Linux-audio-dev

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7513 discussions

by James W. Morris

Hello people, I've just released Wav Composer Not Toilet 1.1001 on sourceforge. (wcnt.sourceforge.net) I wondered if anyone could compile it on different distros etc, as I'm not overly confident it will. The reason being I'm not sure I've got autoconf etc set up correctly, as when I create a new project in Anjuta, it will not autogenerate it. I've had to import the project from the last release and modify the makefile.am's so it will compile the new source files. It compiles alright on my system anyway. Just a side note, for those who remember the wcnt filters thread. I changed the filters so they now have a feedback input. I've linked together 3 lowpass filters, the first gets feedback from the last, and the second and third feedback into themselves, one of which has a negative feedback level. Also there is a range_limit module before the feedback is fedback which clips the signal. The result sounds to me like a cross between resonance and echo/reverb. I was pleasantly surprised. This can be found in the examples within the tarball. James. ~(sirromseventyfive)~ _________________________________________________________________ Find a cheaper internet access deal - choose one to suit you. http://www.msn.co.uk/internetaccess

21 years, 6 months

[linux-audio-dev] [ANN] TAP-plugins 0.4.0

by Tom Szilagyi

Hi all, TAP-plugins 0.4.0 is just released. Check it out at: http://tap-plugins.sf.net New plugins: * TAP Pitch Shifter Gives you the opportunity to change the pitch of individual tracks or full mixes, in the range of plus/minus one octave. Audio length (tempo) is not affected by this plugin, since audio is completely resampled. * TAP Rotary Speaker Simulates the sound of rotating speakers. Two pairs of rotating speakers are simulated, each pair fixed on a vertical axis, with their horns spreading the sound in opposite directions. * TAP Vibrato Modulates the pitch of its input signal with a low-frequency sinusoidal signal. It is useful for guitar and synth tracks, and it can also come handy if a strange effect is needed. Bugfixes: * Fixed crashing bug in TAP Reverberator (this bug appeared with hosts that call activate() before connect_port()). * Fixed lock-up bug in TAP DeEsser (the bug appeared when an input sample had +/- INF value). * Various smaller bugfixes (almost all plugins locked up when control input values of +/- INF were appiled). A recommended upgrade ;-) Enjoy, Tom

21 years, 7 months

[linux-audio-dev] Interesting MIDI project

by Erik de Castro Lopo

Hamster controlled MIDI: http://www.nbb.cornell.edu/neurobio/land/STUDENTPROJ/2002to2003/lil2/ Erik -- +-----------------------------------------------------------+ Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid) +-----------------------------------------------------------+ Microsoft : Yesterday's software running on today's hardware tomorrow.

21 years, 7 months

[linux-audio-dev] have to restart jack between ecasound sessions

by Eric Dantan Rzewnicki

I'm starting jack via jackstart like this: jackstart -v -R -d alsa -d ice1712 -r 44100 -p 64 -n 2 I can run ecasound in many ways (different chain settups, via ecaplay, wav or ogg input, etc.) one time only. If I try to run another ecasound session without restarting jack the second ecasound exits like this: audiobox:~$ ecaplay /mnt/audio/in_progress/guitar_mix/s2004-0217-2242.ogg *********************************************************************** * Message from libecasoundc: * * 'ECASOUND' environment variable not set. Using the default value * value 'ECASOUND=ecasound'. *********************************************************************** (ecaplay) Playing file '/mnt/audio/in_progress/guitar_mix/s2004-0217-2242.ogg'. exiting... It doesn't return ... just hangs like that until I ctrl-c it. If I strace the ecaplay it's waiting for ecasound: audiobox:~# strace -p 450 Process 450 attached - interrupt to quit waitpid(451, <unfinished ...> Process 450 detached which is stuck in a futex(): audiobox:~# strace -p 451 Process 451 attached - interrupt to quit futex(0x413e6bf8, FUTEX_WAIT, 453, NULL Once I kill the ecaplay w/ctrl-c the "ecasound -c -D" remains until I use kill -9 on it. Anything else I can do to help debug this? System details below. Thanks, Eric Rz. asus a7v8x-x athlon XP 2800+ (2071.203 MHz) 1.5GB PC2700 RAM 12GB / /dev/hda2 ext2 (actually a 40G disk) 160GB /mnt/audio/ /dev/hdc1 ext2 2GB swap /dev/hda1 (onboard via8235 -- disabled) ice1712 M-Audio Delta-66 w/omni i/o ymfpci guillemot maxisound fortissimo -- midi only debian testing (sarge) 2.6.2 (pre-empt on, drives tuned -- kernel.org sources compiled via debian's make-kpkg) realtime-0.0.2 lsm (insmod'ed since make install screws up all other modules) alsa-1.0.2 (lib, envy24control, tools) alsa-1.0.2c drivers: ./configure --with-isapnp=no --with-sequencer=yes --with-oss=no \ --with-cards=dummy,virmidi,ice1712,ymfpci,via82xx libsndfile-1.0.6 from tar.gz jack-0.94.0 ./configure --enable-capabilities --enable-optimize \ --with-default-tmpdir=/dev/shm --disable-portaudio ecasound-2.3.2 ./configure --enable-pyecasound --disable-oss --disable-arts \ --with-largefile ams-1.7.3

21 years, 7 months

[linux-audio-dev] [ANN] simsam-0.1.7 released

by Christian Henz

simsam-0.1.7 was released. Changes include: - multiple instruments - multiple outputs (JACK only) - config loading (at last) - some fixes & cleanups - some new bugs ;-) Source tarball and i386 .deb for Debian/unstable are available at simsam's SourceForge download page. http://sourceforge.net/project/showfiles.php?group_id=65022 http://simsam.sourceforge.net/ cheers, Christian Henz

21 years, 7 months

[linux-audio-dev] MIDI interface latency

by jrobbins＠nymu.net

Anybody know what's the minimum latency that can be achieved passing MIDI notes from one computer to another? I'm wondering if it's possible in theory to set up an audio generation cluster to be used as a realtime instrument. Basically, have a network aware synthesis app running on all machines, administer the setup/modification of the signal flow architecture from one master machine in non-rt over sockets, and then pass notes to the appropriate machines using midi interfaces. Naturally this depends on a signal path separable into big chunks. I have a hunch that midi over ip has latency too high and unpredictable for this. -jacob

21 years, 7 months

[linux-audio-dev] [ANN] Gungirl Sequencer 0.2.0

by Richard Spindler

Hi, Gungirl Sequencer is an easy to use Audiosequencer. It includes a simple Filemanager and uses Drag & Drop to arrange Audiosamples. This is the new Release 0.2.0 of Gungirl Sequencer, it comes with a bunch of new Features, and for your convinience is provided in the preferred standard Distribution Formats for both Linux and MS Windows: - Statically Linked (all dependencies included) RPM-Package - Autoconfified (./configure, make and make install) Source Package - Executable ( .exe) Installer for MS Windows The new features in this Release are: * A simple Ruler * Select Samples with rubber-frame * MiniPlayer for Pre-Listening * Customizable Snap-Value * Track Mute and Volume * Master Volume * Snapless Sample-Positing * Info-Box for Sample-Files * Progress-Bars for File loading * uncountable Bug-Fixes Check it out at: http://ggseq.sourceforge.net/ -Richard Spindler

21 years, 7 months

[linux-audio-dev] [ANN] Gungirl Sequencer 0.2.0

by Richard Spindler

21 years, 7 months

[linux-audio-dev] caps 0.1.0

by Tim Goetze

pleased to announce the initial release of the caps audio plugin suite under the GNU public license. quoting http://quitte.de/dsp/caps.html : caps, the C* Audio Plugin Suite, is a collection of refined LADSPA units including instrument amplifier emulation, stomp-box classics, versatile 'virtual analog' oscillators, fractal oscillation, reverb, equalization and others. some of caps is an improvement over previous efforts (the rewritten amplifier emulation plugins for example do 8x oversampling using polyphase filters, for much cleaner sound) but most of the plugins are ex nihilo creations. for those with an interest in DSP effects but no time or opportunity to run the plugins, the data sheets provided through the above document may be interesting. enjoy, tim

21 years, 7 months

Re: [linux-audio-dev] TAP Scaling Limiter - how's it work?

by Tom Szilagyi

Paul Winkler wrote: > Hi folks, and Tom if you're listening, > > The description of the TAP Scaling Limiter is > very interesting - http://tap-plugins.sourceforge.net/#limiter > I'm just curious, having done no real DSP coding - > it must do some internal buffering, right? Right, i have to admit it does :) That's why the plugin has to have latency. > So how does it deal with half-cycles that fall on the edge of a > buffer? It seems to me that you can't process the final > half-cycle without refilling the buffer, but you can't refill the > buffer until you've processed all its data - or can you? Yes you can. No, you can't actually, but you can eliminate the problem so you don't have to solve it. There is a ringbuffer, with a length chosen so it is capable of containing a whole half-cycle even at low frequencies (40 Hz is the limit in my implementation, which means varying number of samples at varying samplerates). Incoming audio is streamed into the ringbuffer (which actually works like a FIFO), and the half-cycles are being scanned starting from the other end of the buffer (that is, where output falls out). Now let's say the buffer is just full of audio. The host calls run(), and the plugin starts scanning backwards (from the oldest sample, that is, the one just about to fall out of the FIFO, towards the newest one), and marks zero-crosses and scales the marked individual zero-crosses. It keeps track of the half-cycle boundaries, and when it reaches (or more likely, overgoes) the N samples the plugin was asked to process in this run() call, it pushes out the N samples, but that of course will be in no relation with the half-cycle boundaries. Some piece of the last half-cycle will remain, and at the next run(), the plugin will continue the zero-cross scan from that point, not from the buffer boundary. But it counts the N samples needed to be pushed out from the buffer boundary, of course. I'm afraid i can't explain it very clearly, but i hope you get the feeling. Tom ps. stay tuned, more plugins are about to arrive... expect them in a week or two. :)

21 years, 7 months

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Linux-audio-dev