Linux-audio-dev

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7513 discussions

[linux-audio-dev] Re: Jack compatible LAME streamers (oddcast, ices2, darkice?)

by ben racher

So I messed around for a little bit and found something that kinda works. I was routing alsa-player in ardour (all this with jack of course), my sound card input also into ardour, mixing the two channels in ardour, and outputting the master out of ardour into oddcast, which then sent to icecast server, which I connected to with xmms, which sent the audio to soundcard output. And here was the trick. I increased the buffer size in Jack from 1024 to 4096 which consumed far less cpu resources therefore decreasing ringbufferfull errors from oddcast, and I also used jack.plumbing to do whatever it does between oddcast and ardour, which kept the stream from crashing. So now whenever I get the ringbuffer full error, the stream just jitters as long as the error persists, and seemed to return back to normal as soon as cpu resources became available again. So... sort of a solution. I'm hoping that in actual practice, I'll be able to divide some of these tasks between multiple machines, probably have oddcast running on its own machine, and pumping in an audio stream with jack.udp I'll try to go check on this later today, and will report back if its stable at all. Pretty impressive though that my g4 1.5ghz could handle all that. All this would probably work better on an intel system though? - Ben > I've been trying to get oddcast to work the past couple days, but keep > running into this problem > > ringbuffer full, tried to write 4096, but wrote 0 > > its usually in response to something distracting the processor for a > second, like dragging a window across the screen. Have tried on a 1.5ghz > ppc processor and 700mhz pIII and problem persists. Have searched for > other users reporting similar problems, and found one hint on the > oddsock site > > .... but unfortunately their site is in a period of transition, plug > above line into google and you'll find a couple similar complaints. > Apparently, the code for oddsock was scrapped from ices2-jack, which > would explain why I have no problems when trying to do this with Ogg, > but maybe Ogg is less processor intensive? I'm only having problems when > trying to encode with LAME. > > If anybody has any ideas, I would love to get this working for our radio > station. But if not oddcast, does anybody know if there is any other gnu > mp3 encoders out there (darkice?) that are jack compatible. Seems like > the darkice guy has been working on livesupport, which I can't wait to > see in action. > > - Ben Racher >

20 years, 6 months

[linux-audio-dev] Jack compatible LAME streamers (oddcast, ices2, darkice?)

by ben racher

I've been trying to get oddcast to work the past couple days, but keep running into this problem ringbuffer full, tried to write 4096, but wrote 0 its usually in response to something distracting the processor for a second, like dragging a window across the screen. Have tried on a 1.5ghz ppc processor and 700mhz pIII and problem persists. Have searched for other users reporting similar problems, and found one hint on the oddsock site .... but unfortunately their site is in a period of transition, plug above line into google and you'll find a couple similar complaints. Apparently, the code for oddsock was scrapped from ices2-jack, which would explain why I have no problems when trying to do this with Ogg, but maybe Ogg is less processor intensive? I'm only having problems when trying to encode with LAME. If anybody has any ideas, I would love to get this working for our radio station. But if not oddcast, does anybody know if there is any other gnu mp3 encoders out there (darkice?) that are jack compatible. Seems like the darkice guy has been working on livesupport, which I can't wait to see in action. - Ben Racher

20 years, 6 months

[linux-audio-dev] Fwd: [ardour-users] no sound when SC1 is activated

by Wolfgang Woehl

Steve, this is from ardour-users. I know the problem too: SC1 to SC4 more or less always stop working after some twiddling. Meaning the audio gets through only when SC* are bypassed or removed. I suppose you don't know the problem but maybe you can tell me what exactly you need to know to debug this. Wolfgang

20 years, 6 months

[linux-audio-dev] Amount of analog channels with nonuncompressing demux on ENVY24HT-S based cards

