Linux-audio-dev July 2006

linux-audio-dev@lists.linuxaudio.org

76 participants
53 discussions

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

18 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

18 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

18 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

18 years, 5 months

[linux-audio-dev] Trying to build Om,

by Stephen Cameron

So, I'm trying to build Smack, which needs Om, which needs libgnomecanvasmm, which my os (Fedora Core 3) doesn't seem to have, so I found it here: http://sunsite.mff.cuni.cz/MIRRORS/ftp.gnome.org/pub/GNOME/sources/libgnome… And when I try to build it, I get this: g++ -DHAVE_CONFIG_H -DG_LOG_DOMAIN=\"libgnomecanvasmm\" -I../../libgnomecanvas -I../../libgnomecanvas -DXTHREADS -D_REENTRANT -DXUSE_MTSAFE_API -I/usr/local/include/libart-2.0 -I/usr/include/gtkmm-2.4 -I/usr/lib/gtkmm-2.4/include -I/usr/include/glibmm-2.4 -I/usr/lib/glibmm-2.4/include -I/usr/include/gdkmm-2.4 -I/usr/lib/gdkmm-2.4/include -I/usr/include/pangomm-1.4 -I/usr/include/atkmm-1.6 -I/usr/include/gtk-2.0 -I/usr/include/sigc++-2.0 -I/usr/lib/sigc++-2.0/include -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/lib/gtk-2.0/include -I/usr/X11R6/include -I/usr/include/pango-1.0 -I/usr/include/freetype2 -I/usr/include/freetype2/config -I/usr/include/atk-1.0 -I/usr/include/libgnomecanvas-2.0 -g -O2 -MT line.lo -MD -MP -MF .deps/line.Tpo -c line.cc -fPIC -DPIC -o .libs/line.o In file included from line.cc:3: ../../libgnomecanvas/libgnomecanvasmm/line.h:374: error: extra qualification ignored ../../libgnomecanvas/libgnomecanvasmm/line.h:375: error: explicit specialization of non-template `Glib::<anonymous class>' ../../libgnomecanvas/libgnomecanvasmm/line.h:375: error: an anonymous union cannot have function members ../../libgnomecanvas/libgnomecanvasmm/line.h:378: error: abstract declarator `Glib::<anonymous class>' used as declaration make[4]: *** [line.lo] Error 1 make[4]: Leaving directory `/home/scameron/software/libgnomecanvasmm-2.6.0/libgnomecanvas/libgnomecanvasmm' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/scameron/software/libgnomecanvasmm-2.6.0/libgnomecanvas/libgnomecanvasmm' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/scameron/software/libgnomecanvasmm-2.6.0/libgnomecanvas' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/scameron/software/libgnomecanvasmm-2.6.0/libgnomecanvas' make: *** [all-recursive] Error 1 [scameron@zuul libgnomecanvasmm-2.6.0]$ Anybody know how to get around this...? (without installing another OS preferably) -- steve __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 5 months

[linux-audio-dev] IR FFT smoothing

by Andrew Gaydenko

Hi! Trying to understand some magic with Ipmulse Response, FFT and so on, I'm playing with scipy, matplotlib (and python as new language for me as a bonus). Concrete IR was got with help of DRC (rec_imp, saying more strictly). Currently I have got this frequency response graph: http://gaydenko.com/mix/needSmooth.png And the question is: what is common way to smooth the result? Some offtopic apps has something like "1/24, ..., 1/3 octave smoothing". What does it mean? Andrew

18 years, 5 months

[linux-audio-dev] What valgrind says

by james＠dis-dot-dat.net

Hi peeps. I've just been running an app through valgrind and I'm getting a few of these: ==11955== Syscall param write(buf) points to uninitialised byte(s) ==11955== at 0x4D51BDB: (within /lib64/libpthread-2.4.so) ==11955== by 0x4B27CDD: (within /usr/lib64/libjack.so.0.0.23) ==11955== by 0x4B29434: jack_deactivate (in /usr/lib64/libjack.so.0.0.23) ==11955== by 0x4B2945F: jack_client_close (in /usr/lib64/libjack.so.0.0.23) ==11955== by 0x40402F: jackpart_close() (jackPart.cc:86) ==11955== by 0x40235F: mainLoop() (sampleplayer.cpp:438) ==11955== by 0x40237B: main (sampleplayer.cpp:444) ==11955== Address 0x7FEFFE934 is on thread 1's stack I can't see anything wrong on my part, so I thought I'd check with people who know more than I before I tear my program apart. I also get them coming from jack_port_register, jack_activate and jack_connect. Any ideas? James

