Linux-audio-dev

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3 participants
7535 discussions

by Aurélien Leblond

Hello, I have been coding the porting of the AMS internal modules to LV2 plugins using the Lv2-C++-Tools. The more I'm progressing, the more I am wondering if there are any limitations involved in using this library, especially these days where a lot of new features are being added to the Lv2 Extensions. For example, I am coding a plugin that output the position of the mouse cursor as two control ports (handy to develop a kind of theremin synth). Getting the cursor position from the X Library is an expensive operation, and the new Worker Extension seems perfect for the job, but I can't figure out if somehow it is compatible or not with the Lv2-C++-Tools. The most obvious pros of using Lv2-C++-Tools is its simplicity. Porting an AMS module takes only 20 to 30 minutes with it, while doing it without the help of the library takes much more time and is way more tedious. (I am much more confortable coding in C++ than C, but the question deserves to be asked, am I doing something wrong here that I find it way more difficult to code without the help of Lv2-C++-Tools?) So I am putting the question to the community: what path you would you advise to follow? With or Without Lv2-C++-Tools?

13 years, 10 months

[LAD] [LAA] [ANNOUNCE] aj-snapshot-0.9.6 has been released

by Lieven Moors

Hi all, I would like to announce the release of aj-snapshot-0.9.6 aj-snapshot is a command line utility to store/restore ALSA and/or JACK connections to/from an XML file. The most important change in this release is that aj-snapshot behaves differently in daemon mode. In the previous version ALSA connections were restored continually, and JACK connections whenever the connection graph had changed. In this version, aj-snapshot will restore connections once when it is started, and after that when new ports are registered in ALSA or JACK. You can specify (in milliseconds) how often aj-snapshot will poll for newly registered ports (-p --poll ms). The default is 1 second. This means that you have a lot more freedom to change the connections while the daemon is running, and gives you the opportunity to save (and maybe reload) a new connection state. Other changes include: - added manual file - rewrite of aj-snapshot.c to make it more readable and maintainable, and to have less code repetition. As always, there will be bugs, and I would be very grateful if you would report them on the sourceforge site. direct download: http://sourceforge.net/projects/aj-snapshot/files/aj-snapshot-0.9.5.tar.bz2… aj-snapshot website: http://aj-snapshot.sourceforge.net/ bug tracker: http://sourceforge.net/tracker/?group_id=311291&atid=1310488 feature requests: http://sourceforge.net/tracker/?group_id=311291&atid=1310491 to clone the git repository: git clone git://gitorious.org/aj-snapshot/aj-snapshot.git Hope you enjoy, lievenmoors

13 years, 10 months

[LAD] Guitarix release 0.22beta1

by Andreas Degert

Guitarix release guitarix2-0.22beta1 The next shiny new Guitarix version! Guitarix is a tube amplifier simulation for jack, with effect modules and an additional stereo effect chain. Download from http://sourceforge.net/projects/guitarix/ You can find some screenshots and explanations of the new version in http://sourceforge.net/apps/mediawiki/guitarix/index.php?title=EnhancedUI Many things changed in the user interface. Now you can move rack units by drag and drop (reflecting the signal flow), store individual settings for each rack unit and use preset banks with several settings for the whole rack. It's easy to take our "factory presets" and make your own customized bank, or make your own from scratch and share it on the Guitarix forum. There is a new "live play mode" with only the info you need on stage (it's fullscreen, no other penguins around), and a preset picking mode with a foot switch (midi or usb, or if you don't have one even the space bar of your keyboard) and the strings of your guitar to switch settings. Rack units are now put into categories, and two new ones are a noise gate for high noise levels and a univibe emulation. Thanks go to the developer of abGate and Ryan Billing from Rakarrack who helped porting his univibe code and made the inclusion of it possible. This is already too long, please check it out and give feedback if you find a problem, this version is still beta. Please refer to our project page for more information: http://guitarix.sourceforge.net/ download site: http://sourceforge.net/projects/guitarix/ please report bugs and suggestions in our forum: http://sourceforge.net/apps/phpbb/guitarix/ here you can find a couple of examples produced by guitarix users: http://sourceforge.net/apps/phpbb/guitarix/viewtopic.php?f=11&t=83 have fun _________________________________________________________________________ For extra Impulse Responses, guitarix uses the zita-convolver library, and, for resampling we use zita-resampler, both written by Fons Adriaensen. http://kokkinizita.linuxaudio.org/linuxaudio/index.html We use the marvellous faust compiler to build the amp and most effects and will say thanks to : Julius Smith http://ccrma.stanford.edu/realsimple/faust/ : Albert Graef http://q-lang.sourceforge.net/examples.html#Faust : Yann Orlary http://faust.grame.fr/ ________________________________________________________________________ guitarix development team

