Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7520 discussions

prototype in header but declaration where?

by Renato

Hi, very noob C question. I do this: renato@acerarch /usr/include/alsa $ grep snd_seq_open * seq.h:int snd_seq_open(snd_seq_t **handle, const char *name, int streams, int mode); seq.h:int snd_seq_open_lconf(snd_seq_t **handle, const char *name, int streams, int mode, snd_config_t *lconf); so I see that the function snd_seq_open has its prototype declared in seq.h... but where is the actual function definition? cheers renato

14 years, 4 months

[ANN] International Conference on Spatial Audio, Nov 10-13 2011, in Detmold, Germany

by Jörn Nettingsmeier

hi *! sorry for the slightly off-topic post, but since spatial audio has been a frequent topic lately, i think some people here might be interested. linux or FLOSS won't be exactly in the limelight, but yours truly will make sure there are at least 2-3 boxes with your favourite OS and audio tools humming along in various places. oh, and you might come early and watch a few high-end mixing consoles boot - the startup screen will bring tears to your eyes (as will the price tag, unfortunately :) unfortunately, there will have to be an admission fee, which we haven't decided on yet. but we're trying to keep it reasonable. don't shout at me when it turns out to be a bit more costly than LAC, though... jörn *.* ICSA 2011 - International Conference on Spatial Audio November 10 - 13, Hochschule für Musik, Detmold Organizers: Verband Deutscher Tonmeister (VDT), in cooperation with Deutsche Gesellschaft für Akustik e.V. (DEGA), and European Acoustics Association (EAA). Contact/Chair: Prof. Dr.-Ing. Malte Kob Erich-Thienhaus-Institut Neustadt 22, 52756 Detmold Mail: icsa2011attonmeister.de Phone: +49-(0)5231-975-644 Fax: +49-(0)5231-975-689 Summary: The International Conference on Spatial Audio 2011 takes place from November 10 to 13 at Detmold University of Music. This expert‘s summit will examine current systems for multichannel audio reproduction and complementing recording techniques, and discuss their respective strengths and weaknesses. Wavefield synthesis systems, a higher-order Ambisonics array, as well as 5.1/7.1 installations in diverse acoustic environments will be available for comparative listening tests during the conference. Structured plenary talks, paper and poster sessions will revisit fundamentals and present latest research. A series of workshops will be dedicated to practical implementations of spatial sound capture and playback methods, and their esthetic and psychoacoustical implications for music perception. Concerts that include music specially arranged for the conference will let you experience various spatial sound systems in "live" conditions. Call for papers and music: Your contributions are welcome, either as presentations, posters, or workshops. Submissions will undergo a review process, and accepted contributions will be published in the conference proceedings. The conference language is English. We are planning structured sessions on the following topics: * Multichannel stereo * Wave field synthesis * Higher-order Ambisonics / spherical acoustics * 3D systems * Binaural techniques An additional session will be dedicated to related miscellaneous contributions, such as hybrid systems and perception/evaluation of spatial music reproduction.

14 years, 4 months

Help with ALSA plugin - adding real time parameter

by Bob DeCarlo

Hi, I am trying to get my own ALSA plug-in to work with some real time controls. ( Is this the right place to ask?) I am successful with the PCM part of it. What I mean is from: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html External Plugin: Filter-Type Plugin static snd_pcm_sframes_t myplg_transfer(snd_pcm_extplug_t *ext, const snd_pcm_channel_area_t *dst_areas, snd_pcm_uframes_t dst_offset, const snd_pcm_channel_area_t *src_areas, snd_pcm_uframes_t src_offset, snd_pcm_uframes_t size) { // my PCM processing works // I want to add a control parameter that can be set in real time by an app } SND_PCM_PLUGIN_DEFINE_FUNC(myplg) { // This all works for PCM processing just like the examples ... err = snd_pcm_extplug_create(&mix->ext, name, root, sconf, stream, mode); if (err < 0) { free(mix); return err; } ... } SND_PCM_PLUGIN_SYMBOL(myplg); Now I want to add a simple real time adjustment, an integer value that can be sent by an application to adjust the sound (PCM samples) in real time. I tried some ways of doing it it but without success, I am not understanding the basics. I first looked at doing a ctl. I was able get separately (without PCM processing) get a control to work but that looks like ts only for hardware control? I can't connect it to my PCM processing. http://www.alsa-project.org/alsa-doc/alsa-lib/ctl_external_plugins.html I looked at LADSPA but I'm not sure where that is going to take me. I am now looking at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html External Plugin: I/O Plugin But, I am confused because I also see pcm type callbacks. So, I do not have a specific coding question (yet) but I just need a general direction. ... -Should I use ctl_external_plugins and figure out how to use it with my PCM? -Should I use a LADSPA example? -Should I go wth External Plugin: I/O Plugin? -Something else? I hope I am clear enough about my question and thanks for any pointers you can provide. Bob

14 years, 4 months

excellent overview article on realtime audio programming

by Paul Davis

Ross Bencina is the author of AudioMulch and has been extremely involved in PortAudio, ReacTable, and other projects. His new article on realtime audio programming is a MUST read for anyone new to the area, and worth reading as a reminder even for experienced developers. http://www.rossbencina.com/code/real-time-audio-programming-101-time-waits-… I'm off to count how many violations Ardour contains ... --p

