it doesn't restore the mixer settings that were set and saved in some other mixer like aumix. YMMV I still think you're missing the /dev/'s though. Or maybe you just need the /etc/modprobe.d/ configuration. Multiple cards can be tricky. From what I've gathered, most of your errors are cannot open device, or device does not exist. If you can't get past that, they you probably wont be able to do anything with that device. I have different cards, but this is how I set mine up. Usb audio does require usb support. And I've been configuring mine custom-ish for a while. There's probably an alsaconf in the alsa-utils package that sets this stuff up for you. But I never got it to work for me back in the day. So I almost always set things up long hand. Fortunately with usb sticks, that's just a cp or cut and paste away. /etc/modprobe.d/alsa_custom #*** alias char-major-116 snd alias char-major-14 soundcore options snd major=116 cards_limit=4 options snd-atiixp index=0 options snd-atiixp-modemindex=1 options snd-usb-audio index=2 options snd-usb-audio index=3 alias snd-card-0 snd-atiixp alias sound-slot-0snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seqr-oss alias sound-service-0-12 snd-pcm-oss alias snd-card-1 snd-atiixp-modem alias sound-slot-1snd-card-1 alias sound-service-1-0 snd-mixer-oss alias sound-service-1-1 snd-seq-oss alias sound-service-1-3 snd-pcm-oss alias sound-service-1-8 snd-seqr-oss alias sound-service-1-12 snd-pcm-oss alias snd-card-2 snd-usb-audio alias sound-slot-2snd-card-2 alias sound-service-2-0 snd-mixer-oss alias sound-service-2-1 snd-seq-oss alias sound-service-2-3 snd-pcm-oss alias sound-service-2-8 snd-seqr-oss alias sound-service-2-12 snd-pcm-oss alias snd-card-3 snd-usb-audio alias sound-slot-3snd-card-3 alias sound-service-3-0 snd-mixer-oss alias sound-service-3-1 snd-seq-oss alias sound-service-3-3 snd-pcm-oss alias sound-service-3-8 snd-seqr-oss alias sound-service-3-12 snd-pcm-oss # *** /home/user/.asoundrc # *** pcm.atiixp { type hw card 0 } ctl.atiixp { type hw card 0 } pcm.atiixp_modem { type hw card 1 } ctl.atiixp_modem { type hw card 1 } pcm.usb_audio2 { type hw card 2 } ctl.usb_audio2 { type hw card 2 } pcm.usb_audio3 { type hw card 3 } ctl.usb_audio3 { type hw card 3 } defaults.pcm.card 0 # This is good enough to use the cards and have their drivers autoload when you try to use them. If I want alsa native apps to use the non zero card, then I mearly change the number on the defaults.pcm.card line. OSS type apps will almost always default to 0, unless you use aoss. Not that much of an issue since you can always change the index numbering to make whatever device be device 0. The asoundrc defaults line is just a lot fewer lines to change, and doesn't require restarting alsa. /etc/init.d/alsasound stop|start Not that you have go the defaults route. Many existing apps let you specify which card to use. By either it's hardware number or it's asoundrc label. $ mplayer -ao alsa:device=hw:0 (or device=atiixp). -D hw:0 for arecord. -d, or -C or -P for jackd. And various gui preferences to help you utilize secondary cards and friends. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Newbie question - cannot get sound to work in Ubuntu
I'm a bit confused about the rest of your comment. The only file called devfs is a directory which has two subdirectories, neither of which seems to have anything interesting in it (one is empty). And I can't find a file called snddevices anywhere, but might have mistyped (I'm not currently on the Linux machine and can't get to it to check up). In the alsa-driver package (alsa-project.org) there's a script named snddevices. It's in the root path of that tarball. Basically it creates the devices your drivers/applications need to access to do sound. Many distros set this up for you if you install the needed packages. But assuming a more basic LFS approach, you need to set them up yourself. You seem to be missing those devices. Hence the snd_ctl_open error. ls -al /dev/* | grep -i audio | wc -l 23 lsmod | grep -i snd | wc -l 21 That's what mine lists. Various /dev/ devices. /dev/mixer* /dev/sequencer* /dev/dsp* /dev/audio* If devfs or udev didn't create these for you, then you're left with the old mknod methods. Which the script alsa-driver/snddevices uses to create the devices. pgrep udev pgrep devfs (if you don't get a pid number, then it's not running) Otherwise you may just need to: apt-get install alsa alsa-base (and various other alsa* packages.) Or run the snddevices script. find / -iname '*snddevice*' HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Delta 1010LT and /dev/dspN
Anyone else have any experience with this kind of thing? IF /dev/dsp is already locked, aoss isn't going to help. Unless you redirect it to another device not in use / locked. Also aoss does NOT cover any children launched by the app started with aoss. i.e. Firefox, any popups are not covered by aoss even if you start firefox with aoss. And there's other apps that don't seem to play well with aoss like festival. From a users perspective anyway. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Newbie question - cannot get sound to work in Ubuntu
Typing alsamixer in a terminal window gives: alsamixer: function snd_ctl_open failed for default: No such device This error normally happens when it can't find the devices. Is your user in the audio group? Are you running udev/devfs? If not, did you run ./snddevices ? Are the sound modules loaded? lsmod HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Monitoring arecord.
