Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

4 participants
7441 discussions

[linux-audio-dev] Letter to kernel developers?

by Lee Revell

Can someone please help me find a link to the open letter to kernel developers regarding horrible latency with kernel 2.6 that Paul Davis, Benno Sennoner and others posted to LKML in early 2004 (NOT the one from 2000) - the one that led Ingo to develop the voluntary preemption patches. I cannot find it with any amount of googling, and my local LKML archive only starts in July 2004. There was such a letter right? Am I crazy or just making this up? Lee

18 years, 6 months

[linux-audio-dev] liboggz 0.9.5 Release

by Conrad Parker

Oggz 0.9.5 Release ------------------ Oggz comprises liboggz and the command-line tools oggzinfo, oggzdump, oggzdiff, oggzmerge, oggzrip, oggz-scan and oggz-validate. liboggz is a C library providing a simple programming interface for reading and writing Ogg files and streams. Ogg is an interleaving data container developed by Monty at Xiph.Org, originally to support the Ogg Vorbis audio format. This release is available as a source tarball at: http://www.annodex.net/software/liboggz/download/liboggz-0.9.5.tar.gz New in this release: * Fixed and updated Windows (Visual Studio) support - added missing exported symbols, projects for oggz tools. (Alex Krumm-Heller, Silvia Pfeiffer) * Support for OggPCM (Draft 2, Main header) OggPCM is an experimental specification for storing uncompressed PCM audio in Ogg bitstreams. - liboggz: Recognition of OggPCM timestamps, and support for seeking in files that contain OggPCM logical bitstreams. - oggzinfo: Display OggPCM header details - oggzdump, oggzrip: New [--content-type pcm, -c pcm] option to filter on OggPCM - oggz-validate: Validate framing of OggPCM logical bitstreams This version is installed on http://validator.annodex.org/ for online validation of OggPCM files. For more information about OggPCM, see: http://wiki.xiph.org/index.php/OggPCM * ./configure support for large (>2GB) files This version adds build configuration support for large files, allowing liboggz to operate on files >2GB. This version does not introduce any API changes; interfaces such as oggz_tell() continue to use off_t externally. However, sequential reading and validation of large files is now possible. * bug fixes and cleanups: - oggz-validate, oggzmerge, oggzdump, oggz-scan, oggzinfo: handle unknown content types (Ian Malone) - remove deprecated oggzed example - various code and documentation build cleanups About Oggz ---------- Oggz comprises liboggz and the command-line tools oggzinfo, oggzdump, oggzdiff, oggzmerge, oggzrip, oggz-scan and oggz-validate. liboggz supports the flexibility afforded by the Ogg file format while presenting the following API niceties: * Full API documentation * Comprehensive test suite of read, write and seeking behavior. The entire test suite can be run under valgrind if available. * Developed and tested on GNU/Linux, Darwin/MacOSX, Win32 and Symbian OS. May work on other Unix-like systems via GNU autoconf. For Win32: nmake Makefiles, Visual Studio .NET 2003 solution files and Visual C++ 6.0 workspace files are provided in the source distribution. * Strict adherence to the formatting requirements of Ogg bitstreams, to ensure that only valid bitstreams are generated; writes can fail if you try to write illegally structured packets. * A simple, callback based open/read/close or open/write/close interface to raw Ogg files. * Writing automatically interleaves with packet queuing, and provides callback based notification when this queue is empty * A customisable seeking abstraction for seeking on multitrack Ogg data. Seeking works easily and reliably on multitrack and multi-codec streams, and can transparently parse Theora, Speex, Vorbis, FLAC, CMML and Ogg Skeleton headers without requiring linking to those libraries. This allows efficient use on servers and other devices that need to parse and seek within Ogg files, but do not need to do a full media decode. Full documentation of the liboggz API, customization and installation, and mux and demux examples can be read online at: http://www.annodex.net/software/liboggz/html/ Tools ----- The Oggz source tarball also contains the following command-line tools, which are useful for debugging and testing Ogg bitstreams: * oggzinfo: Display information about one or more Ogg files and their bitstreams. * oggzdump: Hexdump packets of an Ogg file, or revert an Ogg file from such a hexdump. * oggzdiff: Hexdump the packets of two Ogg files and output differences. * oggzmerge: Merge Ogg files together, interleaving pages in order of presentation time. * oggzrip: Extract one or more logical bitstreams from an Ogg file. * oggz-scan: Scan an Ogg file and output characteristic landmarks. * oggz-validate: Validate the Ogg framing of one or more files. License ------- Oggz is Free Software, available under a BSD style license. More information is available online at the Oggz homepage: http://www.annodex.net/software/liboggz/ enjoy :) -- Conrad Parker Senior Software Engineer, Continuous Media Web, CSIRO Australia http://www.annodex.net/ http://www.ict.csiro.au/cmweb/

