Linux-audio-dev

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7513 discussions

[linux-audio-dev] [ANN] QjackCtl 0.2.21 is out!

by Rui Nuno Capela

Greetings, In the wake of the JACK 0.102.20 release, here goes the respective friendly GUI: QjackCtl 0.2.21 is out. As usual, the change-log says it all: - GPL address update. - All window captions can now be set smaller as tool-widgets. This option takes effect when child windows are kept always on top. - For the brave of heart, specially the ones brave enough to try with Stephane Letz's jackdmp, a win32 build should be now possible. - The main window button text labels are now optional (after a kind suggestion by Geoff Beasley, thanks). - Increse default maximum number of ports setting from 128 to 256. - Initial freebob backend driver support. Also changed the coreaudio backend driver command line device name/id parameter (EXPERIMENTAL). - Closing the main window while not as an active JACK client, nor under a server running state, will just quit the whole application, even though the system-tray icon option is in effect. - The most relevant transport commands (Rewind, Play and Pause) are now made available on the main window context popup menu. - The post-shutdown script is now also being called when using the Stop button, whether the jackd server has been started internally or not. The initial hard-coded default is now on and set to `killall jackd` (as a workaround to an old request from Stephane Letz). - The main window buttons display are now optional. One can choose whether the left, right and/or transport buttons are hidden, making it for a total of six different modes for the main window presentation (after a much simpler suggestion from Paul Davis and Stephane Letz). - Added configure support for x86_64 libraries (UNTESTED). Nuff this time. Hope you enjoy. -- rncbc aka Rui Nuno Capela

18 years, 11 months

[linux-audio-dev] Re: Paper on dynamic range compression

by Ben Loftis

Andreas, I want to thank you for bringing this up. This is very important work. Interest in open-source audio is going to grow DRAMATICALLY within the pro audio industry in the next year or so, and it will be quickly squelched if basic principles ( like smooth gain adjustments given a wide range of inputs ) are ignored. Personally I find many of the plugins, particularly the filters and compressor/gates, to be nearly unusable and certainly not in the same class as commercial offerings. Why should this be true? Commercial developers have the advantages of full-time development, years of focused experience, and expensive test equipment. Open-source developers have the advantage of peer review. I hope that your study is just the first in a series of highly critical looks at the open-source offerings, and that it results in a definition and refinement of the "best practices" that are already known by professional developers. For our part, Harrison has taken an interest in Ardour and has actively guided development towards professional needs for the last 6 months or so. You can expect to see more industry involvement as the benefits of "Free sofware" become obvious to the pro audio world. 10 years of proprietary development has resulted in the stagnation of our industry. For example, there are certain limitations: bit depth, file compatibility, and inter-app communication, that cannot be lifted given the current audio software business environment. These limitations are NOT imposed by Free software like Ardour. Open Source and/or Free software is the only avenue open for true innovation and improvement in sound quality. Please continue to look critically at the available plugins, and continue to raise these issues before there is an influx of professional interest. Open-source plugins may never sound as good as the best commercial offerings ( *cough* Harrison *cough* ), but they shouldn't be allowed to sound BAD, given the wealth of knowledge available in the LAD group. -Ben Loftis www.harrisonconsoles.com

18 years, 11 months

[linux-audio-dev] Gaussian Mixture Models

by Yosvany Llerena Rodríguez

Hi all. I'm student of Telecommunication Systems and my ending homework is related about GMM's - EM algorith for classification. My searchs in IEEE I found a few papers but this paper are just a resume of the technique and I need a Detailed information to develop the full algorithm in C/C++... All your help about any document, or sourceode will be grateful... Thank's in advanced. Yosvany

18 years, 11 months

[linux-audio-dev] avoiding signal invertion

by Andrew Gaydenko

(Sorry for crossposting, it seems like Alsa-user list hasn't an answer) I use ICE1724 ALSA driver for Terratec Aureon 7.1 Space card. I have noticed both default (line out) and plug:spdif devices'es outputs are inverted. Is there a way to add additional inversion and, as a result, to get "normal" output signal? Andrew

18 years, 11 months

[linux-audio-dev] [ANN] Aqualung 0.9beta6 released

by Tom Szilagyi

The developers are pleased to announce the latest release of Aqualung, a music player for GNU/Linux. Website: http://aqualung.sf.net Without further ado, the ChangeLog is attached below. Enjoy, Tom 2006-10-03 Tom Szilagyi <tszilagyi at users dot sourceforge dot net> Aqualung 0.9beta6 http://aqualung.sf.net This release introduces a fair number of substantial improvements: * Music Store builder: automatically build a Music Store by scanning the files on disk. Perform CDDB lookups & extract metadata on the fly. * MPEG decoder enhancements: robust file recognition, VBR and UBR file support, frame-accurate seeking, true gapless playback via eliminating encoder padding+delay read from LAME headers. * Fully revamped metadata support using TagLib. The result is a more complete implementation also supporting APE tags in Musepack files. * Automatic output driver detection: ability to startup without command line arguments (using default driver parameters). * Systray (a.k.a. Notification Area) support. * Handling of compressed MOD files (.gz and .bz2). * Resolved issue with JACK memory locking (which previously resulted in runaway memory consumption when running with realtime JACK output). * Aqualung compiles & runs under FreeBSD and Cygwin. NEW LIBRARY DEPENDENCIES: * TagLib >= 1.4 is now required for metadata support. http://developer.kde.org/~wheeler/taglib.html * GTK+ >= 2.10 is needed for the (optional) Systray support. DROPPED DEPENDENCIES: * libid3tag library is not required anymore (succeeded by TagLib).

18 years, 11 months

[linux-audio-dev] VST API license

by Ivica Ico Bukvic

Hi all, I am in the process of considering to, on behalf of Linuxaudio.org, negotiate with Steinberg possible change of VST API license in order to allow its easier adoption within Linux LGPL/GPL environment. As Daniel has already pointed out, with Yamaha now owning Steinberg, we ought to have a better chance at making this a reality. For this reason, I was wondering if anyone else is currently working on this and if so what is its status. I also recall quite a few years ago Paul trying to compile a letter to Steinberg. Whatever happened to that? 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

18 years, 11 months

[linux-audio-dev] [ANN] San Dysth V0.1.0, Snd-ls V0.9.7.1, E-Radium V0.61f

by Kjetil S. Matheussen

1. === San Dysth is a standalone realtime soft-synth written in SND. It was first developed as final project for the 220c course at CCRMA. For information about the used synthesizing routine and sound examples, check out San Dysths homepage at: http://www.notam02.no/~kjetism/sandysth/ Requirements ------------ Snd-ls >=0.9.7.1 G2Reverb ladspa plugin. 2. === 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. Download from http://www.notam02.no/arkiv/src/snd/ Changes 0.9.7.0 -> 0.9.7.1 --------------------------- -Updated Snd from 8.4/13.9 to 8.4/26.9. Many important fixes. -Added workaround for menu problem. Bug found by Dragan Noveski. -Added check that initialization is complete. -Some realtime fixes. 3. === E-radium is Radium and a special version of E-UAE (with support for realtime scheduling and alsa midi). Radium is a unique type of music event editor made to be efficient and provide many possibilities. The user interface is inspired by trackers, but Radium is more versatile and can be used for all kinds of genres. http://www.notam02.no/radium/ Changes 0.61e -> 0.61f ---------------------- -Fixed all build errors in 0.61e (0.61e should never have been released.) -Made sure everyone can run e-radium after installation, not only root and the user doing the compilation.

18 years, 11 months

[linux-audio-dev] right way to announce an app update

by Andrew Gaydenko

Hi! What is the most acceptable way to announce an app update? I use "[ANN] ..." message sending to LAD/LAU lists, but am not sure it is the best way for LA community. (the app under question is QLoud) Andrew

18 years, 11 months

[linux-audio-dev] [ANN] QLoud v.0.18 - Step Response plotting is added

by Andrew Gaydenko

QLoud is a tool to measure loudspeaker frequency and step responses and distortions. Find it here: http://gaydenko.com/qloud/ Changes: - Step Response plotting is added, this is a direct screenshot link: http://gaydenko.com/qloud/screenshots/shot04.png These measurements are done for the same 2-way loudspeaker (see plots titles). It is clear, a tweeter must be shifted lightly beyond from a listener.

18 years, 11 months

[linux-audio-dev] How to test resampling quality?

by James Courtier-Dutton

Hi, I was wondering if there are any tools out there to test audio resampling quality. I am particularly interested in 44.1kHz to 48kHz resampling due to the fact that most sound cards prefer 48kHz. At least with up sampling (low rate to higher rate) one does not get aliasing. I really just want to find some algorithm that I can use to compare 44.1kHz audio signal with an 48kHz audio signal, and to see if there has been any lose of quality during the up sample. James

18 years, 11 months

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Linux-audio-dev