Linux-audio-dev

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4 participants
7502 discussions

by Juhana Sadeharju

I have now written a vocal reverb. It needs a pluginization. An example would do fine but the example must be exactly what I need, otherwise it may just confuse me. Latest plugin stuff, of course. Unlike GVerb, the new reverb has a preset structure. The reverb is initialized this way: effect = effect_new(config_ptr,preset_ptr); The preset structure can be modified during the processing by switching the pointer or by modifying the parameters in the structure. The audio is processed by calling effect_do(effect, input_array, count, outleft_array, outright_array); I'm also glueless with the following: 1. Denormalization. Feeding noise at the reverb input would save me trouble of finding all places where the problem occurs, but is this wise? How to generate the proper noise? 2. Effect modulation is done by two extra arrays at effect_do( ..., mod1_array, mod2_array, ...) They should be filled with slowly varying sin() signals, but how to do it efficiently? I have used earlier the sin() call. 3. The audio is divided to blocks of N samples. Apparently I should interpolate the tweaked parameters over the N samples inside the reverb, or I should provide arrays for all parameters and leave the interpolation to the plugin wrapper. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

17 years, 10 months

new lossless/lossy audio compressor

by Gregory Alan Hildstrom

Hello. I just finished writing and documenting a new lossless and lossy audio codec called j2kaudio. It uses the wavelet compression in JPEG 2000 to transcode audio wav data to and from j2a files. I have not written any sort of media player plugin yet; more testing and file format refinement are necessary first. I tested with 44100Hz, 16-bit, stereo CD-quality wav files, but it also supports other sample rates, precisions, and channel numbers. I tested against flac, lpac, shorten, and monkey's audio for lossless encoding. j2kaudio beat the nearest competition by over 10%. If anyone is motivated to test some additional songs, provide me with additional test data, or suggest another untested codec, please let me know. http://geocities.com/hildstrom/projects/j2kaudio/index.html Let me know what you think. Thanks. -Greg Gregory Alan Hildstrom Secure Systems Engineer - Trusted Computer Solutions trustedcs.com ghildstrom(a)trustedcs.com Software Engineer - Hildstrom Engineering geocities.com/hildstrom hildstrom(a)yahoo.com Mobile:(210)413-6082 Fax:(210)599-0469 Office:(210)340-3151x117

17 years, 10 months

PHASEX-0.11.1

by William Weston

PHASEX-0.11.1 contains fixes for the segfault issues some users have seen with version 0.11.0. Special thanks goes to Adam Sampson for tracking this down, and to the rest of you who sent in bug reports. Upgrading to 0.11.1 is recommended for all users, since it appears that this bug will corrupt memory used by the synth engine even if it doesn't trigger a segfault. As usual, source tarballs and FC6 RPMs (src, i386, i686, and athlon) are available at: http://sysex.net/phasex Thank you for your continued support. Here's to happy music-making under Linux! Cheers, --ww -- /* William Weston <weston(a)sysex.net> */

17 years, 10 months

PHASEX-0.11.0 released

by William Weston

Version 0.11.0 of the [P]hase [H]armonic [A]dvanced [S]ynthesis [EX]periment is ready for download (source tarball and FC6 RPMs) at: http://sysex.net/phasex/ This release contains many fixes and improvements and is highly recommended for all PHASEX users. Here's a brief list (see the ChangeLog for more details): Engine fixes: Filters and oscillator bandlimiting are now properly tuned (taking into account filter oversampling) and work as documented. Calculations for lookup tables are more accurate. The envelope curve has been improved and sounds a lot more natural. Additional bugfixes are found in the ALSA MIDI startup and JACK shutdown code. Engine optimizations: A few simple optimizations have been worked out, with an average of about 30% CPU cycle reduction per voice, and about 30MB of memory overall. For custom builds, the engine can easily be fine-tuned in phasex.h, and the --enable-arch= configure option supports all x86 CPU types supported by gcc-4.1.2. Improved patch bank: The entire bank is preloaded into memory. There is now support for patches in alternate directories. The memory modes have been reworked (and finally documented in the runtime help system). Multiple patches from a directory may now be loaded into the patch bank at once. GUI updates: Special thanks goes out to Pete Shorthose for providing the initial backing store code and some pearls of GTK wisdom to make window redrawing and tab switching faster and smoother. For those of you who haven't done so already, now would be a good time to enable backing store in your X server. Knob and control updates: Scroll wheel support has been added to the knobs and alternate controls. Middle-click centering of knobs is more responsive. The cumbersome combobox lists have been changed to detent knobs with clickable value labels. Parameter locking: Parameters can now be locked down to their current values, effectively blocking out program changes, patch loads, and MIDI controller messages on a per-parameter basis. New settings: The base A4 tuning frequency and backing store can now be set in the preferences dialog. Cheers! --ww -- /* William Weston <weston(a)sysex.net> */

17 years, 10 months

ANN: Quicktoots 17 - this one with video...