18 years, 5 months

[linux-audio-dev] Call for all LA projects

by Ivica Ico Bukvic

Linuxaudio.org is getting ready to announce new memberships in the coming weeks. For this reason, I would like to invite all Linux Audio projects and its members, as well as other allied projects, institutions, companies, and hardware vendors to consider joining our organization. HOW TO JOIN Simply send an e-mail to ico at linuxaudio dot org stating your interest to join. There are no dues or hidden costs and the membership can be terminated at any time via member's written request. BENEFITS As the consortium grows, we strive to provide a focal, converging point through which we will represent our community, offer new opportunities for collaboration and interaction, represent and preserve interests of the community, as well as serve as a point of contact for commercial industry and hardware vendors. For more info please visit linuxaudio.org or do not hesitate to contact me directly. Best wishes, Ivica Ico Bukvic, D.M.A. Composition Virginia Tech Dept. of Music - 0240 Blacksburg, VA 24061 (540) 231-7047 (540) 231-5034 (fax) ico(a)vt.edu http://www.music.vt.edu/people/faculty/bukvic/

18 years, 5 months

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

by Forest Bond

> You are... Linux sampler doesn't support built-in effects because of what > working with audio is like under linux. JACK (jack audio connection kit) > is an application that allows one to manage connection between various > audio software. This includes linux sampler, but also plugins based on the > LADSPA sdk. You are right to say that there are no built-in effect in > Linuxsampler. It's not a lack of functionnality. It's meant to be like > this because you can connect linuxsampler's output to a effect management > rack like jack-rack (which also benefits of being jack compatible) have > any effect you like applied (and there's loads ! see > http://www.ladspa.org/). I can appreciate this modularity as much as the next fellow, but having effects support in the sampler itself is near-critical. Creating sophisticated sounds using a sampler requires it. I need to have multiple layers of samples, each filtered and modulated differently. You can't do this in post-processing. It just isn't possible. Even supposing Linux Sampler could route sounds out to effects and back in to the sampler, that would be far to complicated to be even remotely usable without giving up huge amounts of sampling felixibility. This is not about applying reverb to a drum kit. It's about having 6 samples triggered when I press a key on the keyboard, and each sample being filtered, pitch bended, saturated differently, and modulating all of those parameters using LFO's, envelopes, and midi parameters. You can't do that using "outboard" effects. -Forest

18 years, 5 months

[linux-audio-dev] [OT] No-one was ever fired for having hired FA

by Fons Adriaensen

Hello all, I finally took the big step and handed in my resignation at my employer, Alcatel Alenia Space. After 3 years of CAD, 3 years of real-time kernels, and 11 years in space telecoms, I want to return to my first love which is audio, acoustics, and music. My activities in LAD have certainly contributed to this desire. It will be at least end of september before I really say goodbye at AAS, and I have at this moment no idea at all which way I will go. Which means I'm open to suggestions. I will consider 'a real job' if it's related to acoustics, audio engineering, ambisonics, electro-acoustic music etc. For this type of thing I'm also prepared to move to France, Germany, and any of the European countries bordering the Mare Nostrum. I speak Portuguese (needs refreshing), I am currently learning Greek, and if necessary I'll add Spanish or Italian. I'm not really looking for a job as a programmer. The alternative is of course free-lance consultancy work. And if all else fails, I will be raising sheep on Crete. Anyone interested or having interesting pointers please contact me off-list. (and no, I'm not quitting LAD :) -- FA Follie! Follie! Delirio vano e' questo!

18 years, 5 months

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Linux-audio-dev July 2006