Is there any way to monitor arecord's progress? In terms of time recorded, space used, and mixer levels? I've got two like USB devices and can only record from both at the same time with arecord. I would like to able to monitor what is recorded, so I can adjust the gain / mic level if it's too weak, or too strong. Thanks - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Recording from multiple interfaces
I recently purchased a second M-Audio Mobile Pre. So I have two usb devices. I wish to record from both interfaces at the same time. One has a phantom powered LDC on it, the other a batterybox enabled electret mic. I'm not so much worried about sync issues at this point. I'm mainly just trying to get a side by side mic comparisson. So far I have been able to accomplish this task with multiple threads of arecord. Is there anyway to setup an asoundrc to allow me to record from both devices with jackd+ardour, or just audacity? At the moment I'm only able to record from one device at a time with those apps. Thanks. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Laptop soundcard recommendations?
Can anyone recommend a PCCard/Cardbus soundcard, or possibly a USB card supported by alsa and which you've been able to run with low latency? The USB bus speed probably isn't going to ensure low latency. Most USB soundcards seem limited to two channels and 48kHz. I'd recommend a PCCard/Cardbus or Firewire device. I also have some M-Audio devices. They sound good and have linux drivers. My Mobile Pre which is USB doesn't sound as good as my Delta 44 which is PCI. But still sounds loads better than the onboard soundcard of my laptop. Is there any reason you're wanting to use something other than the onboard soundcard? Aside from most of them sounding about as low end as one can get. I'm gonna assume that your java is compiled, and not interpreted at run time. And that you've stripped your system down to ensure low latency. No autofs, dbus, avahi, apache, mysql, exim, cups, proftp, cron, atd, portmap, nfs, running while you're making said demo. And that audio has been given realtime permissions at the user level. Plus a low latency kernel. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] can't get mic to work on snd-intel8x0
When I do arecord -d 10 -f S16_LE -r 48000 -c 2 -t wav foobar.wav and I literally put one of the speaker at full volume in one of my ears I can hear a very vere low volume version of myself. That's always a good sign. _REFOUT=High-Z High-Z is high impedence(Mega Ohms). Generally meant for guitars that plug directly in for recording. Put that on a mic and if you don't have awesome gain, it's gonna make that signal very weak. If you can change that setting, then do. That is if you're not playing electric guitar as the input source. $ aumix -q That's what I normally use to show me my settings. Mic + iGain need to have values other than 0. At least that how it used to work for me. Aumix doesn't seem to play well with multiple soundcards. So I don't use it much anymore. $ amixer scontents A bit verbose, but it should give you some details. Check out the Capture, Mic, and Mic Boost settings. $ alsactl store (or restore) To save or restore whatever settings you've adjusted. My distro(debian) checks for these stored settings before it'll even think of restoring them from some other app like aumix. Basically if /etc/asound.state exists, mixer settings are restored from there. $ alsamixer -c 0 To adjust mixer settings the alsa way(F1 for hotkey list). For some settings, it's the only way to access them. At least that was the case for some headphone jacks on dell laptops. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] can't get mic to work on snd-intel8x0
Simple mixer control 'V_REFOUT',0 Capabilities: enum Items: 'High-Z' '3.7 V' '2.25 V' '0 V' Item0: '0 V' What type of mic are we talking? Electret type mics(the kind that most PC style mics seem to be/use) need a voltage(called a bias voltage, or plug-in power for voice recorders). Without the voltage you need a battery box. Set it to 3.7V, if you can. 2.25 V should also work. 0V and High-Z would be my last choices. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Help with arecord