13 years, 10 months

[LAD] JACK session API in Perl, Python, Ruby, and Lua

by Joel Roth

https://github.com/navicore/Jacks Includes sample code for several languages. Developed by Ed Sweeney -- Joel Roth

13 years, 10 months

[LAD] Updates

by Fons Adriaensen

Updates available now at <http://kokkinizita.linuxaudio.org/linuxaudio/downloads> zita-alsa-pcmi-0.2.0 * Now includes some dirty hacks (implemented via the debug options parameter so they don't clutter the API) to support the -L option of zita-ajbridge-0.2.2 * Fixed the bug that made it fail with Jack period sizes >= 1024 (filter coefficients going out of range). * Added some hacks, activiated by the -L option, to make it work with ALSA's loop devices. The latter requires some clarification. The two apps provided by zita-ajbridge were never intented to be used with the ALSA loop devices. They were developed to provide additional and full quality capture or playback channels to Jack, using a real soundcard. Yet most people trying the first release used them with the loop devices, to provide a Jack interface to some misbehaving apps such as flash player or skype. Using the loop device is required since those apps refuse to work with ALSA's Jack plugin. It's a dirty hack using an unwanted resampling step, but it seems there is no working alternative. The ALSA loop devices presents themselves as hw: devices, but they don't behave like a real one: * Timing of period wakeups is quantised to a much too high value, and produces lots of jitter, worse than even the most crappy USB interface. It also makes using the device with small period sizes impossible, as wakeups occur when it's almost too late. The next cycle will xrun in that case. * The loop devices have 32 channels. They allow to be opened with a different channel count at both ends, but then make a mess of it. For example mplayer's two channels will be spread out over the available 32, making it run at 16 times normal speed. To run zita-a2j or zita-j2a with the ALSA loop device: * Use the -L option. This forces two things: the sample format will be 16-bit, and the device is opened in 2-channel mode. This makes it usable to apps at the other end which can't deal with floats or more than 2 channels. Note that just using -c 2 (which is also the default) still opens the ALSA device with all its channels, as must be done on a real hw: device, so that won't work. * Use at least -p 256 -n 3, or -p 512 or higher (on the ALSA device - Jack's parameters don't matter). Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

13 years, 10 months

[LAD] Timing of the ALSA loop device

by Fons Adriaensen

Hello, Does anyone know what sort of timer is used by the ALSA loop device, and if this can be changed in some way ? On the system I'm working on now, wakeup times while waiting for a period to be available (as done by Jack and zita-alsa-pcmi) seem to be quantised in steps of 3.3ms or so. Which means it sometimes wakes up 3/4 or more of a period too late. Next one will be xrun in that case. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

13 years, 10 months

[LAD] First release of zita-ajbridge

by Fons Adriaensen

Hello all, The first official release of zita-ajbridge is now available at <http://kokkinizita.linuxaudio.org/linuxaudio/downloads>. Quoting the README: This package provides two applications, zita-a2j and zita-j2a. They allow to use an ALSA device as a Jack client, to provide additional capture (a2j) or playback (j2a) channels. Functionally these are equivalent to the alsa_in and alsa_out clients that come with Jack, but they provide much better audio quality. The resampling ratio will typically be stable within 1 PPM and change only very smoothly. Delay will be stable as well even under worse case conditions, e.g. the Jack client running near the end of the cycle. You will also need the latest zita-resampler and zita-alsa-pcmi libraries. All feedback welcome. Ciao, -- FA Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.

