Linux-audio-dev

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7516 discussions

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, 7 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, 7 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, 7 months

[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, 7 months

[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, 7 months

[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

20 years, 7 months

[linux-audio-dev] Re: linux-audio-dev Digest, Vol 18, Issue 16

by Romain Beauxis

Le Dimanche 13 Mars 2005 18:04, linux-audio-dev-request(a)music.columbia.edu a écrit : > Subject: Re: [linux-audio-dev] ALSA OSS Emulation and emu10k1 > To: "The Linux Audio Developers' Mailing List" > <linux-audio-dev(a)music.columbia.edu> > Message-ID: <1110655551.6865.0.camel@mindpipe> > Content-Type: text/plain > > On Sat, 2005-03-12 at 19:51 +0100, Romain Beauxis wrote: > > How can I manage to use the skype device as an OSS device under skype? > > AIUI skype is closed source. Has anyone asked them when they plan to > add proper ALSA support? I think they surely will: http://www.skype.com/help/guides/soundsetup_linux.html Furthermore: http://195.38.3.142:6502/skype/ seems very interesting! Romain

20 years, 7 months

[linux-audio-dev] [ANN] A-weighting IIR filter

by Fons Adriaensen

Hi LAD, If ever you need a high precision A-weighting filter (as used for sound level metering), you can find one in the usual place: <http://users.skynet.be/solaris/linuxaudio> The tarball contains a C++ class implementing the filter (easily converted to C if you want that), and both a JACK in-process client and a LADPSA plugin using it. -- FA

20 years, 7 months

[linux-audio-dev] PADsynth (or how to generate beautiful sounds with longterm FFTs)

by Paul

I am Paul, the author of an opensource (GPL) software synthesizer for Linux and Windows (it's at: http://zynaddsubfx.sourceforge.net). I am writing this mail to you because I have seen your program (Mammuth) and the way how it process the sound, by using long-term ffts. I made a synthesis tehnique, that use longterm ffts (no windows) and is very intuitive, even for a musician. I implemented this ideea into my softsynth (as the "PADsynth" module) and it produces very good result and the ideea itself is very simple. To understand this and to use this intuitively, is very recomanded to read what I consider the "bandwidth of each harmonic" at http://www.kvraudio.com/forum/viewtopic.php?t=74129 . The base ideea of the bandwidth of each harmonic is that the harmonics of the sounds have larger bandwidths on higher frequencies. This happens in choirs, in detuned instruments, in vibratto,etc ; some ways to obtain the bandwidth are well known (but there are other ;-) ). Now, if I am considering the bandwith of each harmonic when I am using long term FFT synthesis, I will get a very beautiful sound and I can manage easily how is the enseble effect of the instrument. So the algorithm is very simple: 1) generate a long real array that contains the amplitude of the frequencies (it's graph looks like this http://zynaddsubfx.sourceforge.net/doc/paul1.png ) 2) convert this array to complex, by considering the phases random (this is the key: that phases are random) 3) do a long-term IFFT of the complex array 4) will result a perfectly looped sound that can be played at several pitches 5) enjoy of this beautifull sound obtained ;) As you see, this is very intuitively (even from musical perspective). Of course I made some variations of how I generate the array and I can make even non-musical sounds like noises, or metallic sounds. This is implemented in ZynAddSubFX and, because it's open source software, it can be studied (look in the src/Params/PADnoteParameters.C from the source code tree). Paul __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

20 years, 7 months

Re: [linux-audio-dev] ALSA OSS Emulation and emu10k1

by Romain Beauxis

Hi all! I managed to get half of the work with a custom .asoundrc file: "pcm.prear { type plug slave.pcm "rear" } pcm.dsnopp { ipc_key 1027 ipc_key_add_uid true type dsnoop slave.pcm "hw:0,0" } pcm.skype { type asym playback.pcm "prear" capture.pcm "dsnoop" } pcm.dsp1 { type plug slave.pcm "skype" } ctl.mixer1 { type hw card 0 }" Then I got an alsa pcm device that uses the Wave Surround as an output, mapped to /dev/sdp1 when running under aoss. I tried to record from the dsp1 with sound-record (oss recorder) and that worked fine! BUT: skype doesn't support aoss... :-/ I thought about lauching it with artsd but I need TWO devices: /dev/dsp(0) as the ring device and /dev/dsp1 as the phone device and artsd cannot handle two devices, as far as I know... How can I manage to use the skype device as an OSS device under skype? Then the second trick would be to set the Main volume controle from skype (and dsp1) as the Wave Surrount control. Couldn't find anythong on this, except for an OSS device, BUT I don't have any /proc/asound/oss card dsp1 since it emerge from aoss... :-/ What params can the ctl section use (I couldn't find any) Thx! Romain > Hi, > > I was wondering, is it possible to assign /dev/dspX devices to the > secondary and tertiary PCM devices on an emu10k1? > > I have a SB Live! Platinum with LiveDrive IR. The stereo out is connected > to my regular set of speakers. The surround output is connected to an > earphone headset, it's mic is connected to the mic input on my sound card. > > Also, another microphone is connected to the mic/line on the LIveDrive. > > Now, I use skype, which is closed source. It uses OSS devices and aoss will > not work with it. I would like to have a /dev/dspX device that records from > the mic input and plays back to the surround output, so that skype, and > skype only, will use the headset. > > The headset works fine in alsa mode and alsa apps can use it perfectly well. > > I tried all /dev/dspX and /dev/adspX devices, to no avail. I tries aoss > with .asoundrc modifications, no luck. I even read the driver source, but > I'm not really conversant with the structure of the driver and couldn't > find anything useful. It would take me forever to figure it out from the > source code. > > Is it possible, maybe with module parameters, to make alsa do this? Would > it need a patch, if yes, does someone have one? If no, what would have to > be done where to make that work? > > Melanie

20 years, 7 months

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Linux-audio-dev