Linux-audio-dev June 2008

linux-audio-dev@lists.linuxaudio.org

51 participants
30 discussions

by Patrick Stinson

I'm just asking these out of curiosity. I know you guys love these kinds of questions :) I know that the rule is to never block in a real time audio thread, but what about blocking for resources shared between real-time threads? In the case of a sequencing app that uses one thread per audio/instrument track (as most do), is it OK to share a lock between real time scheduled tracks? I ran into this question after implementing a scripting engine for our commercial audio plugin using python, which uses a global lock to serialize access to it's data structures. Also, I've gathered are that adding app-thread => RT-thread message-passing to avoid using locks while modify the application's basic shared data structures is useless, since the real-time thread will have to wait for execution of that code one way or another, given the overhead of acquiring the lock is negligable. This would mean that you should just use locks and assume the user will cope with hearing small overruns when adding/removing audio components. True? Not true? I hope I worded these well enough. Cheers!

16 years, 9 months

Let's kill EPAMP???

by Stefano D'Angelo

Let's stop this flame for a moment and see what LV2 misses in order to let me kill EPAMP and live an happier life. #1. Support for interleaved channels and non-float data Input and output data is often found in these formats. #2. Changing sample rate without re-instantiating all effects. Gapless playback when chaning songs, for example, should be possible without performing black magic. #3. Some serious connection logic thing (all the "equal channels" thing etc.). This needs a thousand flame wars and *deep* thinking. #4. Support for time stretching when using non real-time audio sources. #5. Informations about delay time introduced by the algorithm itself to do syncing with video-sources (for example). #6. Some way for the host to make sense of the meaning of some parameters and channels, to better support global settings and stuff. #7. Global explicit initialization/finalization functions for more exotic platforms (they wouldn't harm, so why not having them). #8. Rules to find plugins possibly platform-specific and outside of the specification; possibly one compile-time valid path. #9. Maybe more strict requirements on both hosts and plugins (especially about thread-safety). I see there is some indication in the core spec, but I don't know about extensions and/or other possible concurrency issues. #10. Something (a library possibly) to make use all of this features easily from the host author's POV. Can we start discussing about these issues and see if they are solved already/how to implement them/how to make them better? Stefano

16 years, 10 months

[ANNOUNCE] AudioScience driver 3.10.00

by Eliot Blennerhassett

Greetings AudioScience linux customers and others, This is to inform you that hpklinux version 3.10.00 is available from our website http://audioscience.com/internet/download/linux_drivers.htm The major change in this release is the addition of ASI89xx tuner series, and removal of ASI4xxx (still supported by driver 3.08). Of course bugs have been fixed, and new minor features added, see the release notes for details. While I have your attention, if you are a user of our cards and have a few moments please reply and let me know which distro(s) and kernel version(s) you currently support, and whether you are using or intend to use HPI or ALSA thanks and regards -- -- Eliot Blennerhassett www.audioscience.com

16 years, 10 months

[EPAMP] an effect plugin API for media players: anyone interested?

by Stefano D'Angelo

Hello, First of all, I'm sorry for the worst cross-posting ever. My name's Stefano D'Angelo and I'm the author of NASPRO (http://naspro.atheme.org), a project whose main aim is to build up a system to make all sound processing APIs interoperable. Since my project was recently accepted to become part of the Atheme community (http://www.atheme.org) and began working close to the audacious (http://audacious-media-player.org) developers, I could see there actually were serious reasons behind the lack of standardization for sound processing in the arena of media players. Now, I wrote a first draft of a new sound processing API called EPAMP especially targetted at media players; you can find it here: http://naspro.atheme.org/content/epamp-draft-1 With this mail, I'm trying to get some feedback (possibly in the Talk page on the NASPRO wiki) about it and possibly to involve anyone who is interested in its development. Anyway, notice that the user interface part of the API is still missing and that my project (NASPRO) will support this new API, giving applications using it the possibility to access non-EPAMP plugins without touching a single line of code (currently supported APIs are audacious' and LADSPA, while DSSI, LV2, GStreamer and possibly VST are more or less planned for the near future). In case you're not interested and want to start a flame, please just ignore me. This message is being sent to the communities/developers behind the following projects: - Amarok - Aqualung - Audio Overload - BMPx - Banshee - cmus - Decibel Audio Player - Exaile - FALF Player - Freevo - GStreamer - Helix Player - Herrie - JaJuk - JuK (KDE multimedia) - Linux Audio Developers - KPlayer - Kaffeine - lamip - The LAMP - MPlayer - Miro - mpg123 - mpg321 - Muine - MPD - music on console - Noatun - ogg123 (vorbis-tools) - Ogle - Quod Libet - Rhythmbox - Sipie - SnackAmp - Sonata - Songbird - UADE - SMPlayer - VLC - wxMusik - XMMS2 - Xfmedia - Xine - Zinf If you know someone else who could be interested, feel free to forward this. Best regards, Stefano D'Angelo zanga.mail(a)gmail.com

