Linux-audio-dev

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4 participants
7502 discussions

Releasing source code is not enough, I think...

by Filipe Coelho

Hi there everyone, specially developers. I think we should stop assuming releasing source code is enough. [GNU/] Linux is getting more user friendly, and most users are not able to compile software, plus some distributions make it specially hard (debian, ubuntu, fedora, opensuse) by having the libs installed but not the headers. Releasing software on windows or mac, even open-source, *always* comes in a binary, and most users come from there. Now, I have a "toolchain" repository for ubuntu 10.04 with gcc4.8, python3+qt4 and a bunch of other useful stuff. I use this to get generic linux binaries that (from what I know) work everywhere. I can make a developer-oriented tutorial on how to use that, so that developers can provide linux binaries to its users. Would that be something useful to Linux Audio?

11 years, 3 months

[LAA] aubio 0.4.1

by Paul Brossier

Hi all, A new version of aubio, 0.4.1, is out. aubio is a library of functions to perform audio feature extractions such as: - note onset detection - pitch detection - beat tracking - MFCC computation - spectral descriptors This version is mostly focusing on media file input and output. Here is a quick overview of the changes. The most interesting feature in this release concerns aubiocut. Thanks to the sponsoring of Mark Suppes, the python script to slice sound steams was extended to be sample accurate, cut overlapping segments, and work on multiple channels. New source and sink objects have been added to let aubio read and write WAV files, even when built with no external libraries. This should simplify the use of aubio on platforms such as Android or Windows. Existing sources and sinks have been extended to read and write from and to multiple channels. This makes python-aubio one of the fastest and most versatile Python module to read and write media files. This release also comes with a stack of bug fixes and code clean-ups. Note: this version is API and ABI compatible with 0.4.0. Since it only adds new features to the existing interface, your existing source and binary code will keep working without any modifications. To find out more about aubio and this release: Project homepage: http://aubio.org/ Post announcing aubio 0.4.1: http://aubio.org/news/20140312-1953_aubio_0.4.1 ChangeLog for aubio 0.4.1: http://aubio.org/pub/aubio-0.4.1.changelog Source tarball, signature and digests: http://aubio.org/pub/aubio-0.4.1.tar.bz2 http://aubio.org/pub/aubio-0.4.1.tar.bz2.asc http://aubio.org/pub/aubio-0.4.1.tar.bz2.md5 http://aubio.org/pub/aubio-0.4.1.tar.bz2.sha1 API Documentation: http://aubio.org/doc/latest/ Happy hacking! Paul

11 years, 3 months

JACK and ALSA elucidations

by Gianfranco Ceccolini

Hi LADs We are working on an ARM based MOD device and while fiddling with JACK some questions appeared. I didn't know whether to post on the ALSA os JACK lists, so I decided to post here :-) When I start Jack in my PC I get the well know message: /usr/bin/jackd -P80 -dalsa -r48000 -p128 -n2 -D -Chw:0 -Phw:0 creating alsa driver ... hw:0|hw:0|128|2|48000|0|0|nomon|swmeter|-|32bit configuring for 48000Hz, period = 128 frames (2.7 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 32bit integer little-endian ALSA: use 2 periods for playback When starting it on the Beaglebone Black I get: /usr/bin/jackd --realtime -P80 -dalsa -r48000 -p128 -n2 -Xraw creating alsa driver ... hw:0|hw:0|128|2|48000|0|0|nomon|swmeter|-|32bit configuring for 48000Hz, period = 128 frames (2.7 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: use 16 periods for capture ALSA: final selected sample format for playback: 32bit integer little-endian ALSA: use 16 periods for playback - what are those "periods" in the ALSA lines? (2 for the PC and 16 for the BBB) ? I read a post by Jeremy Jongepier about the Cubieboard2 stating that the hardcoded values for the Cubie (Minimum number of periods was 4 and minimum buffer size was 1024) where unfit for realtime operation. Why is that so? - on the ArchWiki page on Jack there is a D-bus call jack_control eps realtime true stating that it "Sets JACK to realtime mode in its own internal setup." What is this internal setup? How do I address this when not using D-Bus? King regards all Gianfranco Ceccolini The MOD Team PS: We will be at LAC againd this year. I'm very eager to meet most of this distinguished community members there. We have lots of good new for the MOD in 2014.

