Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7520 discussions

by Maarten de Boer

Hello, I am looking for a cross-platform implementation of an atomic integer. Under Linux, a build an c++ class "atomic" around asm/atomic.h, (which I can use as if it where an int), but I'd like to have a solution that also works on Windows XP and Mac OS X. Thanks for any suggestions, Maarten

19 years, 5 months

Re: [linux-audio-dev] fst, VST 2.0, kontakt

by Forest Bond

> > In any case, Linux Sampler and Kontakt are > > different sorts of samplers at this point. Linux Sampler is more in the spirit > > of Gigasampler, and Kontakt is more like an emulation of a hardware sampler (but > > vastly more flexible). > > The key words there are "at this point." As noted, the *long-term* > potential of Linux Sampler is probably still much greater than Kontakt-in-wine. Agreed. Maybe time is best spent improving linux sampler. Then again, good VST has benefits, although I do see how it may hurt the long term goals of free software. I'm just not sure if the best way to help free software is to limit it's ability to integrate with proprietary software. > > > > Noted. However, I remain unconvinced that Linux Sampler is intended to be a > > drop-in replacement for something like Kontakt. > > I don't think it's intended to be a drop-in replacement for Kontakt; I > think it's just intended to be cool. I can appreciate that. -Forest

19 years, 5 months

[linux-audio-dev] [ANN] jack_capture V0.3.1, das_watchdog V0.2.2 and Mammut V0.22

by Kjetil S. Matheussen

http://ccrma.stanford.edu/~kjetil/src/ jack_capture ************************************************************************* jack_capture is a small program to capture whatever sound is going out to your speakers into a file without having to patch jack connections, fiddle around with fileformats, or set options on the argument line. This is the program I always wanted to have for jack, but no one made. So here it is. Changes 0.2.4 -> 0.3.1: ----------------------- *Reduced CPU usage a lot because of better disk handling. (25% -> 1%) *Make sure the rest of the recorded file is not garbage in case of an overrun. *Added the port argument, which can be specified many times and accepts both input and output port names (including regexp expressions). This makes jack_capture to completely replace jackrec. *Rewrote buffer handling. Silence is now inserted when underruns occure. Previously, the file became shorter than the recording in case of underrun. It can still happen though, but much more seldom, and a warning about that will be printed to the terminal. *Last rests of jackrec code has been rewritten. Well, all the code with substance, at least. *Nicified code a lot. *More efficient way of handling overruns. *Fixed really stupid compilation error. Thanks to Dragan Noveski for spotting it. das_watchdog ************************************************************************* Whenever a program locks up the machine, das_watchdog will temporarily sets all realtime process to non-realtime for 8 seconds. You will get an xmessage window up on the screen whenever that happens. Changes 0.2.2->0.2.3 -------------------- *Fixed commandline arguments for increasetime, checktime and waittime. *Nicified source a bit Mammut ************************************************************************* Mammut will FFT your sound in one single gigantic analysis (no windows). These spectral data, where the development in time is incorporated in mysterious ways, may then be transformed by different algorithms prior to resynthesis. An interesting aspect of Mammut is its completely non-intuitive sound transformation approach. Changes 0.21->0.22 ------------------ *Added patch and instructions from Owen Green on how to make mammut compile on OSX. Thanks! (Sorry, I forgot to release this version for almost a year...)

