Linux-audio-dev

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7501 discussions

[linux-audio-dev] LAD-Conference in german "Linux-Magazin"

by Frank Barknecht

Hallo, in case nobody noticed it yet: The 06/2003 issue of the german "Linux-Magazin" has a feature on our LAD-conference in Karlsruhe by Stefan Klett. Nice work, Stefan. ciao -- Frank Barknecht _ ______footils.org__

22 years, 1 month

[linux-audio-dev] Rectifier algorithm and Exponential FM question

by nickt

Hi, Sorry if this is the wrong list. I have never found a list with people who can help me with these kind of questions. I am trying to figure out how a rectifier works in code. I have created something like it using tanh() distribution ( which will turn a sine wave into a sort-of-square wave). I can also achieve it with hard clipping/saturation. In the same sense I'd also like to know how a smooth operation works as well. I can not find any source code or explanations to help me figure out the algorithm yet. My other question is how are exponential signals generated? I want to experiment with exponential FM. My only understanding of Exp. Fm is that it sweeps half down and full up ( at 440hz, it would sweep 220hz down and 880 hz up). I haven't much of an idea how this is achieved. How do I convert a sine wave into an exponential signal for Exp FM?

22 years, 1 month

[linux-audio-dev] sonic foundry + sony

by Paul Davis

all my interactions with sonic foundry were unpleasant, and i have no idea what "rich media" means, but now we have another 800lb gorilla in the tent, and another valuation of how much its worth to create a program like Acid: US$18M. http://news.harmony-central.com/News/2003/Sony-Sonic-Foundry.html

22 years, 1 month

[linux-audio-dev] JACK 0.70.4 release

by Taybin

JACK 0.70.4 release JACK is a low-latency audio server, written primarily for the GNU/Linux operating system. It can connect a number of different applications to an audio device, as well as allowing them to share audio between themselves. Its clients can run in their own processes (ie. as normal applications), or can they can run within the JACK server (ie. as a "plugin"). JACK is different from other audio server efforts in that it has been designed from the ground up to be suitable for professional audio work. This means that it focuses on two key areas: synchronous execution of all clients, and low latency operation. CHANGES: qjackconnect/jack_lsp bug fixed. now can use either SysV or POSIX shm methods. Linxu 2.4.4 is required for the POSIX method. Defaults to SysV, but can be switched with a ./configure option (--enable-posix-shm) new --enable-optimize configure time flag, for compiler optimization. jack_set_buffer_size_callback() de-deprecated. added port metering API. added option --timeout (OR -t) for client timeout. The default time is as before: 500 msec. jack_lsp now has usage (-h or --help) and version (--version) options. various bug fixes and documentation updates. CAVETS: The shm code requires the tmpfs kernel module either loaded or compiled in. A tmpfs filesystem must be mounted as well. This can be done by adding: tmpfs /dev/shm shm defaults 0 0 to /etc/fstab and running "mkdir /dev/shm; mount /dev/shm". In versions of linux prior to 2.4.4, tmpfs was known as shmfs. Client programs compile with the previous stable release (0.61.0) will need to be recompiled to work with 0.70.4. Client programs now need to also link against librt. That library should be automatically included via pkg-config. Taybin Rutkin

22 years, 1 month

[linux-audio-dev] more LAD meeting photos...

by Joern Nettingsmeier

hello * ! frank neumann has sent me some photos he took during the lad meeting at the zkm in march. they have been uploaded to the lad page and can be viewed at http://www.linuxdj.com/audio/lad/contrib/zkm_meeting_2003/photos/frank_n_ph… have fun. -- All Members shall refrain in their international relations from the threat or use of force against the territorial integrity or political independence of any state, or in any other manner inconsistent with the Purposes of the United Nations. -- Charter of the United Nations, Article 2.4 Jörn Nettingsmeier Kurfürstenstr 49, 45138 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)

22 years, 1 month

[linux-audio-dev] Synchronizing two Delta 1010 with ALSA ?