$ arecord -d 10 -f S16_LE -r 48000 -c 2 -t wav -D hw:0 foobar.wav Recording WAVE 'foobar.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo arecord: pcm_read:1347: read error: Input/output error (-c 1 gives a different error). The I/O error is an issue. What does dmesg or other /var/log/ areas say about the error? (See: http://pastebin.ca/903687) Everything seems fine there, EXCEPT for the /etc/asound.conf. It appears to explicitly state hw:0,0, which is the playback device(and only playback). hw:0,1 would be recording. Or just hw:0 if you want both. Or I could be wrong. You might try removing / moving / renaming /etc/asound.conf. And also .asoundrc. To see if it works without any configuration. If it does then there's your problem. If that's the case, then changing hw:0,0 to hw:0 in /etc/asound.conf might be a solution. You're probably better off without a global conf if you're only using one user anyway. Other than that you appear to have some sort of hardware issue. Given the I/O errors. Perhaps an IRQ conflict? Reminds me a little of the disable PnP and UPnP options in the CMOS/BIOS to get around it days. Might try that as well. Don't fret alsamixer much. It is a bit cryptic from a first exposure POV. The L R stuff means that it is indeed stereo and has 2 channels. intel_8x0 is quite old and well supported afaik. So upgrading alsa versions probably wont help. hda-intel on the other hand... pciids.sf.net says those vender:device numbers correspond to: 8086:24d5: 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller although the sub-device doesn't have an entry yet. There's a dsnoop entry/option for recording devices in .asoundrc. But I normally don't have to specify anything there as the defaults work fine for me. But if you're gonna travel through google and the alsa wiki, dsnoop might help you narrow down what you're looking for. I don't normally use arecord. And don't normally have any .asoundrc or /etc/asound.conf as long as I have a single sound card. I normally use audacity to record. Or jackd and ardour. Depending mainly on which one has a buggy distro/svn version on any given day. I prefer ardour, since it saves directly to hard drive. So I don't have to wait for 3.5 minutes for it to save it to the hard drive after recording for over an hour. Especially if I need to record again in the coming hour. I generally stop/start recording every hour-ish so the file sizes stay under 1GB. Which helps a lot come edit time. Anyway give audacity a try, if it works fine, then something might be out of whack in your asoundrc or arecord options. One thing just popped into my head though. Is your user in the audio group? groups user || cat /etc/group | grep -i audio /// Otherwise give arecord a try as root to see if it also errors out. Just a thought. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Help with arecord
strace arecord -d 10 -f cd -t wav -D copy foobar.wav try: arecord -d 10 -f S16_LE -r 44100 -c 2 -t wav -D copy foobar.wav or arecord -d 10 -f cd -t wav -D hw:0 foobar.wav Make sure copy is the right alias in your .asoundrc, otherwise you might need to use the card number (/proc/asound/cards (-D hw:0)). The -f S16_LE -r 44100 -c 2 is the long hand of -f cd. Long hand is easier to modify, if you want to use a different rate, or only one channel(mono). - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Help with arecord