18 years, 6 months

[linux-audio-dev] Old Specimen mockups

by Thorsten Wilms

Hi! Specimen might be dead or in keep-functioning-maintainance only, and work based on these mockups had been canceled before due to them being too time consuming to implement and doubts on extendability, anyway :( BUT, instead of them just bit-rotting on my hd, I rather present them here, since they could still be an inspiration to someone :) 1. Keys and Main: http://xs71.xs.to/pics/06102/specimen_18_keys_main_i.png 2. Mixer and Pitch: http://xs71.xs.to/pics/06102/specimen_18_mixer_pitch_i.png Like in the actual Specimen, all sliders are fan-sliders. (http://leute.uni-wuppertal.de/~ka0394/en/fan-sliders/index.html) On the left there are 16 MIDI channels down. The numbers also act as MIDI activity indicator (3 is receiving something). They're followed by volume slider, mono, solo and name. The selected channel (1) is connected to the Keys/Mixer area. This works a bit like tabs, but allows that section to come up from the sunken field, resulting in cleaner look. It's all meant to be GTK theming friendly. On the keys area, Patches are assigned to keyboard ranges. They can be put in columns to allow layering. The base note is marked (D here, could have made that more obvious). The selected Patch is shown in inverted colour. The Mixer area is a bit redundant as it is shown here. Initialy it was meant to not just allow access to panning over what the channels section on the left offers, but also Cut/Cut by (a more versatile, but not exactly self explaining system than exclusion groups, used for cutting off a open hi-hat on a closed one and similar things). Every selection of Parameters could be shown in that table, though. The right section shows the selected Patch. It was a goal to keep the Specimen window rather small to make using it with other apps more handable, so a number of tabs were needed. The not shown Volume, Panning, Cutoff and Resonance tabs would be much like the Pitch one. Contents of that tab should be pretty selfexplainatory, or it's no good anyway :) The 3 sections layout and the Keys area where all my own idea, but Pete Bessman's influence and decisons are all over the place. Cheers, Thorsten Wilms --- http://xs.to - Free Image Hosting

18 years, 6 months

[linux-audio-dev] Roland CG-8: another Linux product

by Artemiy Pavlov

Hey all, just noticed that Roland's CG-8 Visual Synthesizer (http://www.edirol.net/products/en/CG-8/) is yet another product based on Linux. You can get their sources at: http://www.roland.com/support/gpl/ - there in the first download you'll see ALSA and GTK sources. Now when will they release a Linux-based workstation that doesn't cost like OASYS? ;-) Artemiy.

18 years, 6 months

[linux-audio-dev] Writing PCM to left and right channel sperately?

by Tobias Scharnberg

Hello List, I'm new to audio programming and have problems finding the right solutions for the development I work on right now: I need to use the left and the right audio channel seperately. The device is an ARM based board with a AC97 soundchip. It seems to work with OSS so far and I did choose libsdnfile to open Soundfiles and get the PCM. As far as I learned the audio data is coded in one stream, one sample left, one sample right. Is there a way to separately fill the soundcard buffers with ALSA? The only solution I can think of right now is to recode the audio data to left only or right only and to use a library like esound to mix them together. Is this the right way to do such a thing? Any suggestions, pointers, hints are highly welcome. Greetings, Tobias

18 years, 6 months

[linux-audio-dev] options in capturing audio off a PlusDeck?

