Linux-audio-dev

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7440 discussions

[linux-audio-dev] LilyPond interview on linuxmusician.com

by Chris Cannam

A quick note to say that I've just put together an interview with Han-Wen Nienhuys and Jan Niewenhuizen, of the LilyPond project, and it's now at http://www.linuxmusician.com/ . The interview is pretty long and covers quite a few subjects, and I think is a very interesting read. I'm very grateful to Han-Wen and Jan for doing almost all of the hard work. Chris

20 years, 6 months

[linux-audio-dev] Glitchs using portaudio

by Ricardo Cirigliano

Hi! I´m using portaudio to capture continous voice using buffers of 200ms. I´m using 2 buffers in order to be able to make some process in one of them while I capture voice with the other one. However, every time I change the buffer that will receive the information a glitch occurs. Does anyone know how can I avoid this without using threads? Thanks!!!

20 years, 6 months

[linux-audio-dev] Hydrogen 0.8.2 is out!

by Comix

Hydrogen 0.8.2 is out! :) Features: __General__ * Very user-friendly, modular, fast and intuitive graphical interface based on QT 3 * Sample-based stereo audio engine, with import of sound samples in .wav, .au and .aiff formats __Sequencer and mixer__ * Pattern-based sequencer, with unlimited number of patterns and ability to chain patterns into a song * Up to 64 ticks per pattern with induvidual level per event and variable pattern length * 32 instrument tracks with volume, mute, solo, pan capabilities * Ability to import/export song files * Unique human velocity, human time and swing functions * Export to .wav * Jack-trasport support __Other__ * OSS and Jack audio drivers, with assignable Jack ports * ALSA MIDI input with assignable midi-in channel (1..16, ALL) * Import/export of drumkits * Midi-in record Changes: * Audio file preview in load instrument dialog * Jack transport improvements * 4 Ladspa FX send per instrument * Show recent used songs * QT Style selection option in Preferences Dialog * New keybindings * Bug fix in load/save song * LRDF support (optional) * Several GUI improvements * Better Midi-in support * virtual keyboard (using qwertyuiop...) * Ability to record midi-in or virtual keyboard notes in a pattern Download: http://hydrogen.sourceforge.net Happy drumming! ^_^ -- Alessandro <Comix> Cominu http://hydrogen.sf.net e-mail: comix(a)despammed.com Icq: 116354077 Linux User # 203765 [...Codito Ergo Sum...]

20 years, 6 months

[linux-audio-dev] Idea: fake OSS passback driver

by Daniel Risacher

I had an idea the other day, and I wanted to bounce it around for feedback before I invest any real time into it. While I have done some minor kernel hackng over the years, I am not experienced in the audio subject area. Background: Despite the various and sundry advances in linux audio, I find there are still legacy apps that are built against the OSS API. This is problematic since the legacy OSS model has blocking semantics. To get multiple audio streams, one must use an audio server such as esd, aRts, etc. Wouldn't it be nice if all the legacy apps "just worked"? Without blocking each other? Idea: Suppose one were to write a kernel module that implemented the OSS API, but had non-blocking semantics, and instead of driving a sound card, the module encapsulated the OSS API calls somehow and passed them back to a user-space audio server. Ao, for example: suppose we had a system with alsa and the proposed passback driver. xmms is playing via esd-passback-alsa, which is a modified version that supports input from the passback driver and outputs via alsa. The user fires up Ogle, which only has OSS output code (I'm not sure if this is true; just an example). /dev/dsp is the passback driver, and when ogle opens it, it succedes, even through someone else is using the sound card (esd). The passback driver "passes back" all of Ogle's ioctls to a modified version of esd, which, in turn multiplexes the audio with whatever else is playing (xmms, in this example), and plays it via alsa. I'm not precisely sure how I would to handle the actual "passing back". Probably a separate device file /dev/dsp-passback, which is read by the audio server. Can anyone say why this idea can not, or should not be implemented? Now that I've articulated the idea, would anyone care to take over the implementation?

