Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7536 discussions

by Olivier Guilyardi

Hi everyone, In Jackbeat I perform sample rate conversion using libsamplerate in the realtime thread. I have recently realized that might be a mistake in regard to performance. The SRC is used on audio samples which sample rate might differ from the external (Jack) one, and also as a simple pitch shifter. But the point is that I do not need to do this heavy computation in realtime. The audio samples content is known in advance, it isn't as operating on live audio input. To my understanding, in the realtime thread one is located in some sort of abstract instant, between (input) now and (output) now, and therefore one has very little time. If some computation can be moved out of the RT thread, it ought to be. In this regard, I thought about setting a (or several) non-RT helper thread(s) up, where the heavy stuff would happen, and feed the converted content into the RT thread through a ringbuffer. I understand that this approach would require implementing a sync callback for the helper threads to get ready (fill buffers, etc...) whenever the transport position changes. I'm asking this because the SRC computation load currently has an impact on the output quality (ticks, xruns) with many tracks or on slower systems, and I'm thinking about adding real pitch and time shifting using Rubberband in the near future, which is certainly going to be a lot heavier. Is my idea of a helper thread correct? Any clue to set things up properly? Cheers, -- Olivier Guilyardi / Samalyse

17 years, 6 months

[LAD] LV2 in Ardour 3?

by Darren Landrum

I seem to recall some discussion involving the implementation of LV2 as a part of Ardour 3, along with the MIDI functionality. I was just hoping to confirm whether this is true or not. If it is, what LV2 extensions are going to be supported? I'm really hoping for MIDI, GUIs, and probably the port grouping as well. Thanks for the help! Regards, Darren Landrum

17 years, 6 months

Re: [LAD] Quick question on theory

by Jonathan Woithe

Hi Arnold > just a quick question so I don't look as a complete retard in my paper for a > conference: > The delta-peak contains all frequencies phase-aligned while white noise > contains all frequencies with random phase, right? That is my understanding, yes. To flesh it out a bit more you could add that the phase alignment in the delta-peak case occurs at the point of the delta-peak (assuming cosine decomposition), but that almost goes without saying. Regards jonathan

17 years, 6 months

[LAD] Quick question on theory

by Arnold Krille

Hi all, just a quick question so I don't look as a complete retard in my paper for a conference: The delta-peak contains all frequencies phase-aligned while white noise contains all frequencies with random phase, right? Have fun, Arnold -- visit http://www.arnoldarts.de/ --- Hi, I am a .signature virus. Please copy me into your ~/.signature and send me to all your contacts. After a month or so log in as root and do a "rm -rf /". Or ask your administrator to do so...

17 years, 6 months

[LAD] need help with Java audio problem

by Dave Phillips

Greetings, I'm writing an article about Java sound applications, and I've run into a problem. HighC, FScape, and jein all demonstrate the problem: They appear to be working as they should, but there's no audio output. According to HighC's author, he uses only the recommended class for audio output. I quote from our correspondence : "All I do is call javax.sound.sampled.AudioSystem.getLine() with the default settings. It should work as is, that's what the Java spec says. Perhaps the Java configuration files allow setting the default audio device on Linux?" So, what is the default audio device for Java 1.6 on Linux ? Is it configurable ? If so, where do I find it and how do I set it ? (Btw, I have the Sun SDK). I also have HighC installed under Wine, along with the JRE 1.6 for Windows. HighC works as expected there. I could not test jein under Wine, it requires javac, which is not included in the JRE. Any and all suggestions are most welcome, I'm totally clueless when it comes to debugging Java. Best, dp