by Sean Dague

I got myself one of the PlusDeck units last summer after having received some documentation from the product team about their serial control protocol, so I figured that I'd have some chance at actually making a recording program for Linux for it. The impetus for all of this is a stack of 300 or so audio tapes that my fiancee has, which contain mostly south Indian music recordings, including training lessons, from her late teacher T Viswanathan. Some of those probably don't exist anywhere else at this point, so putting them to digital would be a good thing (tm). Anyway, I have 2 goals. First, the practical, get all this music to digital dumps (flac in ogg containers for metadata seems to be the best bet). I'm trying to figure out the best ways to do this on a Mandriva Linux 2.6.12 kernel (on an AMD64 3000+ processor running 32bit, with Via onboard sound.) The PlusDeck just ends up being an input line jack into the sound card. I've run some tests with ecasound both straight off the alsa device, and through ecasound to jackd (both as root to get real time). It is hard to tell if I'm doing better with jackd or not, but I'm making some long recordings and letting my musical fiancee tell me if she can hear the difference. :) Given that I've got the serial protocol sorted out, it seems like the practical approach is just a command line ripper that will start up all the right audio bits, start the tape, and stop when the tape gets to the end. Full automation is what I'm looking to create. Is ecasound a good option here? How much better will ecasound + jackd be over ecasound at realtime? Would building some kind of plusdeck control plugin for another recording program make more sense? If so, which one? Second, I'd like to turn whatever I learn during the practical phase into a reasonable open source Linux tool for using the PlusDeck. I'd be a bit unnerved if that required that I set up all jack & ecasound bits, as well as kicked them off as setuid root, but better options are failing me at the moment. Any advice, ideas, suggestions, questions... even flames would be welcomed. I've spent a bit of time learning more than I ever thought I'd end up knowing about Linux audio already, and know I have a whole lot more I need to learn still. Thanks in advance, -Sean -- __________________________________________________________________ Sean Dague Mid-Hudson Valley sean at dague dot net Linux Users Group http://dague.net http://mhvlug.org There is no silver bullet. Plus, werewolves make better neighbors than zombies, and they tend to keep the vampire population down. __________________________________________________________________

18 years, 6 months

Re: [linux-audio-dev] Writing PCM to left and right channel seperately?

by Ben Loftis

Tobias, I wrote a program that does just this some years ago for IED (www.iedaudio.com). What you are doing is not trivial. When an announcement starts on the left channel (for example) you will want to fill the right channel with zeroes. However you must fill the buffers in small blocks so that if an announcement starts on the right channel, you can start stuffing the right channel without a long delay. You might want to investigate Paul's program JACK for this. It splits the audio into separate buffers for each I/O. If you really have to have it ASAP, why not just buy an IED system? :) -Ben

18 years, 6 months

[linux-audio-dev] Reliability of SPDIF sync w/ onboard audio?

by drclaw＠dogsolitude.org

Ok, given a pair of commodity Linux PC's with cheapo onboard audio codecs capable of SPDIF input and output, assuming I sent identical input signals into both machines and didn't perform any processing in the middle, could I reasonably expect the output signals to be synchronized, or should I expect unpleasant phasing / delay effects when listening to both channels simultaneously as a result of the inherently flakey consumer-grade codecs? The above is a simplification that should address my major concern... The goal here is to use a pair of redundant, inexpensive machines for live midi-controlled (headless) playback where the physical security of the location is lax enough that using my laptop + hammerfall would be too risky, not to mention 35 channels worth of overkill. Ideally these two machines would accept some sort of SPDIF format square-wave from some low power clock logic on their inputs and play identical, pre-cached wav files on their outputs when they recieve simultaneous "go" signals from the MIDI playback desk. Any experiences or opinions are welcome... Thanks, - Mike Piantedosi

18 years, 6 months

[linux-audio-dev] jack.udp and synchronisation

by Asbjørn Sæbø

jack.udp may be used to transmit audio between two computers, each computer running their own jackd. My understanding, based upon earlier threads in the linux audio lists, is that to avoid xruns due to clock drift in this situation, the sound cards of the two computers must be synchronised in some way, e.g. using word clock. Is this (still) correct? Or has jack.udp been developed to include synchronisation? (On a side note, it seems that Rohan Drape's web pages that are linked to from the jack pages, http://www.alphalink.com.au/~rd/sw/jack.html, have disappeared.) Asbjørn Sæbø

18 years, 6 months

[linux-audio-dev] Gerard van Dongen/Gilcher 1967-2006

by derek holzer

Gerard van Dongen/Gilcher passed away on Friday the 4th of March, after a brave battle with colon cancer. Gerard worked as a composer, musician and computer programmer in Rotterdam as well as played live with many improvisational and electronic musicians in the Netherlands including Bart Maris, Tom Wouters, Joe Williamson, Henk Bakker, Peter van Bergen and Ann LaBerge. He performed solo-concerts with electronics and/or piano compositions and played for numerous dance and theatre productions, produced radio-plays for the internet, and was a technical designer and software programmer for multimedia theater. Gerard was also an active member of the Pure Data and Linux Audio Developers communities, and contributed code for the user interface of the Ardour project. Gerard kept a very positive outlook throughout his illness, and he will certainly be missed. Marie Louise Gilcher has organized a memorial service for Thursday, 9 March at the "Crooswijk" Cemetery in Rotterdam. -- derek holzer ::: http://www.umatic.nl ---Oblique Strategy # 118: "Make what's perfect more human"

18 years, 6 months

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Linux-audio-dev