16 years, 10 months

alsa sequencer and sysex editing

by rdxesy＠yahoo.de

hello ") Ok, I am writing a remote and SysEx-editor for e-mu synthesizers (http://ppcontrol.sf.net). I am asking for help on how to use the alsa sequencer the way it was intended to be used for this kind of application. You know, there are so many options and flags and stuff to be used and I feel a little lost. Basically I send a lot of SysEx messages and need to make sure to catch all the answers. My current implementation is as following: I have a single sequencer (duplex, nonblocking), a queue and two ports: one with read and one with write caps. To send a SysEx command I just queue an event with that message. (So far so good?). Now I just wait for an constant time (usleep) before I check the input queue. Problem: if the time is too short, i would have missed the answer, if I wait to long I waste time. My question is: how would I make sure to catch all answers from the device without using a thread and without just waiting for an constant time before I check if there is an answer( which is alo not very fail save, a message might take a little longer if there is traffic on the midibus)? Thank you. Jan

16 years, 10 months

LADSPA latency port, was Re: Let's kill EPAMP???

by Chris Cannam

On 03/06/2008, Steve Harris <steve(a)plugin.org.uk> wrote: > In LADSPA there's a "magic" control out port called "_latency" or > something, that should apply to LV2 aswell, but I'm not sure if the > spec says so. For the record -- since this is something I've tried to search for in the past and have had trouble finding a definitive answer to -- the conventional LADSPA port name is apparently "latency", with no underscore. Some hosts (such as Rosegarden) will accept either "latency" or "_latency"; in RG's case that's because I wasn't sure which was supposed to be correct when I coded it. But others (such as Ardour) will only accept "latency", as I discovered when I released a plugin that used "_latency" and forgot to test it in Ardour first. Chris

16 years, 10 months

[EPAMP] an effect plugin API for media players: anyone interested?

by Stefano D'Angelo

16 years, 10 months

Re: [LAD] [EPAMP] an effect plugin API for media players: anyone interested?

by Sampo Savolainen

Quoting Stefano D'Angelo <zanga.mail(a)gmail.com>: > It would work like this: > > stereo input -> hosts converts to 5.1 (the effect states it needs 6 > channels) -> apply effect with 6 in channels and 6 out channels -> 5.1 > output How would the host convert the stereo input to 5.1? In my view, that is the whole purpose of the effect. Sampo

16 years, 10 months

LAC on German radio

by Frank Barknecht

Radio Recommendation For those interested in Linux Audio, Pd, SuperCollider, .... It's in German, though. Streaming info: http://www.dradio.de/streaming/ The HQ-Ogg-stream is the best: http://www.dradio.de/streaming/dkultur_hq_ogg.m3u Deutschlandradio Kultur - Neue Musik - 03.06.2008 - 00:05 CEST(!) Musik aus dem digitalen Baukasten Die Linux Audio Conference 2008 in Köln Von Hubert Steins Computer sind erschwingliche, universelle Musikmaschinen. Doch die vermeintliche Freiheit trügt, denn Softwaregiganten wie Microsoft und Apple dominieren das Innenleben auch der Musikrechner. Künstlerische Autonomie erfordert aber Kontrolle über die Produktionsmittel, sagen viele Musiker und Medienkünstler, die deshalb das quelloffene Linux-Betriebssystem einsetzen. Einmal im Jahr treffen sich die Linuxmusiker zur Linux Audio Conference, zuletzt im Frühjahr 2008 an der Kunsthochschule für Medien in Köln. Die Szene ist bunt: Die Präferenz für Linux als Betriebssystem ist der gemeinsame Nenner, der hier Komponisten akademischer Neuer Musik mit Technomusikern oder den Programmierern von Aufnahmesoftware an einen Tisch, auf eine Bühne bringt. Für Deutschlandradio Kultur hat Hubert Steins die Protagonisten der Szene wie den Musiker und Mathematiker Miller Puckette getroffen http://www.dradio.de/dkultur/programmtipp/vorschau/793304/ Ciao -- Frank Barknecht _ ______footils.org__

16 years, 10 months

lo_server_thread_new pause

by Dave Griffiths

Hi all, I think this is a network setup user error rather than anything complicated, but anyone know why I get long pauses (5-10secs) when calling lo_server_thread_new for the first time in a process? Is it likely to be some timeout I can adjust? cheers, dave

16 years, 10 months

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Linux-audio-dev June 2008