Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7520 discussions

[linux-audio-dev] Re: Re: alsa, oss , efficiency?

by Kjetil S. Matheussen

lemmel: > > Le mercredi 01 novembre 2006 16:51, Kjetil S. Matheussen a écrit : >> Only the oss modules in the linux kernel are deprecated. Programs using >> the OSS api will still continue to work, currently most importantly >> because of the oss emulation module in alsa. > I knew that the oss module was deprecated, and I infered that using the OSS > mecanism was too deprecated :-P. Are sure [1]? Hmm, well, who decides... I think its pretty safe to write code for OSS and be sure it will work for a long time. Maybe longer than code written for ALSA as well, who knows. >> If you should choose between alsa and oss, and the oss version works >> just as well, or better than the alsa version, choose oss, because its a >> more portable API than alsa. > What do you mean by "the oss version works just as well, or better than the > alsa version" ? Thats a liberal quote... But sorry, I don't think I can write what I ment any clearer... (well, you can remove the comma after "well", but thats as clear as I can make it, english is not my native language either. :-) ) >> However, if I were you, I would use sndlib, portaudio, jack, or some other >> higher level audio input/output library instead of oss or alsa. > Well, all the files, that I will play, will have the same charactistics, do I > really need to bother with an hi-level API [2] ? Why not? > > [1] I noticed that a lot of applications still use OSS, and I thought it was > because the migration to ALSA was too complex. That might be one reason. But if it works well with OSS, thats often good enough. Its a shame it sometimes is hard to make OSS programs work with jack though, but thats a problem that goes for alsa programs as well. (Actually, its worse with alsa programs.) > [2] Futhermore I am afraid that an hi-level API will produce time drifts in > the playing. > No, I don't think so. Many people use portaudio, it could be your best choise.

19 years, 1 month

[linux-audio-dev] Pd Q and Faust interfaces / ICMC 2006 demo

by Albert Graef

Hi all, (Sorry for crossposting.) This should be interesting for all who want to extend their Pd with programs written in languages other than C/C++. I have created Pd plugin interfaces for these two languages: - Q, a functional programming language based on term rewriting (my own creation; see http://q-lang.sf.net). Q is a modern-style functional programming language in which functions are defined by equations. It also has an extensive collection of modules for doing graphics and multimedia. My Pd/Q external allows you to execute Q functions in order to do complicated control stuff in Pd, and it also provides a way to access Q's multimedia interfaces inside Pd. This is available as a source tarball (pd-qext-0.1.tar.gz); RPMs for Linux systems are also available. - Faust, Yann Orlarey's functional DSP programming language (http://faudiostream.sf.net). Faust's programming model combines two approaches: functional programming and block diagram composition. You can think of Faust as a structured block diagram language with a textual syntax. The resulting C++ code is heavily optimized and can compete in speed with handwritten C code. My Faust architecture file allows Faust programs to be translated to Pd externals using the Faust compiler. This makes it very easy to create new audio externals for Pd, and a bunch of examples are readily available. I have also written a Q script which generates complete Pd patches with graph-on-parent GUIs from Faust programs. This stuff can be found in the faust2pd-1.0.tar.gz package. You can find all the good stuff, including documentation and a lot of examples on the Q website at: http://q-lang.sourceforge.net/examples.html#Multimedia (see the bottom of this page). The Faust interface will also soon be available as a part of the mainstream Faust distribution. Yann and me will show Faust, Q and their Pd and SuperCollider interfaces at the International Computer Music Conference (ICMC) next week in New Orleans, so if you have an opportunity to come we hope to meet you there. (The presentation is on the very last day of the conference, Sat Nov 11th, 3:30 p.m., see http://www.icmc2006.org.) For more information please see http://faudiostream.sf.net and http://q-lang.sf.net. Enjoy. :) Albert -- Dr. Albert Gr"af Dept. of Music-Informatics, University of Mainz, Germany Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikinformatik.uni-mainz.de/ag

19 years, 1 month

[linux-audio-dev] alsa, oss , efficiency?

by lemmel

Hi everyone. for a project, we need to be able to play sound (at first look wav file), and we made several tests ; with a created stereo sound, we try to use alsa but the results doesn't fullfill our needs : sound played at the time, T, we want, and finished at the date, D=T+sound duration. (thi is a software with strict time constraints) The sound was always troncated (even with finished software such as xmms, amarok), and even randomly truncated, (sound created with audacity, and exported as WAV 16/32 bit etc). When we use OSS, all seems to be perfect. But, it seems that OSS is nowadays "deprecated", and consequently we shouldn't use OSS. What we can do ? Are our alsa results due to misconfigurations ?

