Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7516 discussions

Re: [LAD] basic RT-thread model questions

by Kjetil S. Matheussen

"Patrick Stinson": > > 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? Yes, that is OK. But beware that a RT thread running with priority x waiting for a RT thread running with priority y, will in practice only run with priority min(x,y). This is called priority inversion, there's probably a wikipedia article about it. In short: If all threads run with the exact same priority, then it's OK. > 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 don't understand the question ("i have gathered are (...) a=>b message-passing (...) while (... useless, since (...), given (...) is negligable. (...)" ???), but it seems like others have, so hopefully their answers are correct.

17 years, 5 months

[ANN] QjackCtl 0.3.3 (unstable-qt4) released!

by Rui Nuno Capela

Howdy! I am pleased to announce this new and long due release of QjackCtl, the Qt GUI for the awesome JACK Audio Connection Kit. Release highlights are mainly about final JACK-MIDI support for the "evil" Patchbay, new Messages file logging and the most intriguing application window instance uniqueness which will make X11 desktop life easier for everyone (ie. no more duplicates as JACK server gets auto-started as candy bonus:) You can grab it from the project source as usual: http://qjackctl.sourceforge.net http://sourceforge.net/projects/qjackctl In case you need an upstream hug, you're free to visit my own forum: http://www.rncbc.org The change-log doesn't say much but... here it goes: - Attempt to load Qt's own translation support and get rid of the ever warning startup message, unless built in debug mode (transaction by Guido Scholz, while on qsynth-devel, thanks). - Messages file logging makes its first long overdue appearance, with user configurable settings in Setup/Options/Logging. - Only one application instance is now allowed to be up and running, with immediate but graceful termination upon startup iif an already running instance is detected, which will see its main widget shown up and the server started automatically (Qt/X11 platform only). - Finally, full JACK MIDI support sneaks into the patchbay; socket types now differ in Audio, MIDI and ALSA, following the very same nomenclature as found on the Connections widget tabs. - Sun driver support (by Jacob Meuser). - Delay window positioning at startup option is now being disabled, on the Setup/Misc tab, when Minimize to system tray is enabled. - Cosmetic fix: Setup/Settings tab, 'Input Device' text label was using a slightly smaller font than the rest of the application (bug#1872545, reported by Jouni Rinne). Cheers && Enjoy -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

17 years, 5 months

basic RT-thread model questions

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!

17 years, 5 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

17 years, 5 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

17 years, 5 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

17 years, 5 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

17 years, 5 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

17 years, 5 months

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

by Stefano D'Angelo

17 years, 5 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

17 years, 5 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev