Linux-audio-dev

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7513 discussions

[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, 6 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, 6 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, 6 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, 6 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, 6 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, 6 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, 6 months

[linux-audio-dev] IUPUI Student Radio Station should be based on Linux

by ben racher

Hello, I'm starting a student radio station at IUPUI in Indianapolis, Indiana and I want our entire audio infrastructure to be based on Linux. I've got a rough sense of all the apps we need and what apps to setup on which computers, but I thought I'd run the blueprints by you guys to see if you could give me any feedback. Streaming/Web Server: Runs apache and icecast or the icecast mod for Apache. Automation Computer: Runs some sort of playback program, I've been keeping my eyes on LiveSupport http://www.campware.org/ to schedule and automate the station when DJs aren't present. Audio Archive: File Server for our digital library, probably all FLAC files, maybe Ogg, but I think we want FLAC in case we want to burn CDs. And this is the part that I need help on... Production Computer... so I've been tooling around with JACK and Ardour and MusE (not to be confused with MuSE) and other JACK apps and its all really cool and exciting. I never got the sound input to even really work in linux until a couple weeks ago. Yay for the 2.6.8+ kernels. So here are my thoughts on setting up a workstation, and I don't even know if this is possible, but that's why I'm mailing you guys. One department has kindly donated a brand new Dell Poweredge Dual Xeon 2.4 ghz somethin or other. The rest of our computers are from the university junkyard of midgrade PowerPC G4s and Pentium 3s. So the Poweredge is our gem computer out of all the other crappy computers. Is there any way for me to set up the speedy new poweredge as some kind of audio production renderfarm, and get the PPCs and the Pentium 3s to connect to it as production terminals? Cause, although multi-tracking on the G4s and Pentium 3s is possible, doing extensive work with FX plugins is probably out of the question. See what I'm getting at? Also, the Poweredge also has about a 500gb raid system with it, which would be nice to use for storing our audio on and maybe even using as our digital archive as well, but that might be pushing it if we are doing audio production work on it as well? I'd imagine this might be the case, but I don't see why ftping flac files on a local network would be too much of a burden on the raid drive or dual processors. Another reason why it would be nice to be able to connect to a poweredge remotely to do audio work, is that it the poweredge makes about as much noise as a 747. So... its not exactly an audio production friendly unit. So these are my thoughts. Am I crazy... or is there some magical way to make this happen? - Ben Racher bracher(a)iupui.edu

20 years, 6 months

Re: [linux-audio-dev] Developing and audio driver

by fred doh

--- james(a)dis-dot-dat.net wrote: > On Thu, 10 Mar, 2005 at 07:42PM -0500, Dave > Robillard spake thus: > > On Wed, 2005-09-03 at 23:13 -0800, fred doh wrote: > > > Hi, > > > > > > I need to develop an audio driver (OSS on kernel > 2.4) > > > for a new hardware. I didn't find any resource > > > explaining how to do that, besides looking at > the > > > sources of other drivers. Could someone direct > me to > > > an appropriate resource? > No offence meant, but from your questions, you have > less of a clue > than I do, and I wouldn't fancy writing a driver > from scratch. > > Are you sure you're not biting off a little too much > here? > > On the othe hand, even if you give up half way > through, you'll > probably have learned a hell of a lot. > > But, as Dave says below, why? OSS? 2.4? Are there > good reasons for > this? > > Why on earth would you write an OSS driver for 2.4 > at this point in > > time? > > Writing for OSS is a constraint, not a choice. I'm not starting from scratch there is some older driver that I have to port, and it's been written for OSS. Later if there is time for it I will make the ALSA conversion. I've read docs about doing kernel drivers so my questions are not general about drivers, but really directed at audio drivers. I didn't find any resource about it. The OSS doc seems to be directed to the user of the API and not for the driver developer. For example, which function from the file operations structure have to be implemented and for what purpose in the case of audio? -fred __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

20 years, 6 months

[linux-audio-dev] Delta 1010/ALSA fluke

by Aaron Trumm

Hello. I wonder if anybody has any ideas on this one: I'm running a gentoo 2.4 kernel with a Delta 1010 sound card, wrote an asound.conf to route audio to various channels for multiple instances of mplayer - very cool - but this fluke happened this morning that it'd be awesome to understand... I got in and it had been looping three mplayer videos (mpgs) all night. They had gotten janky, ie: weren't really playing anymore, so I did a killall mplayer - nice. Then I tested some play back, one at a time, of the same movies. They routed fine, everything was great, except, on analog outputs 3, 5 and 7 (not 1), the audio was a little garbled. Like a digital garble type of thing almost sounding like a sample rate mismatch or something. I tested with an .mp3 and the same happened, and I tested an mp3 with alsaplayer and it happened, and then I did the tests again, and the second time, the .mp3 and the alsaplayer instance didn't do the weird thing, but the .mpgs still did it. So I rebooted it and it went away. *shrug* Anybody have any ideas on that? ----------------- Aaron Trumm www.nquit.com -----------------

20 years, 6 months

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Linux-audio-dev