Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7516 discussions

by Fons Adriaensen

On Wed, Sep 10, 2008 at 05:02:12PM +0200, Kjetil S. Matheussen wrote: > jack_capture > ============ > jack_capture is a program for recording soundfiles with jack. Its default > operation is to capture whatever sound is going out to your speakers into > a file. (But it can do a number of other operations as well...) Very useful, just one remark: for more than 2 channels the default should be WAVEX instead of WAV (easy enough if you use libsndfile). WAVs with more than 2 channels are technically illegal and refused by some software. Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia Wie der Mond heute Nacht aussieht ! Ist es nicht ein seltsames Bild ?

17 years, 1 month

[OT] what are the good wireless headphones?

by Marcell Mars

i would like to have solution where i can use any headphones and where there is no bigger latency then 10ms (at least ;).... all pro solutions are so f* expensive....

17 years, 1 month

Getting the most out of onboard HDA

by Jens M Andreasen

Assuming a 7.1/192K/24bit $3:- chip onboard ... What next? First I will say, it actually will do the planned send and return to/from the two pieces of stereo outboard equipment I have (that is not going anywhere,) as well as do separate headphones and monitors. So I should be a happy camper ... Suppose one was to face the real world of playing live again? In my experience this can be a torturous venture into the realms of cheap lights and the evil thyristors and diacs of this world, all trying to make their voices heard through my equipment ... So, how about turning a single unbalanced stereo jack into a single balanced mono jack by converting a single mono signal to a stereo signal with inverted phase, would that be a good idea? If so, I believe this might work in a simalar way on microphone input as well. If the above holds and we then have two (instead of one) DAC's working push/pull on the line, would it be possible to take advantage of this? What I have in mind here is that, the lower you get in the "bittiness", the more the systenatic errors of distortion will be apparent. For starters, how about having one the phases at a level slightly below the other? This should ideally trigger transisitions between absolute levels at slightly different times for the two phases, giving us an extra 6dB of useable headroom. mvh // Jens M Andreasen

17 years, 1 month

Speaker Recognition

by Bluefuture

Hello all, I want to know if is there anybody that know anything about open source implementation of Speaker Recognition[1] or what kind of tool i can use to develop this kind of tool on linux?   Regards, Ste [1]http://en.wikipedia.org/wiki/Speaker_recognition -- Email.it, the professional e-mail, gratis per te: http://www.email.it/f Sponsor: Cerchi un hotel a Riccione, Rimini o Misano Adriatico ? Visita il sito rivieraparkhotels.it. Gli alberghi dei parchi divertimento Clicca qui: http://adv.email.it/cgi-bin/foclick.cgi?mid=8262&d=20080910

17 years, 1 month

Program execution in a program with good communications

by Julien Claassen

Hello everyone! This is again a bit OT, but still. I have a program with two pthreads. One pthread is running an audio player on a file repeatedly, the other is waiting for a signal. If "the other" gets the signal or timeout, then it should tell the player to stop. Unfortunitely with mplayer kill -15 doesn't have the desired effect. So I thought: What if I could make mplayer's stdin available to the program and simply send character 'q'. But because of pthreads, I left fork and exec* alone and went with system. Any ideas what to do? If someone knows something nice with fork and the like, I've seen something with pipes for this. Otherwise the player-command I call can be called from within a shell script, so I can do funny things with redirections. I hope I made myself clear and didn't write too much again... :-( Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

17 years, 1 month

[ANN] ProcessPP

by Florian Schmidt

This is totally alphabetagamma = unfinished and unpolished software. Release early - release often is my guide here ;) Here's the summary: ProcessPP is the "realisation" of a rather silly idea: A livecoding environment for C++. As C++ is not interpreted, but compiled, this indeed does sound like a really silly idea. But the twist is that while “true” livecoding cannot be done, it can be at least made as easy as possible to try out new code snippets. ProcessPP consists of three distinct parts - procespp: A host application that can load shared objects and execute the code contained in them - qprocesspp: A Qt GUI application which serves as an editor for hacking away functions and to compile them easily into a shared object. Then it tells processpp to load and run the shared object - libprocesspp: A shared library which is the “glue” between processpp and user generated shared object files. Right now it only containes code for the shared object to find out stuff like samplerate, buffersize, number of input/output channels, etc.. For illustrative purposes here are the steps involved in making noise: - File->Open from Template (choose process.template.cc) - Add some code in the inner loop to make noise - Process->Run - Hear the noise!!! If the above don’t work, take a look at the log in the lower half of the main window. Often some slight tweaks to the config file (~/.config/Ugh!/process++.conf) need to be made. Get it here: http://tapas.affenbande.org/wordpress/?page_id=91 Regards, Flo -- Palimm Palimm! http://tapas.affenbande.org

