Linux-audio-dev March 2005

linux-audio-dev@lists.linuxaudio.org

68 participants
63 discussions

[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

[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

[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

[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

[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

[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

[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

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

by David Cournapeau

Hi there, 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, and a recent news on /. about some demos at CeBit ). There also were some emails here about using GPU for sound synthesis recently, mainly about the shortcomings of those technics (memory bandwidth problems, difference of rate scales, if I understood corectly). Unfortunately, I know nothing about computer graphics, I don't know exactly what this ray tracing cpu technology is (maybe vaporware ?), but 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 ? Cheers, David

20 years

[linux-audio-dev] Lock-free gtk/jack interaction

by Olivier Guilyardi

Hi, This may be a general IPC question, but I feel lost so I thought you may help me as you did in the past for some other issue. In Jackbeat (written in C), I have the gui running in one thread (the gtk main loop) and jack in its thread. Currently, when tracks get updated, the main loop sends a message to the jack thread, and then waits for an acknowledgement. I currently use syscalls (message queues) for these messages, but that's not my point (I'll turn that later into syscall-free fifos). Waiting for an ack from the jack thread has a consequence : it locks the gui for a brief moment, which seems to drive my spin buttons mad : when clicked they will sometimes step the value by 1, sometimes 2, etc... Additionally, I think the gui should never lock, for confort. So I though about a "shadow" data structure. It would act as a intermediary area between my two threads. The jack thread runs into what I call a "sequence". When processing sound it relies on a structure called sequence_t. I'm about to add a new member to this structure, called shadow, which would actually be of the sequence_t type itself. This nested shadow structure would never be accessed by the jack thread. The public sequence functions could safely update this shadow, so that the gui can read from it anytime without any latency. Message queues would still be used to let the jack thread update itself the "real" sequence. This is where would reside some latency : waiting for the jack thread to get in sync. But the gui would never lock. I'm not sure this is a very clear description, but do you have any advice about this kind of issue ? Is my idea a good one ? Cheers -- og

20 years

[linux-audio-dev] Re: EMI26

by Andrew Burgess

>The firmware wasn't loaded. >> ok. this is the output of lsusb after I modprobed emi26: >> Bus 004 Device 002: ID 086a:0102 Emagic Soft-und Hardware GmbH >The emi26 module expects the device to have product ID 0100. It seems >you have a new hardware revision. >I'll see if I can update the emi26 driver. 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? Thanks! athlon:aab > lsusb|grep -i emagic Bus 001 Device 005: ID 086a:0102 Emagic Soft-und Hardware GmbH

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Linux-audio-dev March 2005