14 years, 4 months

New Java binding for JACK

by Neil C Smith

Hi All, Just a quick post to let you know about a new JACK binding for Java, called JNAJack, at http://code.google.com/p/java-audio-utils/ JNAJack is a minimal object-oriented wrapper to the JACK Audio Connection Kit API. It uses Java Native Access (JNA) rather than custom JNI to interface with the underlying Jack API, simplifying development and deployment - no compilation required and it (*should*) work cross platform. Use of JNA means that performance is not quite on a par with the custom JNI code in JJack, but it is still fine for low latency usage, and some further performance optimisations are in the wings. Unlike JJack the aim of this project is to support full and typesafe OOP access to the Jack API from Java, and nothing else. Most important aspects of the audio API are included. MIDI and transport support will be implemented in the future. Well, I say this is new, it was mostly written quite a while back as part of my Praxis InterMedia System project http://code.google.com/p/praxis/ (a Java based cross-media patcher environment). I'm finally getting around to releasing some of this stuff separately as well. Best wishes, Neil -- Neil C Smith Artist : Technologist : Adviser http://neilcsmith.net

14 years, 4 months

Testdriven development with JACK?

by Adrian Knoth

Hi! This question could have also been asked on jack-devel, but since LAD probably has a broader audience: I recently started hacking on a jack-driven matrix mixer (goals so far: GUI, maybe network controls (OSC?), maybe LV2 host), and I wonder if there are "frameworks" for test-driven development, so I can come up with unit and acceptance tests while implementing new functionality. Has anyone ever done test-first with jack? One could start jackd in dummy mode with a random name, start some clients, wire inputs to outputs and compare the generated signal to the expected result, maybe with some fuzzy logic to allow for arbitrary delays. OTOH, if there are existing mocking libraries for jackd, things might be a bit more straight forward (provide an input buffer to be returned by jack_port_get_buffer, call the process function and check the result that's written to the output buffer). Any pointers will be highly appreciated. Cheers

14 years, 4 months

Python Gstreamer Live Video examples

by Nathanael Anderson

I'm just getting started with some python gstreamer coding, and Having trouble finding what I'd consider some basic examples: These I've already got working: 1. build a pipeline showing the attached video device like: gst-launch v4l2src device=/dev/video0 ! 'video/x-raw-yuv,format=(fourcc)YUY2,width=640,height=480,framerate=20/1' ! xvimagesink 2. Record and show video at the same time: gst-launch v4l2src device=/dev/video0 ! 'video/x-raw-yuv,width=640,height=480,framerate=20/1' ! tee name=t_vid ! queue ! xvimagesink sync=false t_vid. ! queue ! videorate ! 'video/x-raw-yuv,framerate=20/1' ! theoraenc ! queue ! oggmux ! filesink location=test.ogg I want to get working: 3. toggle recording, so I can show a video monitor, and without destroying and recreating the pipeline start or stop video recording at will. 4. Sync multiple pipelines, my end goal is to be able to record x video streams in sync, while playing back y videos with all play and record sources in sync (think video mixing). I'm looking for examples of 3 and 4, and improvement ideas to 1 and 2. Thanks, Nathanael

14 years, 4 months

Question about LV2 UI extension

by Paul Coccoli

I was looking at the (unfinished) example sampler plugin here: https://gitorious.org/gabrbedd/lv2-sampler-example I ran through the mental exercise of trying to figure out how to finish it, and I have a question about the UI extension. How does a UI tell its plugin to load a sample file? The example has a TODO in its run function that indicates it will react to an LV2_Event that contains a pathname for a file. I don't understand how a UI will create this event. On lv2plug.in, I see there's an experimental string-port extension that defines a transfer mechanism for strings. Is this the recommended method? Do any hosts support this extension? There's also an atom extension, but I don't think I grok it yet. Can I create a port of type atom:MessagePort? How does a UI make use of that?

14 years, 4 months

Fwd: Fwd: lv2 extension bugs

by Lieven Moors

On Mon, Jul 25, 2011 at 4:02 PM, David Robillard <d(a)drobilla.net> wrote: > On Mon, 2011-07-25 at 13:05 +0200, Lieven Moors wrote: > > OK, what happened was that I landed on the http://lv2plug.in/ns/ext > > page, was expecting a download extensions link, didn't find it, and > > downloaded > > the files manually from the links on those pages. > > The event extension page > > http://lv2plug.in/ns/ext/event > > does have a link to the latest release (1.2). > > -dr > > That is really odd. Did it change in the last couple of days? I'm sure I got the header from there, and I'm sure it had http in the address on line 28. Otherwise this is not a bug, it's a ghost... lievenmoors

14 years, 4 months

[RME - hdsp] Multiface/Digiface/RPM detection

by Adrian Knoth

Hi! I've recently added support for the RME RPM to hdspmixer. Unfortunately, I don't have one, it's been done blindly with user feedback. This very user now reports that he needs to upload the device firmware from windows. I've checked hdsploader, and of course, it needs patching. I'll take care in a second. More surprisingly, though, the kernel wasn't able to upload the firmware itself, because it fails to detect the RPM and hence tries to upload a multiface firmware. After reading the kernel source, I think the code in hdsp.c is wrong: if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) hdsp->io_type = RPM; else hdsp->io_type = Multiface; } else { hdsp->io_type = Digiface; } Who here owns a Digiface and can confirm or deny that the kernel correctly detects it as Digiface? Same for Multiface, though I guess since it's more or less the default, users wouldn't notice it. What's wrong with the code above? I think all occurrences of HDSP_control2Reg in hdsp_check_for_iobox need to be changed to HDSP_controlRegister and the second hdsp_fifo_wait needs to be inverted. But this is pure guesswork. If I come up with a patch, who here has a RPM, Digiface or Multiface to test it? TIA

14 years, 4 months

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Linux-audio-dev