11 years, 3 months

OpenAV : ArtyFX 1.1

by Harry van Haaren

Hey All, Its my pleasure to announce ArtyFX 1.1, with three new plugins! The plugin are distortion, feedback delay, and 4-band eq. Demo video: http://www.youtube.com/watch?v=aPQOZK-yKy8 ArtyFX page: http://openavproductions.com/artyfx/ OpenAV wishes to thank Steve Harris for authoring Barry's Satan Maximizer: Satma's DSP routine is derived from that work. OpenAV also wishes to thank Fons Adriaensen for writing the 4-band parameteric equalizer, as Kuiza uses his implementation as DSP routine. Contributions to release ArtyFX 1.1 welcomed, see the ArtyFX page for details: http://openavproductions.com/artyfx Cheers, -Harry

11 years, 3 months

plugin updates

by Fons Adriaensen

Some updates to <http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html> REV-plugins-0.7.1 * Removed G2reverb from the .so, see below. g2reverb-0.7.1 * Contains the G2reverb plugin from earlier REV-plugins. * Plugin file name is now the same as in the original (2003) version, so AMS patches using this should load without error. WAH-plugins-0.1.0 * Some cleanup, unique ID of the auto-wah changed to 1949 to avoid conflict with recent stereo panner plugins. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

11 years, 3 months

No sound with JACK over net

by Simon Price

Hi all, I'm trying to configure jackd -d net The master machine with the hardware sees my soundcard fine. All looks good as it comes up. I start the slave, which connects to the master machine fine and picks up the right settings such as sample rate. I've tried changing this on the master and they are successfully picked up on the slave so I know this is all talking properly. I also get the 'Connected :-)' message on the master so I'm pretty sure this is all OK. On the slave where I run the application all of the connections appear and look good. However, no matter what I do I never get any sound! I know the card and everything else is OK because if I use the simple alsa 'play' command on the master it comes through loud and clear. I'm using CentOS (headless) on the master which reports jackdmp 1.9.5 On the slave I'm running Mint which reports jackd 0.122.0 These are the standard downloaded packages from the Any ideas? Thank you, Simon

11 years, 4 months

[ANN] Vee One Suite 0.4.0 - A proto-beta party!

by Rui Nuno Capela

Howdy, as to no surprise (rly?) the so called Vee One Suite of old-schoolyards gets bullied down to another bump in the head: now turning into a proto-beta party? righto! again as usual, all made available in dual form: - a pure stand-alone JACK client with JACK-session, NSM (Non Session management) and both JACK MIDI and ALSA MIDI input support; - a LV2 instrument plugin. all free and open-source Linux Audio software, distributed under the terms of the GNU General Public License (GPL) version 2 or later. [1] synthv1 - an old-school polyphonic synthesizer synthv1 is an old-school all-digital 4-oscillator subtractive polyphonic synthesizer with stereo fx. LV2 URI: http://synthv1.sourceforge.net/lv2 website: http://synthv1.sourceforge.net downloads: http://sourceforge.net/projects/synthv1/files - source tarball: http://download.sourceforge.net/synthv1/synthv1-0.4.0.tar.gz - source package: http://download.sourceforge.net/synthv1/synthv1-0.4.0-14.rncbc.suse131.src.… - binary packages: http://download.sourceforge.net/synthv1/synthv1-0.4.0-14.rncbc.suse131.i586… http://download.sourceforge.net/synthv1/synthv1-0.4.0-14.rncbc.suse131.x86_… [2] samplv1 - an old-school polyphonic sampler samplv1 is an(other) old-school all-digital polyphonic sampler synthesizer with stereo fx. LV2 URI: http://samplv1.sourceforge.net/lv2 website: http://samplv1.sourceforge.net downloads: http://sourceforge.net/projects/samplv1/files - source tarball: http://download.sourceforge.net/samplv1/samplv1-0.4.0.tar.gz - source package: http://download.sourceforge.net/samplv1/samplv1-0.4.0-14.rncbc.suse131.src.… - binary packages: http://download.sourceforge.net/samplv1/samplv1-0.4.0-14.rncbc.suse131.i586… http://download.sourceforge.net/samplv1/samplv1-0.4.0-14.rncbc.suse131.x86_… [3] drumkv1 - an old-school drum-kit sampler drumkv1 is (yet) an(other) old-school all-digital drum-kit sampler synthesizer with stereo fx. LV2 URI: http://drumkv1.sourceforge.net/lv2 website: http://drumkv1.sourceforge.net downloads: http://sourceforge.net/projects/drumkv1/files - source tarball: http://download.sourceforge.net/drumkv1/drumkv1-0.4.0.tar.gz - source package: http://download.sourceforge.net/drumkv1/drumkv1-0.4.0-10.rncbc.suse131.src.… - binary packages: http://download.sourceforge.net/drumkv1/drumkv1-0.4.0-10.rncbc.suse131.i586… http://download.sourceforge.net/drumkv1/drumkv1-0.4.0-10.rncbc.suse131.x86_… see also: http://www.rncbc.org/drupal/node/770 enjoy && have fun! -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