19 years, 5 months

[linux-audio-dev] More pictures from LAC2006

by Asbjørn Sæbø

A bit late, but maybe still of interest. A number of pictures from LAC2006 may be downloaded from <URL: www.q2s.ntnu.no/~asbjs/pictures_lac2006_asbjorn-sabo.tar >. (The file is around 9MB.) * netjack workshop * Linux sound night * various With thanks for and interesting conference and nice company there! Asbjørn Sæbø

19 years, 5 months

Re: [linux-audio-dev] A bit of an [ANN] :)

by Dave Robillard

james(a)dis-dot-dat.net wrote: >Howdy peeps. > >Not really much of a release, so not much fanfare, but in the >interests of sharing effort I give you... > >Powernap! > >http://dis-dot-dat.net/?item=code/powernap/ > >If, like me, you need to drive an app from a Python interface, >powernap can help. It's a small extension that switches to the >real-time scheduler and provides two handy sleep-like functions: > >nap() naps for a given number of milliseconds, timed with the RTC > >rnap() is a "rolling nap" that tries to make the time between calls >the given number of milliseconds. A padding nap, if you will. > >It works for me and it might come in handy for someone else. Nice, this will definitely come in handy for me. Have you done any benchmarking on what kind of timing precision you can get? -DR-

19 years, 5 months

[linux-audio-dev] modular sequencing environment/synth // any projects to dig in?

by Niklas Klügel

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Oi, I am currently writing on a modular synth with great emphasis on sequencing (but allowing tight interaction between both parts) in a comparable modular way (pattern creation/algortihmic composition/chord filters etc), I wrote some parts of the core of the application but I am not really keen on coding something that has been done several times before; that's why I thought about joining/"forking" a more complete modular synthesizing environment and adding the core components I need/have. The point is, that I don't know enough about the existant modular synthesizers to evaluate how modular their sourcecode is written so that my modifications are easily applicable in their source. Here are some parts/functionalities that I already have (partially) implemented and would like to add to the core: - - the main graph of interconnected modules supports heterogene module types for example it could support FAUST code-pieces represented in modules as well as ladspa-modules, the subgraph created by interconnected modules of the same type is the sent to the type-specific "compiler" creating executable signal paths - - the executable graph of interconnected modules is analyzed and split up in parts that can be executed in parallel, locks in the necessary parts are added and the whole graph can be executed in several threads - - modules are mainly connected using a zoomable patch-matrix; the zoomlevel corresponds to the level of typing (of connections) As I mainly write QT-Applications I'd prefer a project that can be either easily extended to use QT or which already uses it. I think it is better to work on an existing project, since I am currently busy with releasing an album and university-shiznit and I can not imagine doing everything in a prolific way in parallel. Thanks in advance for any input! So long... Niklas -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.3 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEq/F++k24EnBNzsMRAlNrAJ97l5nZeqM8B86w4FDrh34joLvZEQCeOxpn 1VD/QOT47Nfd2X6kEItBFRw= =jz9s -----END PGP SIGNATURE-----

19 years, 5 months

[linux-audio-dev] A bit of an [ANN] :)

by james＠dis-dot-dat.net

Howdy peeps. Not really much of a release, so not much fanfare, but in the interests of sharing effort I give you... Powernap! http://dis-dot-dat.net/?item=code/powernap/ If, like me, you need to drive an app from a Python interface, powernap can help. It's a small extension that switches to the real-time scheduler and provides two handy sleep-like functions: nap() naps for a given number of milliseconds, timed with the RTC rnap() is a "rolling nap" that tries to make the time between calls the given number of milliseconds. A padding nap, if you will. It works for me and it might come in handy for someone else. James