strace arecord -d 10 -f cd -t wav -D copy foobar.wav try: arecord -d 10 -f S16_LE -r 44100 -c 2 -t wav -D copy foobar.wav or arecord -d 10 -f cd -t wav -D hw:0 foobar.wav I tried the second and got: $ arecord -d 10 -f cd -t wav -D hw:0 foobar.wav Recording WAVE 'foobar.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Warning: rate is not accurate (requested = 44100Hz, got = 48000Hz) please, try the plug plugin arecord: pcm_read:1347: read error: Input/output error and I get a 44 byte foobar.wav file. My .asoundrc is: pcm.copy { type plug slave { pcm hw } route_policy copy } I'm not sure if that's the right syntax. Perhaps something like: pcm.copy { type plug pcm.slave alias of actual hardware } Or maybe pcm hw:0 for your version. .asoundrc is not my primary language, and I am but a lowly user. However with the previous recommendations, I got a 1.7MB audio file. But I have a mic and recorded sound in mine. I tried your conf and the hw:0 version of your conf. And both worked on my setup. Perhaps you have some other syntax issue in the asoundrc. Try moving that part to the top to get it interpreted before it runs into an error that stops it from interpretating. (just guessing though, any missing { { { or } } }'s?) In either case you should be able to do it long hand. arecord -d 10 -f S16_LE -r 48000 -c 2 -t wav -D hw:0 foobar.wav (based on previous error) (might also try -c 1, since many cards only record mono anyway) Perhaps the soundcard is locked by something like artsd, esd, jackd, ... As I wonder if any of the wrappers work. artsdsp, esddsp, aoss, ... In either case you probably want to stop any sound daemons that might be running interference. Also check your alsamixer settings. Make sure the mic / capture isn't muted. And otherwise exists. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] alsa + festival + card 2
I can't seem to get festival to use any card other than the default card 0. $ echo here kitty kitty kitty | aoss festival Segmentation fault Without aoss, it generates sounds to card 0. Even with the following .asoundrc. #--- START .asoundrc ---# pcm.atiixp { type hw card 0 } ctl.atiixp { type hw card 0 } pcm.atiixp-modem { type hw card 1 } ctl.atiixp-modem { type hw card 1 } pcm.usb_audio { type hw card 2 } ctl.usb_audio { type hw card 2 } defaults.pcm.card 2 pcm.dsp0 { type plug #pcm.slave atiixp pcm.slave usb_audio } ctl.dsp0 { type plug #pcm.slave atiixp pcm.slave usb_audio } #--- END .asoundrc ---# artsdsp -m festival also segfaults. esddsp -m surprisingly doesn't segfault. But the text gets all garbled like it's multi-threading several festivals at once. It starts saying several words at the same time. I don't normally run a sound daemon at all. Except when recording audio, I'll manually start jackd. Any tips to get festival to play nice with alsa? I want the sound to go to my usb device which is piped to my stereo. If only to annoy the cat. I could reindex the drivers, but I'd rather not go that route. Any other options? /proc/asound/cards 0 [IXP]: ATIIXP - ATI IXP ATI IXP rev 2 with unknown codec at 0xd0003400, irq 16 1 [Modem ]: ATIIXP-MODEM - ATI IXP Modem ATI IXP Modem rev 2 at 0xd0003800, irq 16 2 [MobilePre ]: USB-Audio - MobilePre M Audio MobilePre at usb-:00:13.0-1, full speed Thanks, James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] High-pitched squeal from audiophile card
I have: Guitar - Behringer Xenyx mixer - M-Audio Audiophile 2496 RCA inputs When the setup is sitting there, idle and active, there is a very very high-pitched light squeal coming from the PC speakers. It's so high-pitched that only myself and so far two other people I've had listen can hear it. However, when I touch my hand to the guitar strings, the squeal completely disappears. Is it possible that there is a feed back interaction between the speakers and the guitar strings? Hope this may help Tom Sometimes guitar strings sympathetically vibrate with various electronics. Like fourescent lighting. You might check there first. Does it stop if you turn out the lights? One other thing to consider is that guiter needs a High-Z input. i.e. Input impedence == Mega Ohms, not Kilo Ohms. Like the DI of a microphone preamp. It might be that your mixer isn't doing this for you and that is the source of the issue. Alsamixer might have an option for High-Z input. But it probably wouldn't apply to RCA jacks. Does that card have 1/4 plugs? TRS / TS? I'm not a guitar player, so I not up on all things guitar. But hopefully some of this helps. - James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Soundcard only works every other login