11 years, 4 months

Re: [LAD] Audio problem for numerical optimization

by Fons Adriaensen

On Mon, Mar 03, 2014 at 11:28:49PM +0100, "Jeremia Bär" wrote: > I'm not sure I get the difference here. As I understand, optimization includes > (1) choice of suitable algorithm for the problem, (2) smart implementation, i.e. > writing code such that the compiler can apply smart optimization (e.g. aliasing) > and (3) optimizing for a particular microarchitecure (leveraging instruction > level paralellism for a fixed CPU model). The difference is that my (2) only requires programming skills, while (1) would require familiarity with the application domain. A trivial example: say you have a vector of 1024 floating point values and you need to compute log10 of all of them. In a general purpose routine you have no choice but to test each value x for x > 0.0, and that is something we like to avoid inside a loop or in vector code. But if you know the application domain, you may know that all values will be > -0.001, and that adding 0.002 to them won't affect the result in any way that matters in practice. So you can just compute log10 (x + 0.002) and remove the test. In real-world cases such changes may be much more invasive. For example, when designing a demodulator/decoder for a telecom system, you will have a 'degradation budget', say 0.1 dB. This means that your algorithm is allowed to perform as if the S/N ratio of the input signal was 0.1 dB less than it really is. You're free to spend that 0.1 dB wherever you want. But you can do this only if you understand the consequences of e.g. using a less accurate computation at some point, and are able to demonstrate (by analysis) that you remain within the budget by doing so. This requires understanding the algorithm at a much deeper level than would be required to code it given a detailed description. > We will have to submit code to a university-internal repository and it will run > through some software plagiarism system. Would that be a problem? No, but I can't reveal the actual way it will be used. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

11 years, 4 months

Audio problem for numerical optimization

by user web210p1

Hi! I'm currently taking a course in University called "How to write fast numerical code". We are to do a group project of 3 persons where we choose a numerical algorithm and optimize it over the course for a particular processor (e.g. apply vectorization, cache optimization etc...). I was wondering if there is an interesting algorithm in audio processing that we could apply. In particular, if there is any library developer out here that would be interested in an optimization of one of the algorithms used, we would be happy to contribute. Ideally, the problem is compute intensive (i.e. bottleneck of algorithm is not the memory hierarchy). FFT is used as an example in the lecture so it cannot be used, unless there is a significant additional computation involved. Thanks very much for suggestions, Jeremia

11 years, 4 months

JACK on Gigaport HD+ at 4800kHz (reduced channel count)

by Jörn Nettingsmeier

Hi *! I'm trying to get my Gigaport HD+ to run at 48000kHz. The specs say it is capable of 8ch @ 44k1/16, 6ch at 44k1/24 and 48k/24. When I try to start it at 48k, it comes up ok but ends up running at 44k1. It shows 8 channels, so that's expected. Now, how do I tell it to use only 6, so that I can get to 48k? I've tried setting -o6, which gives the usual "cannot set playback channel count". Next I looked at amixer -D hw:3 controls, and there is a playback switch map that is set to "on,on,on,on,on,on,on,on". I set it to "on,on,on,on,on,on,off,off" and retry. No luck. Still fails to set playback channel count, JACK still comes up with 8 outs and falls back to 44k1. What am I missing? Is there some .asoundrc magic that might come to the rescue? Maybe define a device that has only 6 channels to begin with? But how do I do that? Any hints much appreciated. Best, Jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net

11 years, 4 months

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Linux-audio-dev