19 years, 1 month

[linux-audio-dev] Alpha release of Aliki

by Fons Adriaensen

For the brave: an alpha release of Aliki (Room Impulse Response Measurement) is now available on: http://users.skynet.be/solaris/linuxaudio along with a manual that should get you started. This is basically the code used at the LAC2006 workshop, cleaned up a but. As said, ALPHA, incomplete, probably lots of bugs. -- FA Lascia la spina, cogli la rosa.

19 years, 1 month

[linux-audio-dev] [ANN] Vmwaredspjack 1.3, Snd-ls 0.9.7.6 and Jack_capture V0.3.8

by Kjetil S. Matheussen

Vmwaredspjack ============= This is the vmwaredsp program, made by Petr Vandrovec, which makes vmware work with esd or arts. This version adds jack support as well. (Unfortunately, jacklaunch (which is a similar program) doesn't work with vmware, but I think Gunter is working on it... :-) .) The program isn't always working that well, but if used with care (don't trust the output too much) and proper tuning, you can use professional windows audio software in vmware using jack for audio communication. Download from http://www.notam02.no/arkiv/src/ Snd-ls v0.9.7.6 =============== Snd-ls is a distribution of Bill Schottstaedt's sound editor SND. Its target is people that don't know scheme very well, and don't want to spend too much time configuring Snd. It can also serve as a quick introduction to Snd and how it can be set up. Changes 0.9.7.1 -> 0.9.7.6: --------------------------- -Proper debug output in case startup fails. -Fixed bug in jack audio. -Temporary remove the fft menu because its not working with the 26.9.2006 version of Snd. Bug found by Dragan Noveski. -Check for the existence of the sndfile.h header file before compiling. If it doesn't exist, snd-ls will refuse to run. Problem reported by Krzusztof Gawlas. -Make sure snd starts up even if no file was loaded during startup. Bug found by Dragan Noveski. -Really apply the workaround for the menu problem. Download from http://www.notam02.no/arkiv/src/snd/ Jack_capture V0.3.8 =================== jack_capture is a small program to capture whatever sound is going out to your speakers into a file. This is the program I always wanted to have for jack, but no one made. So here it is. Changes 0.3.7 -> 0.3.8: ----------------------- *Added the --recording-time option to stop recording after a certain number of seconds. *Quitting with CTRL-C/SIGINT writes remaining buffer to disk before ending program. Download from http://www.notam02.no/arkiv/src/

19 years, 1 month

[linux-audio-dev] Re: Alpha release of Aliki

by Ben Loftis

Hi Fons. Can you please tell us more about Aliki? I am using Denis Sbragion's DRC program to generate an impulse response file, and his suite of graphing tools to generate various views of the measurement. What is the output of Aliki? Just an impulse response file? In what way is your method different than DRC? Does Aliki use ALSA or JACK for I/O? I'll have a look at the code tonight but I was hoping you could fill me in on the "big picture". Thanks, Ben Loftis

19 years, 1 month

[linux-audio-dev] Clock/Sync tutorials, hints?

by krampenschiesser＠freenet.de

Hi, do you know any good clock/sync tutorials out there? Or do you have any good hints on this? I hope you can help me. Scar -- Finagle's Eighth Law: If an experiment works, something has gone wrong. Finagle's Ninth Law: No matter what results are expected, someone is always willing to fake it. Finagle's Tenth Law: No matter what the result someone is always eager to misinterpret it. Finagle's Eleventh Law: No matter what occurs, someone believes it happened according to his pet theory.

19 years, 1 month

[linux-audio-dev] Linuxaudio.org -- call for new member projects

by Ivica Ico Bukvic

Apologies for cross-posting. Please forward the announcement to your respective lists (if applicable). Yes, it's that time of the year. In our continued bi-monthly track of membership drives, I would like to extend an open invitation to all Linux audio projects who are not already a member of the consortium to consider joining and therefore help us continue our efforts at consolidation of Linux audio resources. The membership is free and the consortium's structure allows members to gauge their level of involvement and subsequently the time overhead it bears. In the recent months there have been a number of exciting new projects introduced to the Linux audio scene. This is very encouraging as it suggests not only continued, but also growing vitality of our platform of choice. Linuxaudio.org's mission is to help maintain this vitality by offering various programs to its membership base as well as the entire Linux audio community. Perhaps one of the most important programs is our mission to consolidate Linux audio resources and by doing so foster collaboration as well as connections with the commercial industry and outside investors. For this reason, I sincerely hope that you will consider joining the consortium. For more info on the membership, benefits, etc. please visit linuxaudio.org. We have our next new member announcement set for this coming Thursday November 2nd. Should you happen to have any additional questions and/or concerns, please do not hesitate to contact me. Best wishes, Ivica Ico Bukvic, D.M.A. Linuxaudio.org Director Virginia Tech Department of Music - 0240 Blacksburg, VA 24061 (540) 231-1137 (540) 231-5034 (fax) ico(a)linuxaudio.org http://www.music.vt.edu/people/faculty/bukvic