17 years, 6 months

[LAD] Intel HDA Six-stack + Two

by Jens M Andreasen

I've now finally gotten an Analog Devices ad1988B with six in and ten out + digital - all in glorious 24bit/192K, which I find mighty impressive for an onboard three dollar chip. The sources for alsa-1.14 were included on the CD that came with the (Asus) box, and this is also what I am using now. The driver has fixes and workarounds for various note-book configurations with built in speakers and microphones as well as various numbers of physical multi-purpose ports that connects to the outside world. Unfortunately nobody thought of the possibility of a vendor wild enough to actually simultaniously implemement /everything/ this chip has to offer. Surround 7.1 on the "six-stack" on the backside is supported by alsa, but appears to then replicate the "surround front" to the headphone socket on the physical front panel which may or may not be what you expected. Getting skype and friends out of the mix - perhaps even on its own little /dev/dsp1 for closed source legacy? - would in my opinion have been more like it. Any ideas how to solve this? There is no such ugly invention as "surround 9.1" nor "surround 7.1+2" as of yet ... But that could of course be changed :-) I have some other minor issues as well, like the headphone amp being activated on the backpanel port labelled "line" but not on the frontpanel jack labelled "headphones" ... But those appears to be comparatively trivial to fix by comparing to how all the other special case configurations are implemented. --

17 years, 6 months

Re: [LAD] need help with Java audio problem: SOLVED, sort of

by Dave Phillips

Pedro Lopez-Cabanillas wrote: > Maybe the default sound device is still /dev/dsp using the OSS API. Can you > test if the OSS emulation is working properly in your system? For some reason I had not tested HighC on my 64-bit system (64 Studio) until this morning. It works ! Okay, at this point I think it's a distribution problem, not a problem with coding practice or ALSA configuration. However, your pointers certainly got me going in the right direction. I'll contact the maintainers of JAD 1.0 to see if they've done anything unusual with OpenSUSE's default Java settings. Thanks again, Pedro, your assistance has been hugely appreciated. :) Best, dp

17 years, 6 months

[LAD] Car engine sound emulation for future electic cars. ideas ?

by Benno Senoner

Hi all, since the price of fossil fuels is quickly going up to unsustainable levels the mass production of electric cars is only a couple years away. (there are already a few nice around but not yet high volume production and cost an arm and a leg) http://en.wikipedia.org/wiki/Electric_car One peculiarity of electric cars is that they emit almost no noise and sometimes engine noise could come handy. For example blind people or even distracted pedestrians about to cross a street cannot hear electric cars coming which can lead to accidents, injuries and even loss of life. There are already electric car companies offering artificial engine sound in their cars. A linux based embedded system would probably be a nice candidate to provide such functionality. (ARM cpu, some RAM and audio DAC, or even a nano-itx x86 board which could do other additional things at the same time, in-car computer etc) A nice system would be one which allows pluggable sounds, for example ferrari or porsche sound :) My question is what's in your opinion the best method to simulate realistic car sounds without making the system too complex or too costly. I was thinking whether multisampling the engine sound at various RPMs (rotations per minute, frequency proportional to RPM) could be a solution or if some kind of synthesis would be needed. The sound generator should in theory take only one input variable, the motor's RPM (which can expressed as 0..100%) and then generate a sound with a frequency proportional to the RPM. A few ideas that come to mind which could provide varying degrees of realism (I don't know because I did not try). 1) sampling an engine sound at middle RPM and then pitchshifting it with formant correction up and down. easy to do but probably does not sound so good at extreme pitchshifting ranges. 2) sampling an engine sound at various RPMs, the more the better let's say more times per octave and then using some sort of sample pitching (standard interpolation or pitchshifting like rubberband) and then crossfade between samples. But if the samples to be crossfaded need probably to be phase-aligned otherwise it could lead to chorus style artifacts etc. 3) sampling a full, very slow RPM sweep both upwards (min to max) and downwards (max to min), lets say a couple of min per direction. Then the sound generator figures figures out the RPM direction (by comparing two subsequent values or some shortterm mean value) and then plays the sound using the right position in the wave. So for example if you floor the accelerator you have to play the upwards audio wave but you have to skip lots of parts so you need some trick to make it sound smoothly (time-compression comes to mind but I'm not sure if it is ideal) Not sure how hard it would be to sample those sound sweeps. You need to accelerate very slowly which can be tricky. Perhaps you need some servo motor which can precisely move either the accelerator pedal or is attached directly to the accelerator cable. Not very userfiendly. 4) sample the engine sound at various RPM. feed into a DSP algorithm which analyzes the timbre and then builds a model for a synthesizer. any other ways I am missing ? What's the best trade off between realism and complexity in your opinion ? Thanks for your comments. cheers, Benno