The Mia now appears to work every time I boot up, but I only managed to do this by disabling the built in Intel sound card in the BIOS. Not an ideal solution. Sound like you might just need to index the alsa modules so they load in a specific order. Automations scripts might be working in strange ways. udev / discover / ... plus mixer restoration. It could just depend on which card get hits first in terms of restoring mixer settings, and/or loading modules. Perhaps something like this in your configuration would help. Modify to use your specific cards. You don't have to disable the card in the bios. At least in a perfect world. /etc/modprobe.d/(whatever yours is called, alsa, alsa-base, alsa_custom) #- START - alias char-major-116 snd alias char-major-14 soundcore options snd major=116 cards_limit=3 # This ensures that my onboard soundcard is card 0, # and therefor the default(unless overridden). options snd-atiixp index=0 # It seems to get assigned in sequence # whether I use it or even want it. options snd-atiixp-modemindex=1 options snd-usb-audio index=2 alias snd-card-0 snd-atiixp alias sound-slot-0snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seqr-oss alias sound-service-0-12 snd-pcm-oss alias snd-card-1 snd-atiixp-modem alias sound-slot-1snd-card-1 alias sound-service-1-0 snd-mixer-oss alias sound-service-1-1 snd-seq-oss alias sound-service-1-3 snd-pcm-oss alias sound-service-1-8 snd-seqr-oss alias sound-service-1-12 snd-pcm-oss alias snd-card-2 snd-usb-audio alias sound-slot-2snd-card-2 alias sound-service-2-0 snd-mixer-oss alias sound-service-2-1 snd-seq-oss alias sound-service-2-3 snd-pcm-oss alias sound-service-2-8 snd-seqr-oss alias sound-service-2-12 snd-pcm-oss #- END - /home/user/.asoundrc #- START - pcm.atiixp { type hw card 0 } ctl.atiixp { type hw card 0 } pcm.atiixp-modem { type hw card 1 } ctl.atiixp-modem { type hw card 1 } pcm.usb_audio { type hw card 2 } ctl.usb_audio { type hw card 2 } # MODIFY this if you don't want card 0 to be your DEFAULT # NOTE: may only work on apps that use alsa natively. defaults.pcm.card 0 # Otherwise you need to modify your indexing # to make your primary card 0, # which doesn't have to be your onboard sound. #- END - HTH, James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Weird microphone issue
High-Z 3.7 V 2.25 V 0 V I have no idea what does the first one means. Does it provides any voltage? Not sure about the voltage. But things like electric guitar need high-Z (high impedence). In the Mega Ohms. As opposed to your standard microphone preamp that only has a few Kilo Ohms of impedence. If that much. I've never seen selectable voltage on any of my cards. Not that anything sold at WalMart would have that. The DC bias on the old SB cards was about 5V iirc. The GS battery box I formerly linked to provices 9V, which supposedly brings the input level up to line level. For those of us who have cards with no voltage. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Weird microphone issue
Nothing that could construed as phantom voltage (which is normally 50 volts in the broadcast industry) is visible for either the intel_8xx on the motherboard, or for the audigy 2 value I use for everything. I also have an sb16 but don't recall it as having such a feature either. Phantom Power (+48V) != bias voltage (+5V) Although they function similarly in that the mics that need them DON'T work without them. A bias voltage imples a high noise floor. Especially when you consider the RFI nightmare that exists in the internals of a computer. So not all cards provide it. http://www.epanorama.net/circuits/microphone_powering.html http://en.wikipedia.org/wiki/Phantom_power http://en.wikipedia.org/wiki/Electret_microphone - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Recommend a good XLR/line/instrument interface (USB or PCI) ?