17 years, 1 month

[OT]: how to cast this damn pointer right?

by Julien Claassen

Hello! Sorry, for this, but my skills left me to unskilled... AGAIN... :-( I'm working with pthreads and now I like to hand over mains argv to the thread run_me. int main(int argc, char *argv[]); void *run_me(void *param); How to properly use param as a char * [] inside run_me? Thanks forany help! Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

17 years, 1 month

[ANN] jack_capture V0.9.30 and Rollendurcmesserzeitsammler V0.0.7

by Kjetil S. Matheussen

Download from http://www.notam02.no/arkiv/src/?M=D jack_capture ============ jack_capture is a program for recording soundfiles with jack. Its default operation is to capture whatever sound is going out to your speakers into a file. (But it can do a number of other operations as well...) Note: This version includes Hermann Meyer's jack_capture_gui2 program. "jack_capture_gui2" is a nice graphical frontend for for jack_capture with lots of options. Many thanks to Herman for the contribution. Changes 0.9.23 -> 0.9.30: *Added Hermann Meyer's jack_capture_gui2 program. jack_capture_gui2 is a nice graphical frontend for for jack_capture with lots of options. Many thanks to Herman for the contribution. *Don't exit in case port is not found. *Print runtime warning and error messages on top of the console to avoid printing the console meter yet another time. (it's much prettier also) *Fixed a bug that could cause (and especially after the switch from calloc to my_calloc apparantely) segfault when specifying --port more than once. Thanks to Peder Hedlund for spotting the bug. *Print error instead of segfaulting when a specified jack port does not exist. *Removed -g option and changed -O0 to -O2. (Oops, don't know how long that's been there) *Make sure the stop semaphore is initialized before it might be called. *Changed the --recording-time / -d option to record exactly the correct number of frames. (The format for the option is still in seconds though). This fixes the problem where the wall clock and the soundcard clock drifts apart. *Always increase the buffer size with 2 seconds when more than than half the buffer is used, unless maximum buffer size is reached. *Added the --maxbufsize / -MB option which sets maximum buffer size. Default value is 40 seconds. *Decreased the default buffer size from 20 to 10 seconds. *Changed internal data representation from lockless ringbuffer to lockless lifo and fifo stacks. Unmodified lifo/fifo code taken from midishare. (Copyright Grame 1999-2005) Rollendurchmesserzeitsammler v0.0.7 ------------------------------------ The Audio Rollendurchmesserzeitsammler is a conservative garbage collector especially made for running inside an audio DSP thread. 0.0.5 -> 0.0.7 * Cleaned up source a bit. * Fixed a bug in "tar_entering_audio_thread" which caused it to return false if currently copying a different heap. * Cleaned up the critical section handling between the DSP thread and the sweep thread. (it was really messy)

17 years, 1 month

Writing a library?

by Darren Landrum

I've been looking around for a library to read and write SFZ files, which is an open sampler format released by Cakewalk: http://www.cakewalk.com/DevXchange/sfz.asp Finding none, I thought I might try my hand at writing a library for this myself, as there is no embedded wave information like with Gig files. SFZ is simply a text file to be parsed. Now, I know about writing a good header file, and its associated class, and all that, but I have no knowledge of how to write it as a dynamic library. Google searches on every possible permutation have been worthless to me as well. I would prefer to write it in C++, as that's what I know, and even then, not too well, hence why I thought I'd start with something simple like parsing a text file. If anyone has any advice, recommendations, or ideas, I'll happily listen and learn. I have yet to think too much about how the data will be stored in the class, and what methods to make available to access it, so if anyone knows any best practices there, I'd really like to know. Consider this a feeler post. I'd ultimately want this for a future project, which you can guess at by now. Thank you for the help! Regards, Darren Landrum

17 years, 2 months

CUDA

by Jens M Andreasen

I am doing some preliminary testing of CUDA for audio, Version 2 (final) has been out for a couple of days, and this is also what I am using. In order to get anything done, one will always have to do something else first, here: Get some data transferred to the board. Surprisingly this appears to be ten times harder than getting the data back, at least for very small data sets representative of a bunch of on/off events or a milliseconds worth of samples. For 1024 bytes the transfer will take about 0.2 ms. That is 20% of the available time if we use a time granularity of a single millisecond. OTOH this appears to be mostly an intial constant, (much) more data can be transferred in the exact same amount of time if that is what is needed. Now my card (8400GS) is neither the latest nor greatest and I therefore wonder if anybody with better equipment is experiencing the same phenomenon? This card does not support aynchronous transfers, which otherwise might have been the thing to use here. mvh // Jens M Andreasen

17 years, 2 months

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Linux-audio-dev