20 years, 6 months

[linux-audio-dev] Cross platform mixer API?

by Eliot Blennerhassett

Greetings, I am aware of PortMixer, which is a simple API that abstracts volume controls. Is anyone aware of any other crossplatform, cross-API mixer abstraction layers out there? AudioScience is looking for/considering building a library that will wrap ALSA, audioscience HPI, windows WAVE, WDM mixers in a common API. If such a library doesn't exist, why is that? 1) Too hard 2) Too easy 3) No use 4) other? I'd welcome your comments. regards Eliot AudioScience Inc.

20 years, 6 months

[linux-audio-dev] [newbie] mixing audio streams

by Alex Marandon

Hello, I'm a trying to make a piece of code to mix 2 WAV files and hear the output on my headphones. I've spent hours trying differents things with no success and now I feel I should ask for some light from experienced people :) I've found explanations on how to mix audio streams on this page : http://www.vttoth.com/digimix.htm. But it doesn't really fits my own situation since samples I get with libsndfile are signed values. My piece of code works well for some files, but for others, it makes unpleasant clicks. I think I use a wrong algorithm, wich one do you use for such a situation ? Maybe I also use wrong data types, wich one should I use ? Is there any framework wich does such a task well ? I had a look at SndObj which seems to do such things but it's also quite old and not maintained against today configurations. Here is the guilty : #include <stdio.h> #include <ao/ao.h> #include <math.h> #include <fcntl.h> #include <string.h> #include <sys/types.h> #include <sndfile.h> #define BUFFER_SIZE 1024 #define DYNAMIC_RANGE 65535 int main( int argc, char **argv ) { /* variables pour libao */ ao_device * device; ao_sample_format format; int default_driver; /* variables pour libsndfile */ SNDFILE * wav1; SNDFILE * wav2; SF_INFO sfinfo; SF_INFO sfinfo2; /* float volumized */ static short buffer1 [BUFFER_SIZE]; static short buffer2 [BUFFER_SIZE]; static short mixed_buffer [BUFFER_SIZE]; int max_frames; int buffer_cpt; int number_of_buffers; int i; /* faut remplir la stucture sfinfo avec des zéros */ memset (&sfinfo, 0, sizeof (sfinfo)) ; /* -- Initialize -- */ if(!(wav1 = sf_open(argv[1], SFM_READ, &sfinfo))) exit(0); if(!(wav2 = sf_open(argv[2], SFM_READ, &sfinfo2))) exit(0); ao_initialize(); /* -- Setup for default driver -- */ default_driver = ao_default_driver_id(); format.bits = 16; format.channels = sfinfo.channels; format.rate = sfinfo.samplerate; format.byte_format = AO_FMT_LITTLE; /* -- Open driver -- */ device = ao_open_live(default_driver, &format, NULL /* no options */); if (device == NULL) { fprintf(stderr, "Error opening device.\n"); } /* la duree de jeu est en fonction de la longueur * du plus long fichier */ if ( sfinfo.frames > sfinfo2.frames ) { max_frames = sfinfo.frames; } else { max_frames = sfinfo2.frames; } number_of_buffers = max_frames / BUFFER_SIZE * 2; for ( buffer_cpt = 0; buffer_cpt <= number_of_buffers; buffer_cpt++ ) { /* -- lecture depuis les fichiers -- */ sf_read_short (wav1, buffer1, BUFFER_SIZE); sf_read_short (wav2, buffer2, BUFFER_SIZE); /* -- mixage, cf. http://www.vttoth.com/digimix.htm -- */ for ( i = 0; i <= BUFFER_SIZE; i++ ) { mixed_buffer[i] = buffer1[i] + buffer2[i] - buffer1[i] * buffer2[i] / DYNAMIC_RANGE; } /* -- envoi dans la carte son -- */ ao_play(device, (char *) mixed_buffer, sizeof(mixed_buffer)); } ao_close(device); ao_shutdown(); exit(0); } /* EOF */ Thanks :-) -- Alex

20 years, 6 months

Re: [linux-audio-dev] Re: [ANN] First public release of Lindrum v0.5.1

by peschler＠t-online.de

> From: Juhana Sadeharju (kouhia_AT_nic.funet.fi) > > Hmm... Linn Drum provided good drum samples, but Lindrum > has only an unexisting sound directory. Is that a good idea? Nope. Not a good idea. But providing the original Linn Drum samples wouldn't be a good idea either. As it was pointed out earlier the company is still in business and the drummachine will be renamed in the next release. So i will not package the original Linn Drum samples. But IANAL and maybe someone can enlighten me on this. How about samples from classics like the 808 or 909 (or Linn Drum)? Can i package these samples with my drummachine? I don't own these machines thus they are not sampled by myself instead i got them from the internet and other sources. PE -- "Without music, life would _O_/ \_O_/ +----------------------+ be a mistake - I would / )) [] | Peter Eschler | only believe in a god who \\ // | peschler(a)t-online.de | knew how to dance." (Nietzsche) // \\ +----------------------+

20 years, 6 months

[linux-audio-dev] Announcement: JackMix 0.0.1

by Arnold Krille

Hi all, I want to announce the first pre alpha release of JackMix. [What:] JackMix is a mixer for jack and my justification to be in this list. Currently it only supports 4 hardcoded inputs and 3 aux (mono) outs and 1 stereo out. But I hope to make this configurable in the next time... The outputs are organized in volumegroups, currently implemented is the aux group with a variable number of out and the stereo group with (guess it) 2 outs. JackMix uses parts of the kde-infrastructure (admin-dir) and Qt for the GUI. You need Qt and jack with their include-files. [Why:] This first release is for testing. Testing what others might think about such an idea. Testing if it builds on other pc's too. Testing my abilitites for getting criticism... [Where:] Download it from http://roederberg.dyndns.org/~arnold/file_share/?showdir=./jackmix Homepage (so far): http://roederberg.dyndns.org/~arnold/jackmix [Building:] At the present time you need to do a "make -f Makefile.cvs" after unpacking and before doing the usual "./configure && make". The configure-checks are probably not working the right way but they check for Qt and do some kind of checking for jack so I assume its enough for 0.0.1... [Future:] Planned features: - Dynamic adding and removing of inputs and outputs. - Saving and restoring the configuration. - More Volumegroups. For example groups with 3 or 5.1 outputs... - Effects using ladspa. Maybe this makes you interested. Thanks for testing JackMix, Arnold -- Get my public-key from pgp.mit.edu or pgp.uni-mainz.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...

20 years, 6 months

[linux-audio-dev] Re: [ANN] First public release of Lindrum v0.5.1

by Juhana Sadeharju

>From: peschler(a)t-online.de (Peter Eschler) > >Lindrum is an open-source drummachine that offers ten channels, realtime Hmm... Linn Drum provided good drum samples, but Lindrum has only an unexisting sound directory. Is that a good idea? Regards, Juhana

20 years, 6 months

[linux-audio-dev] [ANN] realtime-0.0.4 with 2.6.4 kernel support

by Jack O'Quin

The latest version of the realtime Linux Security Module is now available in the usual directory, <http://www.joq.us/realtime>. This version handles the new concurrent groups mechanism Linus introduced in 2.6.4. It still works with earlier 2.6 kernels. There are no functional changes. Unless you are running 2.6.4, there is no reason to upgrade. The realtime LSM is an installable kernel module that enables realtime capabilities for any 2.6.x kernel without needing to directly patch the kernel. It was written by Torben Hohn and Jack O'Quin, who make no warranty concerning the safety, security or even stability of your system when using it. It is provided under the provisions of the GPL. Thanks to Martin Habets for pointing out this kernel change and then showing me how to handle it. -- joq

20 years, 6 months

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Linux-audio-dev