17 years, 6 months

[LAD] [OT] vector drawing software

by Fons Adriaensen

Anyone knows a good vector drawing program for Linux ? Absolute requirements are: - Lines, arrows, boxes, circles, etc. - Linewidths and styles, colors, filling. - Text - PDF or PS export. - PNG and JPEG import (no bitmap editing required). - Accuracy. I've been using TGIF for years, but I'm more and more being blocked by its main flaw which is that it seems to use a unit of 0.2mm internally (in metric mode) which is orders of magnitude too big. Tried QCAD and INKSCAPE, both fail basic requirements (and have other problems). Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia O tu, che porte, correndo si ? E guerra e morte !

17 years, 6 months

[LAD] [ANN] CLAM 1.3, 'Shooting flying plugins' release

by David García Garzón

The CLAM team enraptured to announce the 1.3.0 release of CLAM [1], the C++ framework for audio and music, code name ''The Shooting of the Flying Plugins release''. Highlights of this release are: - Automatic binary generation of LADSPA plugins containing the network you are editing in NetworkEditor. - Also a new simple API to code CLAM based Ladspa by hand. See [2] - More FAUST integration into network editor: edit faust code, compile, reload, view the svg diagrams (Natanael Olaiz GSoC) - Lots of usability enhancements on the NetworkEditor: cut&paste, context menus to connect ports, keyboard shortcuts, default double click actions, and a processing tree filter (Natanael Olaiz GSoC) - Annotator has also enhanced its functionality (Wang Jun GSoC): - You can build a project that aggregates content from several extractors - Extractors may have a config file - Extractors can write back data (useful if the extractor is a database of webservice and needs to upload modifications) - New ProgressControl widget and paired AudioFileMemoryLoader processing to support seeking (Pawel Bartkiewicz GSoC) - A bunch of new 3D spatialization processings from CI Barcelona Media[3] audio research group. - Scripts [4] and graphical front-end [5] to generate a native CLAM plugin project from scratch. - Experimental Python bindings [6] - TickExtractor example is compiling again (many thanks to Amaury Hazan from MTG-UPF) - Development deployment for Windows native compilation using MinGW (Wang Jun GSoC) And a lot of small nice features and fixes you will appreciate for sure. Source and binary packages for different platforms are available at the CLAM download page [7]. See also: development screenshots [8], the CHANGELOG [9], and the version migration guide [10]. We are also very excited on what next releases promise us. Some ongoing work: - Generating other types of network based plugins and programs (LV2, JACK, VST...), - Subnetworks (Natanael Olaiz GSoC) - Improved OSC support, 3D scene descriptors parametrization receivers processings and Blender exporter to the spatialization processing choreographer. (Natanael Olaiz GSoC) [11] - Typed controls (Francisco Tufro GSoC) - A new musician-oriented standalone chord extraction application (Pawel Bartkiewicz GSoC) [1] http://clam.iua.upf.edu [2] http://iua-share.upf.edu/wikis/clam/index.php/Building_a_LADSPA_plugin [3] http://www.barcelonamedia.org/ [4] http://audiores.uint8.com.ar/blog/2008/07/07/clam-processing-generator-scri… [5] http://clam.iua.upf.edu/wikis/clam/index.php/Image:ProcessingCodeGenerator.… [6] http://audiores.uint8.com.ar/blog/2008/08/03/interactive-clam-programming/ [7] http://clam.iua.upf.edu/download.html [8] http://clam.iua.upf.edu/wikis/clam/index.php/Development_screenshots [9] http://iua-share.upf.edu/svn/clam/trunk/CLAM/CHANGES [10] http://iua-share.upf.edu/wikis/clam/index.php/Version_Migration_Guide [11] http://dadaisonline.blogspot.com/search/label/blender related blogging -- David García Garzón (Work) dgarcia at iua dot upf anotherdot es http://www.iua.upf.edu/~dgarcia

17 years, 6 months

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Linux-audio-dev