19 years, 1 month

Re: [linux-audio-dev] about MIDI timing...

by Chris Cannam

Me: > I'm not aware of anyone these days successfully > using Rosegarden with snd-rtctimer - if anyone out > there is, do say so. To test: * start RG (version 1.0 or newer) * go to Settings -> Configure Rosegarden -> Sequencer -> Synchronisation * change the sequencer timing source option to RTC * close configuration window, press play. It probably doesn't matter whether you have a file loaded or not. Success -> play pointer moves smoothly Failure -> system locks up solid, reboot required. If it does lock up, you may need to edit your rosegardenrc to restore the timer setting before you can run RG again. Chris

19 years, 1 month

[linux-audio-dev] [ANN] FreeBoB 1.0 released - Firewire Audio for Linux

by Pieter Palmers

Greetings, Now that JACK 0.102.20 and QJackCtl 0.2.21 being released, the FreeBoB team is proud to present libfreebob 1.0. The FreeBoB project aims to provide a generic solution for using Firewire (semi-)pro-audio devices in Linux. This release provides support for the devices based on the BridgeCo DM1000 or DM1500 chipset that are running the BeBoB firmware. For a list of supported devices, consult our website at freebob.sf.net. FreeBoB currently provides an interface library that allows firewire audio devices to be used with the JACK audio server, using a dedicated backend. This backend is included in the official JACK releases, from this version on (i.e. 0.102.20). The latest version of QJackCtl also includes support for this FreeBoB backend. MIDI support is provided through ALSA sequencer. Feature list: * Automatic detection & configuration of devices. If there are multiple devices attached to the same firewire bus, freebob merges them into one big device. The devices have to be synced externally (wordclock/spdif) so that they don't drift. Note that this release cannot setup the boxes to be synced yet, being synced is a precondition at the moment. (I tested this with 2 phase88's connected with wordclock, and this works without the need for any special setting because the Phase88 automatically chooses wordclock slave when there is a wordclock signal present. This can be different for other models). * Audio I/O on all analog channels at all sample rates supported by the device. SPDIF/ADAT I/O works in most cases (when presented as analog IO by the device). AC3 passthrough doesn't work. * Midi I/O for all midi ports the device implements, using alsa-sequencer * Round-trip latency figures around 5ms (depends on system configuration). 10ms is achievable on all well-configured machines Not supported yet: * Hardware mixing ("zero latency" mixer) * Device-specific configuration (input gain switches, sync source selection, midi control mappings, ...) * ALSA for audio IO * Special SPDIF/ADAT stream support You can download FreeBoB 1.0 at our sourceforge page: http://sourceforge.net/project/showfiles.php?group_id=117802 more info at freebob.sf.net What's next? We are working on the second generation of the FreeBoB codebase. The 1.0 release is the endpoint for the codebase that dates back to the start of the project. The 2.0 codebase is a complete redesign of the system using 1.0 as a 'golden spec'. While the 1.0 version is BeBoB-only, the 2.0 codebase is designed as a framework to support all firewire based audio boxes. The current level of functionality is almost the same for both codebases. The main difference is that 1.0 had one year of testing and 2.0 doesn't, it's still in the alpha stage. Needless to say that 2.0 will outperform 1.0 by far ;). Of course this redesign isn't for the sake of aesthetic beauty or lack of things to do... I can announce that we are already working on broadening the supported device list. Currently there is support for the Motu Traveller and the Motu 828 (through reverse engineering). There are also contacts with the DICE-II developers to implement generic support for devices based upon their chipset. As an extra, support for Metric Halo devices is also in the pipeline. Once all of these devices are supported, we will cover a very large part of the Firewire audio device spectrum. The most important void will be RME Fireface support, and for the real budget users: Behringer and Hercules devices. That's all folks! Pieter Palmers (on behalf of the FreeBoB team)

19 years, 1 month

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev