Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7513 discussions

Re: [linux-audio-dev] Linux audio presentation, 17.3. at Otaniemi,Finland

by Taybin Rutkin

What is a GPOS? Taybin -----Original Message----- From: Kai Vehmanen <kai.vehmanen(a)wakkanet.fi> Sent: Mar 16, 2004 4:13 PM To: linux-audio-user(a)music.columbia.edu, linux-audio-dev(a)music.columbia.edu Subject: [linux-audio-dev] Linux audio presentation, 17.3. at Otaniemi, Finland Hello all, this comes a bit late, but in case you happen to be somewhere in South Finland tomorrow... I'll be giving a short presentation on Linux audio tomorrow at Otaniemi (hut.fi) at 14.00 [1]. I'll talk a little about general issues of developing apps for GPOSes, introduce ALSA, JACK and LADSPA, and finally show a quick demo of well-known Linux audio apps (full PlanetCCRMA setup). The presentation is part of the "Digital Signal Processors and Audio Signal Processing" course [2] at HUT, but everyone is welcome to come and listen (and ask tricky questions ;)). The presentation is in Finnish, but I can translate on-the-fly if needed. :) [1] Auditorium S1, Otakaari 5 http://www.hut.fi/Current/getting_to_otaniemi.html http://www.hut.fi/Current/otaniemi_map.html [2] http://www.acoustics.hut.fi/teaching/S-89.510/ -- http://www.eca.cx Audio software for Linux!

21 years, 6 months

[linux-audio-dev] Linux audio presentation, 17.3. at Otaniemi, Finland

by Kai Vehmanen

Hello all, this comes a bit late, but in case you happen to be somewhere in South Finland tomorrow... I'll be giving a short presentation on Linux audio tomorrow at Otaniemi (hut.fi) at 14.00 [1]. I'll talk a little about general issues of developing apps for GPOSes, introduce ALSA, JACK and LADSPA, and finally show a quick demo of well-known Linux audio apps (full PlanetCCRMA setup). The presentation is part of the "Digital Signal Processors and Audio Signal Processing" course [2] at HUT, but everyone is welcome to come and listen (and ask tricky questions ;)). The presentation is in Finnish, but I can translate on-the-fly if needed. :) [1] Auditorium S1, Otakaari 5 http://www.hut.fi/Current/getting_to_otaniemi.html http://www.hut.fi/Current/otaniemi_map.html [2] http://www.acoustics.hut.fi/teaching/S-89.510/ -- http://www.eca.cx Audio software for Linux!

21 years, 6 months

[linux-audio-dev] ICMC 2004 paper/demos/poster submission extension

by tae hong park

Please pardon cross postings. ICMC 2004 : The online submissions deadline for papers, demos, and posters have been EXTENDED to Midnight EST Friday 19, 2004 and are available at http://www.icmc2004.org. Please note that full paper, camera-ready submission is required. Thank you. Best Regards, Tae Hong Park, ICMC 2004 Publicity Chair

21 years, 6 months

[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

21 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!!!

21 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...]

21 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?

21 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.

21 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

21 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) // \\ +----------------------+

21 years, 6 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev