Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7520 discussions

by holborn

Hi! New Release : horgand 1.0 News in v1.0 (19/06/2003) ------------- -Added small auto-accompaniment, Looped Drums and a Bass line in a wave table way. -added chord recognition -some free drum loops and bass samples. -bugs solved in DSP effects, now more stereo, minus noises :-) First i want to say something, personally i dont like any kind of auto-accompaniment, sequences, organs, because many people use it for work, and this is not good for the musiciens. In fact i wonder the day when the musiciens dont want to record anything, anymore. Only write the papers, and play live, that's music. Please read the Readme file for how to add new loops and other questions. REQUERIMENTS: * FAST COMPUTER * LINUX * ALSA * JACK * FLTK 1.1 Web Page : http://personal.telefonica.terra.es/web/soudfontcombi/ Josep

22 years, 6 months

[linux-audio-dev] Subharmonic synthesizer, is there one?

by Kimmo Sundqvist

Hello I'd like to know if there is a subharmonic synthesizer for linux. That is a plugin which adds low frequency content to sound not by boosting, but by synthesizing frequency components that are below what the signal otherwise contains. I'm thinking about something like Delaydots Phat Pro. If there is no such thing, is anyone thinking about developing one? -Kimmo S.

22 years, 6 months

[linux-audio-dev] Re: Tune for read (reiserfs)

by Andrew Burgess

>I have two linux computers with ext3 and reiserfs file systems >cat /proc/sys/vm/bdflush >echo "100 200 64 512 31 2000 50 1884 2" > /proc/sys/vm/bdflush >breaks again sounds little bit shorter >does somebody knows best paremeters >for comands >elvtune -r -w >echo "p1...pn" > /proc/sys/vm/bdflush This does it for me: echo 10 0 0 0 50 300 60 0 0 > /proc/sys/vm/bdflush for x in a b c d e f g h; do elvtune /dev/sd$x -r 512 -w 1024 elvtune /dev/hd$x -r 512 -w 1024 done I also renice the sound programs to -19 I don't use any special kernels yet.

22 years, 6 months

[linux-audio-dev] patches for kernel 2.4.21 ?

by Maarten de Boer

hello, it seems that the preemptible and the low latency patches are not yet available for the latest kernel, 2.4.21 has anybody succesfully applied older patches? any idea if the new patches are being worked on? maarten

22 years, 6 months

Re: [linux-audio-dev] patches for kernel 2.4.21 ?

by Karsten Wiese

look here: http://members.optusnet.com.au/ckolivas/kernel/

22 years, 6 months

[linux-audio-dev] Linux && dsp additions

by happyguy

So based on the feedback from the LAD community I'm guessing a Pulsar driver isn't forthcoming from Creamware. Too bad, nice hardware with a fairly well developed platform. Is there anyone with contacts inside Creamware that might be able to put me in touch with the right person to get something like this moving. Their support group went as far as to pretend they've never heard of Linux upon contacting them... but my guess is that not everyone may be singing the same tune. I'd like to see if A) creamware will release some specs for an ambitious individual such as myself to write a linux/alsa driver or B) creamware will sell their specs for OSS/GPL development (and hopefully there is enough interest out there to promptly donate some paypal funds to afford such a measure :) ) Is this avenue worth pursuing further, or should I look beyond a $1700 PwPulsar and find another DSP/DAW dev platform? Anyone have any leads? I ran into TCelectronics Powercore Firewire which seems decently assembled, but I do not know of any Linux efforts being made on this... obviously it is not supported directly. Of course everyone knows RME Hammerfall, but alas no DSP power :(. Still, I would like to give a little return investment to all those DSP apps the community has already written for Creamware's SCOPE platform... For those familiar with SCOPE, know of any other integrated platforms like this that have a good backing of community support? Can we live without a good DSP & 24x96? Not me. -s

22 years, 6 months

[linux-audio-dev] FREE Great Opportunity!

by Jonathan Paul Calledo

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22 years, 6 months

[linux-audio-dev] 8bit sound wav playing to a 16bit sound card...

by Derrick

I'm new to OSS Programming, and I'm attempting to play some 8bit wav files. However OSS is telling me that my sound card will not play 8bit , only 16bit. If I force it. The sound changes pitch, and is very fast. ( obviously ). Is there anyway to convert 8bit to 16bit on the fly? I've noticed that XMMS also fails to play the 8bit wav file correctly. I've even tryed to convert the file from 8bit to 16bit using SOX. But with the same results. I would like to support 8bit file wavs in my program as MOST of the wavs available are in 8bit format... Any one Have some pointers? PS: The command I used with sox is " sox -V -r 11025 -w -c 1 backup.wav temp.wav " -- >From the moment I picked your book up until I put it down I was convulsed with laughter. Some day I intend reading it. -- Groucho Marx, from "The Book of Insults"

22 years, 6 months

[linux-audio-dev] Q-Audio 1.0, Q-Midi 1.10 released

by Dr.Graef＠t-online.de

Q-Audio is a digital audio library for the Q programming language, which interfaces to Phil Burk's PortAudio and Erik de Castro Lopo's libsndfile libraries. Q-Midi is a MIDI interface for Q, built on top of Grame's MidiShare. Q is an interpreted functional programming language based on symbolic expression rewriting, providing a high-level interactive programming environment for scientific, computer music and other advanced applications. All software is distributed under the GPL. Sources, binaries and further information can be found on the Q homepage: http://www.musikwissenschaft.uni-mainz.de/~ag/q/ Enjoy! Albert Graef -- Dr. Albert Gr"af Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikwissenschaft.uni-mainz.de/~ag

22 years, 6 months

[linux-audio-dev] Re: [MusE] More ideas :) [Track freeze] - Alsa Sequencer

by Robert Jonsson

tisdagen den 10 juni 2003 13.21 skrev Frank van de Pol: > On Tue, Jun 10, 2003 at 08:30:39AM +0200, Robert Jonsson wrote: > > Hi, > > > > > In fact the bounce feature in MusE is "realtime". It means that you > > > have to wait the real duration of the track to be rendered. > > > In a non "realtime" mode the track is rendered as fast as computer can. > > > > AFAICT the realtimeness of the bounce feature is like that because of > > design constraints. Okay, bouncing wavetracks should be possible in > > non-realtime, but not when using softsynths. > > > > This is because all softsynths use alsa-sequencer as the input interface. > > And if I'm not missing anything, this interface is strictly realtime > > based. (perhaps it can be tweaked by timestamping every note and sending > > them in batches? it seems very hard though.) > > You are right, with the current alsa-sequencer the softsynths are driven by > realtime events. Though an application can enqueue the events to the > priority queues with delivery timestamp, the scheduling is handled > internally by the alsa sequencer. This causes some problems (especially for > sample accurate synchronisation with JACK or LADSPA synth plugins (XAP?)), > but also for network transparency and support for MIDI interfaces which > accepts timing hints (Steinberg LTB or Emagic AMT ... if specs of the > protocol were available :-( ). > > During the LAD meeting at Karlsruhe we discussed this and sketched a > alsa-sequencer roadmap that focusses on transition of the alsa-sequencer > from kernel to userspace and better integration with softsynths / JACK. > A few things from this are very much related to your track bouncing / > off-line rendering thing: > > - Provide facility to delegate scheduling to the client. The implementation > would be to deliver the events directly (without queuing) with the > timestamp attached to the registered client port. This would allow the > client to get the events before the deadline (time at which the event > should be played) and use that additional time to put the events at the > right sample position. > > Note that for the softsynth to get advantage of this the application > should enqueue the events (a bit) ahead of time and pass the timestamp. > Some of the current applications (including MusE) use the alsa-sequencer > only as event router and drive it real-time. > > Since the softsynth/plugin has no notion of the acutal time (only the > media time and sample position), rendering at arbitrary speed should be > possible: bounce faster than realtime or even slower than realtime for > those complex patches. > > - JACK is real-time, and bound to the sample rate of the soundcard. Since > the audio sample rate can also be used as a clock master for the alsa > sequencer this would be a good option to ensure synchronisation. The > transport of JACK and alsa sequencer can be tied together (either one of > the two acting as master, a run-time configurable option) to provide > uniform transport and media time amongst the applications that hook into > the JACK and/or alsa sequencer framework. > > For the offline rendering no nice scheme has been worked out yet; I guess > it would be something along the lines where the application that owns the > sequencer queue has full control on the transport, moving media time at the > speed the frames are actually rendered, and the app(s) generating the > events keeping at least one sample frame ahead of time. > > Frank. Okay, I didn't know that this had been up on the table, how far has this work progressed, was it just the Karlsruhe meeting or has more thinking occured? (fyi I'm CC:ing LAD, it might be a more appropriate place for this discussion..). Regards, Robert

22 years, 6 months

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Linux-audio-dev