by Patrick Shirkey

Hi, Announcing the latest release to the quicktoots archive. DJing with Freewheeling & Hydrogen. DJing is an artform and freewheeling with JACK on Linux makes it possible to amaze your audience with live remixes of rare cuts and hidden gems that when put together produce sample Heaven... Ringheims Auto shows us how it's done. http://quicktoots.linuxaudio.org As always we would love to have more submissions on new ways to use the latest tools. If you are interested in having your name up in lights let me know. ----------------------- Background: The purpose of the Quicktoots are to provide a community resource of informative guides for using Linux audio applications. The Quicktoots are the brainchild of Dave Phillips the man responsible for the most comprehensive webpage devoted to Linux audio applications. The Linux Sound and Midi page. This has now been taken over by the community and moved to the Linux-Sound Application Index your one stop resource for Linux sound software. http://apps.linuxaudio.org/ If you are in the position to link to any of the quicktoots please link to the main page. That way we can keep more accurate statistics on the amount of people viewing the pages. If you are interested in contributing to the Quicktoots please let us know. All work must be submitted me and we I will give advice or make changes to ensure our publishing standard is met. If you are not able to send html don't worry because I am willing to format any braindumps. Enjoy. ----------------------- Cheers. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com Http://lau.linuxaudio.org - The Linux Audio Users guide ======================================== "Anything your mind can see you can manifest physically, then it will become reality" - Macka B

17 years, 10 months

[ANN] Mammut V0.60 and Snd-ls V0.9.8.1_beta

by Kjetil Svalastog Matheussen

Mammut 0.60 =========== Mammut will FFT your sound in one single gigantic analysis (no windows). These spectral data, where the development in time is incorporated in mysterious ways, may then be transformed by different algorithms prior to resynthesis. An interesting aspect of Mammut is its completely non-intuitive sound transformation approach. *Homepage: http://www.notam02.no/arkiv/doc/mammut/ *Screenshot: http://www.notam02.no/arkiv/doc/mammut/mammut.png Changes 0.59 -> 0.60 -Updated source to work with Juce 1.44. (later or earlier versions of juce might not work with this version of mammut) -Fixed a couple of ugly bugs in the progress bar code. -Set moving camera to be off by default. Moving camera seems to stall mammut when using newer juce's. :-( Snd-ls v0.9.8.1_beta ==================== 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. (Its named beta, because it hasn't been tested that much. But I think it should work very well. However, a new non-beta version is going to be released quite soon with some additional fixes.) Changes 0.9.7.12 -> 0.9.8.1_beta -Fixed bug that caused snd to fail starting if no previously used soundfile was opened during startup. -Updated Snd from 8.4/12.9.2006 to 9.3/30.7.2007. Many important fixes. -Another attempt at fixing the rt_readin_tag startup bug. Download from http://www.notam02.no/arkiv/src/snd/

17 years, 10 months

non-blocking intra-process messaging

by Stephen Sinclair

Hi, I have some multi-threaded audio software that passes data around using pointers to data structures. However, I'd like a more generalized message passing system. The only requirements are that it cannot block the sender or receiver (they are considered soft 'real-time'), and also that multiple senders should be able to send a message to the same receiver at the same time (which makes non-blocking circular buffers a difficult approach). I'm thinking of designing something that uses atomic_ops to implement quick locking of portions of a message queue. Anyways, my idea is to use something like OSC for message passing, but without the "path" part of the OSC address. (That is, the path is already resolved, since threads share memory..) Does anyone know if something like this has been done previously? Ideally it would easily scale to inter-process communication, if in the future I wanted to turn my threads into processes. (Another reason for basing it on OSC..) Does any of this make sense? Basically I've been running into a lot of shared-memory threading problems, which in retrospect should have been expected and avoidable if I'd spent more time designing my message passing system. (At the time I hadn't heard of the atomic_ops library.) I might have designed the whole system to use timers instead of threads and avoided a lot of headaches, but I wanted the threads to be splittable across CPU cores. (Which arguably is a good reason to use processes.. admittedly the whole thing is starting to point me towards the "threads are a bad idea" camp.) Cheers, Steve

17 years, 10 months

Goodbye ReplyTo munging...

by Marc-Olivier Barre

Dear list members, ReplyTo munging is now deactivated on Linux-audio-dev and Linux-audio-user. We now have a RFC compliant list. To reply to the list, you may use the "reply to all" button of your email client, or the "reply to list" button if your email client has one. Best regards, __________________ Marc-Olivier Barre.

17 years, 11 months

intergrating osc

by Patrick Shirkey

Hi, Any pointers on where to start for integrating osc compatibility into jackEQ? Cheers. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com Http://lau.linuxaudio.org - The Linux Audio Users guide ======================================== "Anything your mind can see you can manifest physically, then it will become reality" - Macka B

17 years, 11 months

rtaudio jack/alsa problem..

by James Stone

I have a problem with a a project I am helping out with (aldrin), which uses rtaudio version 3.0.3 for jack/alsa/oss handling. When aldrin starts, it calls libzzub to detect all possible sound input/output, and then selects the best one (defaulting to jack if it is running). This is working fine on my system if I have selected a sub-device from my SBLive card as the sound input/output device (hw:0,3 works fine). However, if I select the default jack output (hw:0), I get loads of xruns when aldrin starts, then jack dies. Doing some further testing, if I disable detection of alsa after libzzub successfully finds a jack device, then hw:0 will work fine without any xruns, so it seems that there is some conflict between jack and alsa. Lastly, I noted that if I select hw:0,3 as the output for jack, then libzzub/rtaudio does not pick up hw:0,3 as a valid alsa output (presumably correctly assigning it as a jack output), whereas if I am using hw:0 as the output for jack, then hw:0,0 is also detected as a valid ALSA device.. this makes me think the xruns are caused by some conflict. The lead dev for libzzub is not happy with my patch to libzzub not to detect alsa if jack is found, and would prefer a fix to the rtaudio code. I am not really sure where to start with this, since I am not 100% sure what is going on. I would really appreciate it if anyone could help with this.. Best wishes, James

17 years, 11 months

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Linux-audio-dev