by Benno Senoner

(Sorry for reposting to LAD, I've already sent this to alsa-user but I'm unsure which list is more appropriate so I'm trying both) Hello, I have a machine with two DELTA 1010LT cards and they work very well with ALSA (using ALSA binaries from Planet CCRMA). I would like to write a single threaded application that is able to write on all out channels (analog outs 1-8 of card 1 and 2 = 16 channels) simultaneously. The card's manual says that in order to synchronize two cards one should select the first card as master and set the internal clock generation (I do this via envy24control) and then connecting the SPDIF-OUT of card 1 to the SPDIF-IN of card 2 and set the card 2's clock source to SPDIF-IN. So far so good (from the hardware point of view) I managed to write ALSA code that writes to a single card: it opens the PCM hw:0 device selects 10 channel interleaved S32_LE mode and then writes out some data. This works perfectly for a single card. Now my question: I would like to extend this app so that it writes to both cards simultaneously without going out of sync. I assume that the SPDIF writing and clocksouce selection provides this from a hardware point of view, but I am unsure how to proceed from with the application code since the ALSA docs do not mention such a scenario. currently for a single card setup, I do the usual card setup and then sit in a while loop that does while(1) { snd_pcm_writei(pcm_handle, audiobuf, MY_PERIOD_FRAMES); } Is the following way to extend the app to two cards correct ? call the functions to setup card 1 (hw:0) call the functions to setup card 2 (hw:1) while(1) { snd_pcm_writei(pcm_handle1, audiobuf1, MY_PERIOD_FRAMES); snd_pcm_writei(pcm_handle2, audiobuf2, MY_PERIOD_FRAMES); } (where audiobuf1 is the buffer sent to card 1 while audiobuf2 is the buffer sent to card 2) Or is some special "sync start" procedure needed ? I remember ALSA had/has provides some functionality in this regard but I was unable to find documentation about it. Thanks in advance for your help. PS: does the DELTA cards only support the interleaved mode ? The model seems a bit inefficient since in most apps the data sources are non interleaved audio streams. (eg a HDR app that outputs each audio track to a different audio channel) cheers, Benno http://linuxsampler.sourceforge.net ------------------------------------------------- This mail sent through http://www.gardena.net

22 years, 1 month

[linux-audio-dev] pawfal [OTish]

by Dave Griffiths

Just a quick notification to you all that pawfal (the webste for arty free software types) has been reborn as a wiki, and is open for all to use and abuse. Feel free to add information of your own - we're especially interested in people using linux and/or free software for artistic purposes to put links to their work there, you can create your own wiki page etc etc. http://www.pawfal.org For more information, if you are new to wiki's: http://www.pawfal.org/index.php/PawfalFaq cheers, dave

22 years, 1 month

[linux-audio-dev] [OT?] glibc problems

by Ivica Bukvic

Hi all, I know this is off-topic, but at this point I could use any help I could get since I am just about ready to do a clean reinstall (Something that I do really not feel like doing right now). I had Mdk 9.0 and upgraded it to 9.1. Since, when I have dynamically linked apps to the /usr/lib/libc.so.6 (or /usr/lib/libc.so) I always get the same error: Error while loading shared libraries: /usr/lib/libc.so.6: invalid ELF header I tried reinstalling 9.1 glibc and glibc-devel and that did not fix it. I also tried installing libelf rpm for mandrake and that did not do a thing. Finally I even installed the newest cooker version of these two files and I still come up with the same error. This brings me to a conclusion that the problem lies somewhere else, but where? Rebuilding of glibc from srpm eventually fails with some weird error (can't remember it any more), so that option is out of question. Furthermore, what is really odd, is that the clean install on another machine of the same distro does not have this problem, yet the files in question libc.so (and other related ones) are IDENTICAL to the ones I have on this machine (at least they were before I installed the latest glibc from the cooker). I also checked binutils package and it is the same on both machines. So what gives? Does anyone have any suggestions as to how could I go about resolving this issue? Any help at this point is greatly appreciated! Sincerely, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico

22 years, 1 month

[linux-audio-dev] General strategies for RT Audio Thread

by Alexander Lenhardt

HI! I'm trying to write very simple software synthesizer under Linux just for my own educational purpose. but one problem i came over is the following: I made the audio thread shced_fifo with max priority. My GUI thread is sched_other. To allow the GUI to rerspond, I thought i could just put the audio thread to sleep for the playing length of the buffer (7ms). But the minimum tick interval of the linux scheduler is 10ms and its not even sure that the thread will sleep for exact 20ms. So how can I avoid underruns if the timing is so unreliable? Thanks in advance Alex

22 years, 1 month

[linux-audio-dev] [ANN] jack_input 0.1 for AlsaPlayer

by Andy Lo-A-Foe

Hi, jack_input is an AlsaPlayer input plugin. Connect any JACK output port to the alsaplayer input ports. Pros: * Volume / pan control through alsaplayer * Visualize your JACK streams with alsaplayer scopes * Feed live input to alsaplayer (line-in, mic, etc) * Export jack streams to other audio subsystems supported by alsaplayer (not recommended, unless output is through the same input hw) * ??? Cons: * alsaplayer's internal format is still 16-bit * possible distortion if machine is heavily loaded (input overruns) * high latency caused by aggressive internal buffering * not configurable enough Usage: $ alsaplayer -ojack jack_input: This will load the jack input plugin and create 2 new input ports: alsaplayer_in:in_1 alsaplayer_in:in_2 Connect your output to these ports. Warning: loops are possible :-) Download: http://alsaplayer.org/jack_input-0.1.tar.bz2 Enjoy! Andy @ alsaplayer.org

22 years, 1 month

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Linux-audio-dev