Not really insight suggestions: But if you can negotiate with PCI try the M-Audio delta series. I've got a delta 1010lt and it works great. There are smaller models, the delta44 and delta66. They have XLR, chinch and digital I/O and MIDI in/out. My card cost about 300 EUR (about 6 years ago). Quality is fine. I have the Delta 44. It's nice in that the microphone preamp feeds into the card. So you can choose any preamp. But it doesn't have it's own preamp(s), so that's extra ca$h out of pocket. But when you consider that a great preamp can run $2,500. And that a $50 per channel builtin preamp probably wont suit your needs, it's an acceptable cost. My CDs have never sounded as good as they do on the 44. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Weird microphone issue
Problem is that I'm able to use an old (crappy) labtec microphone, but the two different headset microphones I tried, don't pick up a thing. I've tried the headsets on another computer so I know the headset mics are working. You're probably missing the DC bias aka bias voltage aka plug-in power. You can get a battery box to overcome that issue. Not that missing it is a bad thing. Many use a battery box with the mic going to the line input to avoid the high background noise of the bias based mic port. It's the bias voltage that lets you use a pair of headphones as a mic for some mic ports. http://www.giant-squid-audio-lab.com/gs/gs-batterybox.html One of many options. They tend to run much more than the mics because they don't get mass produced. You could get an Art Tube MP and an SM57(used) for about the same money. Plus cables and adapters. My old laptop didn't work with mono plugs, but did with stereo plugs. I have an old radio shack mic with it's own plugin power / battery. Which has a mono plug which didn't work on that machine. But plugging in a pair of headphones did work. Over the ear headphones anyway. The in ear ones don't produce enough of a signal to be of use. Hardware: ALC888 on a Shuttle XPC SN68SG2 using the snd-hda-intel driver with a model=6stack-dig option. All the other functions on the soundcard that I've tried are working. My MCP61 uses the snd-hda-intel module and does have a bias voltage. And works with either mono or stereo plugs. And can actually record stereo with a stereo plug. Unlike my newer laptop that takes either stereo or mono, but only records mono. As opposed to my old laptop that only took stereo, but only recorded mono. Make sure to enable all of your mixer settings. Mic + Mic Boost + Capture. There may be more than one, even if you only have one mic port. Tab Tab in alsamixer to show all mixer settings. You might also need to normalize what you recorded to hear what got recorded. Mic Boost does some of that, but IMO introduces more background noise than you get from it's gain. Using audacity or sox to normalize might work a little better for you. - James - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] MCP67 audio and alsa 1.0.15
I just got a new HP laptop. Nvidia based. It has MCP67 components from lspci. I installed alsa 1.0.15 and it does not find any sound cards. # modprobe snd-hda-intel At least that's the one for my MCP61. Does the card show up in lspci? - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] two apps using sound
Apps that use the old OSS API block the soundcard if your sound device lacks hardware mixing. Upgrade to the latest Flash plugin which uses ALSA and software mixing will work. Also try starting firefox with aoss. Note that aoss will not be wrapping any child instances (pop-ups) launched from said firefox instance. $ aoss firefox HTH, - James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Best supported cards?
Here's what I have not been able to get to work: Full duplex, Not sure about the full duplex part, as not all cards are full duplex. My M-Audio Delta 44 says Full Duplex on the box. And seems to be just that. My laptop(ATI IXP) on the other hand can do full duplex, BUT... there's some bleed over of the playing track into the recorded track. And there's just too much noise from the plug-in power / bias voltage that powers electret style microphones. You might also try disabling the onboard soundcard in the bios/cmos. Or otherwise indexing the Audiophile's alsa driver to card 0. Or using defaults.pcm.card # in your asoundrc, where # is the card you want to use by default as shown in /proc/asound/cards. # in /etc/modprobe.d/alsa_custom # my own non distro supplied modules.conf style configuration # for my Delta 44 options snd-ice1712 index=0 # for my Mobile Pre options usb-audioindex=2 # note: my ati ixp takes up 0 AND 1 because of the ac97 modem part. # end alsa_custom # in ~/.asoundrc # when I'm too lazy to use jackd or re-index my drivers # or customize whatever sound app to use something other # than the default card(0). defaults.pcm.card 2 # where 2 is the card that I want to use # as listed in /proc/asound/cards # end .asoundrc Recording in Audacity without introducing clicks into the captured audio, Getting Audacity to play without a few second delay and a bunch of clicks and pops, I've found several things to cause this on my laptop. The cable running to the mic not quite being fully shielded from RFI. Even though it may claim to be shielded. Just don't put the cable on/near your LCD or CRT. The other cause is lack of realtime scheduling. i.e. If I'm doing some network traffic while recording. Solution below (for me