19 years, 5 months

[linux-audio-dev] Set rate return zero

by Huu Phuoc

Hi everybody! I am a newbie to alsa programming. I am trying to follow the article which locates at http://www.suse.de/~mana/alsa090_howto.html to develop a playback program. My code here: snd_pcm_t*pcm_handle; unsigned int rate = 8000; // Sample rate returned by // snd_pcm_hw_params_set_rate_near int exact_rate; // exact_rate == rate ----> dir =0 //exact_rate < rate ----> dir =-1 //exact_rate > rate ----> dir =1 int dir = 0; // Number of periods int periods = 2; snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; // This structure contains information about the hardware //and can be used to specify the configuration to be used for the PCM stream. snd_pcm_hw_params_t *hwparams; // Name of the PCM device, like plughw:0,0 //The first number is the number of the soundcard, //the second number is the number of the device. char *pcm_name; // Init pcm_name. Of course, later you //will make this configurable ;-) pcm_name = strdup("plughw:0,0"); printf("Device name:%s\n",pcm_name); //Allocate the snd_pcm_hw_params_t structure on the stack. snd_pcm_hw_params_alloca(&hwparams); // Open PCM. The last parameter of this function is the mode. // If this is set to 0, the standard mode is used. Possible // other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. // If SND_PCM_NONBLOCK is used, read / write access to the // PCM device will return immediately. If SND_PCM_ASYNC is // specified, SIGIO will be emitted whenever a period has //been completely processed by the soundcard. if (snd_pcm_open(&pcm_handle, pcm_name, stream, 0) < 0){//SND_PCM_NONBLOCK fprintf(stderr, "Error opening PCM device %s\n", pcm_name); return; } // Init hwparams with full configuration space if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) { fprintf(stderr, "Can not configure this PCM device.\n"); return; } // Set access type. This can be either SND_PCM_ACCESS_RW_INTERLEAVED or // SND_PCM_ACCESS_RW_NONINTERLEAVED.There are also access types for // MMAPed access, but this is beyond the scope of this introduction. if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { fprintf(stderr, "Error setting access.\n"); return; } //Set sample format if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0) { fprintf(stderr, "Error setting format.\n"); return; } // Set sample rate. If the exact rate is not supported by the hardware, use nearest possible rate. exact_rate = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &rate, &dir); if (exact_rate < 0) { fprintf(stderr, "Error setting rate.\n"); return; } if (rate != exact_rate) { fprintf(stderr, "The rate %d Hz is not supported by your hardware.\n=> Using %d Hz instead.\n", rate, exact_rate); } // Set number of channels if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2) < 0) { fprintf(stderr, "Error setting channels.\n"); return; } // Set number of periods. Periods used to be called fragments. if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) { fprintf(stderr, "Error setting periods.\n"); return; } // Set buffer size (in frames). The resulting latency is given by // latency = periodsize * periods / (rate * bytes_per_frame) snd_pcm_uframes_t buffersize = ( snd_pcm_uframes_t )( number_of_frames * periods ); if (snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &buffersize) < 0){ fprintf(stderr, "Error setting buffersize.\n"); return; } printf("Buffer size asked for %d, set %d \n",buffersize,exact_rate); // Apply HW parameter settings to // PCM device and prepare device if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) { fprintf(stderr, "Error setting HW params.\n"); return; } snd_pcm_prepare(pcm_handle); Unfortunately, exact_rate is zero.The following lines were printed out: The rate 8000 Hz is not supported by your hardware. => Using 0 Hz instead. If exact_rate > 0 is OK but it is zero. I don't know how to solve the problem. Please help me. Thanks! Phuoc Nguyen

19 years, 5 months

[linux-audio-dev] Converting a 24bit sample to 16bit

by James Courtier-Dutton

Hi, Is there a standard way of converting a 24bit sample to 16bit? I ask because I think that in different scenarios, one would want a different result. 1) scale a 24bit value to a 16bit by simple multiplication by a fraction. 2) bit shift the 24bit value, so that the "most useful 16bits" are returned to the user. The problem here is what are the "most useful 16bits"? I have one application where just using the lower 16bits of the 24bit value is ideal, due to extremely low input signals. Option (1) looses information due to compression of the signal. Option (2) is more likely to loose information through clipping. James

19 years, 5 months

[linux-audio-dev] Re: LinuxSampler license

by Juhana Sadeharju

In gimp list, I mentioned that I don't want my software to be used in Windows. That would encourage people to install Linux. My plan was to use GPL + Windows exclusion. I was very clearly informed that it would not work. Then why similar works for Linuxsampler? BTW, I'm still puzzled on what kind of license I should use. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 5 months

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Linux-audio-dev