13 years, 10 months

Re: [LAD] [Jack-Devel] First release of zita-ajbridge

by Stéphane Letz

Le 19 mars 2012 à 16:25, Nedko Arnaudov a écrit : > Stéphane Letz <letz(a)grame.fr> writes: > >> Le 19 mars 2012 à 16:15, Nedko Arnaudov a écrit : >> >>> Stéphane Letz <letz(a)grame.fr> writes: >>> >>>> Le 19 mars 2012 à 15:20, Fons Adriaensen a écrit : >>>> >>>>> On Mon, Mar 19, 2012 at 01:19:25PM +0100, Stéphane Letz wrote: >>>>> >>>>>> I'm wondering how your code could replace the "adapter" mechanism >>>>>> that I did in JACK2 for that never worked fully >>>>>> reliably. Basically the "adapter" idea is a bit more general in >>>>>> the sense that it aims at "adapting" streams in different >>>>>> context: like network <==> Audio API (CoreAudio for OSX, ALSA for >>>>>> Linux, PortAudio for Windows...), or several ALSA devices..etc.. >>>>>> (The adapters are also developed as "in server" clients, this >>>>>> make a bit more easy to add them in a running session using >>>>>> "ladish" kind of studio management approaches...) >>> In fact, ladish doesnt manage in-server clients at all. It can only >>> restore jack settings, and they could include stuff like in-process >>> clients and slave drivers. jackdbus provides engine parameter for >>> the latter but doesnt for the former. >>> >>> -- >> >> Well there is a control API to load/unload in-sever client. I was >> thinking this API could be used with ladish... > > ladish could be made to use this API. But it is not the case. At least > not yet. > > -- > Nedko Arnaudov <GnuPG KeyID: 5D1B58ED> Well this could be interesting to have access to "netjack2" components : netmanager, netadapter, audioadapters.... Stéphane

13 years, 10 months

Re: [LAD] First release of zita-ajbridge

by Fons Adriaensen

On Mon, Mar 19, 2012 at 06:58:29PM +0100, Robin Gareus wrote: > # mplayer -ao alsa,device=hw=1,0 some_music.wav > # zita-a2j -dhw:1,1 > > I get endless messages: > Starting synchronisation. > Detected excessive timing errors, waiting 15 seconds. > This may happen with current Jack1 after freewheeling. (meanwhile I'm back home, the loopback device is hw:3 here) [terminal 1] fons@zita1:/audio/audiofiles/tracks> mplayer -ao alsa:device=hw=3.1 diana-krall-almost-blue-44.wav MPlayer SVN-r34426-4.6.2 (C) 2000-2011 MPlayer Team 181 audio & 384 video codecs mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing diana-krall-almost-blue-44.wav. Audio only file format detected. Load subtitles in ./ ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... A: 4.0 (04.0) of 244.0 (04:04.0) 0.1% ... [terminal 2] fons@zita1:~> zita-a2j -d hw:3,0 -r 44100 Starting synchronisation /me makes connections in qjackctl and hears lovely piano intro... So this just works. I do indeed have to start mplayer first, but only the first time. After that I can stop and restart mplayer without problem. This is probably some quirk of the loopback device. There is no such thing as 'requesting exclusive access' in the ALSA pcm API - it depends only on the device. > The message is repeated every ~15 sec; and it does not stop. > Each time a message is printed there's a short very jittery sound (I can > discern the music) but it remains silent for the remaining time. The silence is intentional - the app waits for the dust to settle down after it has detected 'impossible' timing info from either Jack's DLL or from the one tracking the ALSA device. There was a bug in the loopback device - it returned the wrong type of event when using poll() in some cases. It's fixed in 1.0.24. That would certainly explain what you see. > How do you conduct this test? > > I assume that there is no other ALSA-client involved and you use the > loopback in float/32 mode, w/native samplerate. Zita-a2j and j2a use libzita-alsa-pmci to access ALSA devices. It uses exactly the same methods as Jack's backend - very early versions (libclalsadrv 7 years ago) were in fact a C++ copy of Jack's backend. That means that whatever works with Jack will work with zita-a2j and j2a or any of my apps using ALSA. AFAIK Jack doens't run well with dmix/dsnoop either. And anyway there's no reason to use them with Jack. To run the delay test (assuming hw:3 is the loopback device): [Terminal 1] zita-a2j -d hw:3,0 [Terminal 2] zita-j2a -d hw:3,1 [Terminal 3] jack_delay and make the required connections. The off-list reply was a typo - back on LAD now. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

13 years, 11 months

[LAD] update of zita-ajbridge

by Fons Adriaensen

Hello all, A bugfix update of zita-ajbridge is now available at the usual place. Thanks to Robin Gareus who discovered the bug: the typical last-minute-after-testing-added-line-typo. In this case it meant that both apps would use the lowest resampling quality in all cases... Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

13 years, 11 months

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Linux-audio-dev