anyway). # in /etc/security/limits.conf @audio - rtprio 99 @audio - nice -10 @audio - memlock unlimited # end of limits.conf In conjunction with using jackd -R -d alsa -d hw:0 -r 48000(as a user in the audio group, with the above mods. Or as root without.), I can get around the clicks and pops. You may have to recompile audacity to use port audio (v19) to use alsa natively. If your distro doesn't already do that. I find ardour to be a little more reliable in the recording department. On a side note, using a cell phone in close proximity to my audio gear seems to introduce some RFI to my stereo. Even having a cell phone that's on, but not currently in use will do some of that at random intervals. I know it's the cell phone because moving it to the other room stops affecting the gear in my room, and starts affecting it in the other room. Sort of a morse code D D as heard on my speakers when stereo is on and the cell phone is close enough to it. Just random static when actually using the cell phone. Apparently cell phones put out quite a signal. HTH - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Bose audio
I don't see it in the aplay -l, but lsusb confirms its existence. Does it show up in /proc/asound/cards ? - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] default to card other than card 0 - asoundrc
I'm wondering on a couple things with asoundrc. If I want the default alsa device to be something other than card 0, is there a simple entry I can put in asoundrc to make that happen? I can get the cards to switch in the driver configuration in /etc/modprobe.d/ but I'd rather have something user based like the ~/.asoundrc file to make that change, if only in effect. Is there an .asoundrc entry to make the output of one soundcard clone the default card? copy? route? I have several interfaces, usb when i'm mobile, pci when I'm studio bound, and onboard when the main speaker device is already being utilized. Instead of rewiring the physical devices or manually reloading the drivers in a different configuration, or explicitely stating which card to use at the application level I'd like to just have all three cards behave as one. At least as far as output goes. Or otherwise be able to change the default card at the user level. Any .asoundrc tips to make that happen? Is it possible to do? Can I setup up a poor mans 5.1 system chaining a couple SB16 cards together in a similar fashion? Thanks. - 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] mic too soft (fedora 7, alsa 1.0.14, via 8235)
even with Mic Boost (+20db) on, the sound is just not loud enough. Mic in is a combination of mixer settings. Make sure you have mic + gain/capture + mic boost to have maximum input levels. It may also be that your mic requires a bias voltage(+5V) and you soundcard doesn't supply one. HTH - 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] MIDI with Intel HDA
But it seems MIDI is not supported: Many cards don't have onboard midi support these days. As long as you compiled with --with-sequencer=yes, you can run timidity to emulate midi. # modprobe snd-seq-oss $ timidity -Oi -iA Assuming timidity is installed and configured with useable sound patches. freepats, eawpats, ??? - 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] Problems with Intel HDA (82801G)
aMoRPHeouS [EMAIL PROTECTED] ~]$ cat /proc/asound/cards aMoRPHeouS 0 [Intel ]: HDA-Intel - HDA Intel aMoRPHeouS HDA Intel at 0xd044 irq 22 Have you tried: $ alsamixer -c 0 You may need to press m to mute/unmute and cursor the selection/levels. Plus tab to get to the capture/all/playback features. If you're just doing stereo out, or trying to at least, you might want to put it to pcm out, instead IEC### and whatever other options are present. If you're having to modprobe the modules manually, you may need to restart udev to get the devices setup correctly. Since you don't seem to be having access issues, that's probably not it. You could also check your bios for if your sound is disabled/enabled/ or auto. Also check that the output cable isn't plugged into an input port. As in line/speaker out, not mic or line in. Not to say that these are your issues, but when you've got the software setup as good as you can, it's time to verify that the hardware is as it should be. And of course make sure you're running the latest versions of stuff. 1.0.15 is current. Which may not be the default for your kernel/distro. My nVidia hda-intel setup gained some usefull mixer options by upgrading from 1.0.14 to 1.0.15. alsa-project.org HTH - 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] looking for a 4 (or 8) tracks firewire sound card supported by ALSA
i need to buy a 4 tracks (at least) firexire sound card http://freebob.sourceforge.net/index.php/Main_Page As far as USB, I have the M-Audio Mobile Pre and it works under alsa. But it only does two tracks. And the one I have needs some solder work done on the usb cable connector. Firewire is probably the better route to go in terms of number of available and supported devices. Most USB devices seem to dawn from the USB 1.1 days and haven't changed much if at all since. But there are a number of nice usb based microphones these days if you'd like to go that route. Most usb devices don't exceed 16 bit @ 48 kHz sampling. Firewire seems more capable than USB for higher sampling rates and multiple channels. (if only my laptop had firewire). - James - 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] 3 cards - only 2 in /proc/asound/