by Hans Davidson

Hello ! I want to setup a multichannel audio environment for use together with software that synthesize the waveforms for each channel. My newest audio card is a "Hollywood@home 7.1" card based on the ENVY24HT-S chip. I have managed to use the card to output mono and/or stereo sound under Slackware 10.1, but still want to find out if the card supports the conversion of uncompressed multichannel audio (for example 6 or 8 16-bit (or 24-bit) samples per frame from a FIFO through operating system and hardware) into analog signals ? If this is not possible with the internal DACs of the card, then I hope that it at least will be possible through ALSA with external DACs connected to the card's optical output, that seems to be of a high speed kind and expected to support ADAT (a standard established by Alesis, I think). I am not so interested in encoding the audio in DTS or some Dolby Multichannel format or simmilar, since i want the channels to be handled independently. For example I do not want to label a speaker as "left" or "right" and then select the speaker by assosiating a direction to the sound. Instead I want to be able to assign D/A channels to MIDI file tracks or MIDI channels and then placing the speakers there I want them (not there any multichannel decoder expect them to be). If I try to use several cards, then a lot of synchronization problems need to be solved, to avoid divergation in FIFO usage between cards and to maintain phase coherency. In addition to arrangements, like pathing cards to use a common chrystal oscilator and/or reserving some DAC-channels for synchronizaion purposes, I also need to convince the motherboard to accept several sound cards without causing problems during resource allocation (for example around IRQ mapping) or distributing the cards between several PCs connected together with maybe 100Mbps network cards. When I have solved these questions, then I may have to think more about the best algorithms to synthesize sound, by looking at existing open source programs and/or implementing additive synthesis with or without the use of FFT or some other optimized code. I am new to this list, so please excuse me if the answers to my questions are hidden in any recent messages in the archive. Thanks in advance for feedback/answers ! Hans Davidson _______________________________________________________ Skicka gratis SMS! http://www.passagen.se

20 years, 6 months

[linux-audio-dev] Linuxtag 2005 audio booth participation

by Christoph Eckert

Hi, first of all, sorry for the cross posting, but I thought the topic is worth it. Again, this year the Linuxtag will happen in Karlsruhe, Germany. It's Europe's most important event concerning free, especially Linux based software. During the last years, Frank Neumann has politely organized an audio booth there. Unfortunately, it can happen that he'll not be able to do all management stuff this year again. So, I'll try my very best to help out this year, otherwise we wouldn't have a booth. We're still looking for people who are interested in helping, mainly being at the booth, answering questions and demoing software. If you like to contribute, please let me know. For detailed information about the Linuxtag see http://www.linuxtag.org/2005/en/home.html or contact me via personal mail. Best regards ce

20 years, 6 months

[linux-audio-dev] Re: EMI26

by Andrew Burgess

>Andrew Burgess wrote: >> Mine comes up as 102, I've had to edit the driver source in the >> past to get it to work. Could you make sure 102 will work too? >The firmware for IDs 0100 and 0102 is indeed the same. >Editing drivers/usb/misc/emi26.c in the kernel source and replacing >0x0100 with 0x0102, and then recompiling the kernel, should enable it >to work. Yep, thats what I did. I posted the note hoping you'd make it work for both since you were going to look at the code anyway...

20 years, 6 months

[linux-audio-dev] Re: Using ray tracing cpu for audio synthesis

by Juhana Sadeharju

>From: David Cournapeau <cournape(a)enst.fr> > > I think I am not the only one here to have heard about so called '3d >ray tracing cpu' (e.g: >http://graphics.stanford.edu/papers/rtongfx/rtongfx.pdf Saarcor have earlier demonstrated their ray tracing graphics chips. One of the screenshots featured Quake game; ray traced version. http://graphics.cs.uni-sb.de People have also coded ray tracers on existing GPUs. >I wondered if this can be used for audio technology, e.g RT reverbs, >etc... Does anyone have an idea about the possible usage of this kind of >chips for audio processing ? If GPU cannot handle audio streams, it could be used to compute the early reflections for fairly complicated buildings. Only the reflection taps would be copied back from GPU. Both mirror and ray tracing methods could be used. The final dense reverberation should still be computed if mirror and ray tracing generates only approximation, but that is easier with a lot of reflection taps. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

20 years, 6 months

[linux-audio-dev] [patch] terminatorX

by Antonio "Willy" Malara

if anyone is interested here is a really trivial patch to terminatorX's tX_mouse.cc file: it allows to trigger the turntables with the numeric keypad (or you can just tweak the switch cases :) without having to focus the specific turntable first during "grab mode" i attach it here, its shorter than some signatures i've seen in some other mailing lists :))) ciao a tutti :) Willy / BeHappy_

20 years, 6 months

[linux-audio-dev] HDCD

by Lee Revell

All this (depressing) talk of the various encumbered formats competing to be the next CD standard reminded me of HDCD. What's the deal with it? AFAICT it's another closed thing that we'll never be able to support properly. The only media player that detects my HDCDs is WMP of course. Lee

20 years, 6 months

[linux-audio-dev] Re: Any interest in DVD-Audio?

by Juhana Sadeharju

>From: Alfons Adriaensen <fons.adriaensen(a)alcatel.be> > >The only limitation of not using MLP would be that you can't have six >channels at 24/96. Why not? Flac + open DVD-audio format for N channels. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

20 years, 6 months

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Linux-audio-dev