I'd put money on the fact that your onboard snd-hda-intel device is disabled in the BIOS. --markc Partially right anyway. It was set to AUTO. I set it to enable and it's listed now. - James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] M-Audio MobilePre - snd-usb-audio
I have the M-Audio MobilePre to use as a preamp for weak microphones. All of my current mics are currently 1/8 or adaptered to 1/4. I've gotten the device working as an audio out device. But I have not been able to capture from it. I do not have any XLR3 mics and I think it's defaulting to those ports for input. How do I get it to input from the other available ports? Output works fine so far on either the laptop or desktop. There's some weak hint of a mic on the other end on the laptop, but so far no such luck on the desktop. I have to normalize it to hear any hint of a recording, and it's 99% noise at that point. I've played with the very few mixer settings and knobs on the device with little effect. Aside from adjusting the level of the noise of the device itself. I'm at a loss now. Any documents I should be referencing? any hints for using one or all of several sound devices? For the moment I have /etc/modprobe.d/alsa-custom setup for the one usb device on the desktop. I have it setup for two on the laptop and it works, but defaults to card0 for pretty much everything. No asoundrc or dmix setups yet. qjackctl just shows two capture and two playback channels for the device which appear to be connected correctly. I just bought the device used from craigslist, and am trying to determine if it works. Aside from the faulty USB cord that came with it, it appears to work. At least for output. Thanks, - James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] M-Audio MobilePre - snd-usb-audio
I have the M-Audio MobilePre to use as a preamp for weak microphones. All of my current mics are currently 1/8 or adaptered to 1/4. Burried deep in the manual is an indication that normal computer mics are powered by current from the computer. This unit does not provide that current. Therefor the only mic I have that works on it is an old radio shack optimus mono mic that probably cost $10 a decade plus ago. It runs off of a AAA battery and only recently started working again after banging it around for bit(literally). So much for my giant squid audio lab collection. And so much for gaining stereo recording capabilities for 1/8 style mics from this unit. At least it has XLR type connectors so I can start working on my other mic collection. - James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] nVidia MCP61 - snd-hda-intel
I'm having issues with ardour and audacity, possibly more to do with jackd than alsa, or something specific to a 64 bit system. But here's my symptoms as I see them. The card works (alsa 1.0.14) or later. There's a few more mixer options in 1.0.15rc2. But when running jackd I need to use -n 3 instead of the -n 2 default for the number of periods. Only using -n 3 makes the recorded sound, sound like the mixer settings are too high, even though the recorded track is normal-ish level wise. With arecord it all sounds fine, but I can't check recording levels very easily that way. With arecord I set the alsamixer settings for mic and digital. With jackd I set the mixer levels for mic plus capture. Mic boost is not used at this point. With audacity, it sounds fine, but there's a click in the recorded track at about 120bpm. These shouldn't be performance issues as it's an AMD Athlon64 x2 dual core. 1.9GHz per core with 1GB of ram. The card is in theory capable of capturing 16, 20, or 24 bit samples at 44.1kHz to 192kHz sample rates. And in arecord it seems happy doing so. But how do I get jackd happy with the card? Is there an .asoundrc I can reference that will help? I'd like to do stereo input, 16 bit, @ 96kHz (or better). For reference I'm running jackd as follows: jackd -d alsa -n 3 -d hw:0,0 -S -r 96000 (without the -n 3 there's considerable distortions) (playback seems fine, but capture is distorted) I'm currently running the following: Debian sid - AMD64 Kernel 2.6.22.6 - from kernel.org sources Ardour 2.0.5 - via the svn branch sources Jackd 0.103.0-6 - debian package Alsa 1.0.15rc2 - from sources Any tips for making things happy? I'd like to get better sampling than my other system (laptop). I'd ultimately like to do multitrack recording which I cannot do with arecord (to my knowledge anyway). But I'd be happy with recordings in a gui app equal to those captured by arecord at this point. Thanks, - James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] problems with dmix, a52 and upmix
0. without dmix I cannot share the sound card which is a big problem! There are plenty of ways to share sound without dmix. artsd, jackd, esd, aoss, and probably others. As long as the apps in question use alsa natively you don't even need to have those to have shared sound. At least not since 1.0-ish versions of alsa. On my older and slower laptop having a dmix configuration caused it to have poor audio latency with the symptoms you've described. Not that I've done anything with spdif, or surround sound type setups. But getting rid of ~/.asoundrc completely got rid of that issue for me. Starting oss based apps with aoss also helps. $ aoss firefox. Not that the children processes (popups) get started with aoss. HTH - 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