Linux-audio-dev

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7513 discussions

the most kickass conference presentation i've ever seen...

by Jörn Nettingsmeier

from 27c3 :it's short, sweet, spectacular, and does away with the "sorry my laptop doesn't work with the projector" dilemma :-D http://www.youtube.com/watch?v=yEG68O6jpjo&feature=player_embedded

14 years, 8 months

interview with Dave Smith

by Dave Phillips

Greetings, http://www.kvraudio.com/interviews/interview_with_dave_smith Not very deep, but enjoyable. "Avoid Kansas". Great advice. :) Best, dp

14 years, 8 months

ALSA multi plugin and JACK xrun callbacks

by Lee Azzarello

Hello, I'm looking for some advice for an interesting symptom of putting my M-Audio Fast Track Pro USB sound card into ALSA multi mode. This sound card has four physical outputs which are mapped to two 2 channel ALSA streams rather than one four channel stream. I used the ALSA multi plugin to combine these two streams into a single pcm device so JACK can interface with all four outputs. It is working nicely. Despite no perceptible problems in sound quality or latency, jackd is printing hundreds of XRUN callback warnings to the message console each second! I can operate the system perfectly fine, run multiple applications, connect them to the four virtual outputs with no audible timing issues. What is the meaning of this many XRUN callbacks per second? What kind of performance will this impact if not sound playback? Thanks for your help, Lee Azzarello

14 years, 8 months

Re: [LAD] [LAU] [ANN] IR: LV2 Convolution Reverb

by Tom Szilagyi

On 13 January 2011 21:57, Darrin Thompson <darrinth(a)gmail.com> wrote: > > Dumb n00b question: > To use a convolution reverb one needs a recording of an impulse response > right? If I found a random one on the internet would it tend to work ok in > any reverb or do I need one made especially for the reverb implementation > I'm using? Technically, an impulse response is just an ordinary soundfile. Of course it has a certain soundfile format, sample format, channel count and samplerate, but if these are compatible with the convolution reverb you're loading it into (you can successfully load it into the plugin) then yes, it should generally work. Of course there are proprietary convolution reverbs that are bundled with impulse responses in a proprietary format, meaning that these impulses are not plain soundfiles, but are in some proprietary (and secret) binary format. Naturally, you cannot use those (and I guess if you could somehow reverse engineer the impulses into an "open" soundfile format, which would technically allow you to use the impulses in an open source plugin, the EULA of that plugin's license would have a word about it). Tom

14 years, 8 months

Linux Audio Conference 2011: Paper submission deadline extended

by Frank Neumann

Dear all, for the Linux Audio Conference 2011(*) in Maynooth, Ireland, we have extended the deadline for paper submissions to February 20th, 2011. So, if you have considered handing in a paper but couldn't make up your mind yet, here is your chance to become active! Never forget that this conference lives through the people participating in it. The deadline for music submissions has been set to the same date. Notification of acceptance of submitted papers has been aligned slightly as well, and will be sent out on March 15th, 2011. (*) http://lac.linuxaudio.org/2011 Please spread this information to anyone who might be interested. Direct any questions through email to lac(a)linuxaudio.org. Thanks, Frank

14 years, 8 months

pd-l2ork 20110109 snapshot now available

by Ivica Ico Bukvic

Please pardon cross-posting. Changelog: *fixed major CPU hit due to a bug in Cord Inspector implementation. Now the CPU usage is back to what it is in pd-extended. *further performance optimizations to the Cord Inspector. *improved edit mode detection and fixed edit menu activation accordingly. Latest snapshot is available from the usual place: http://l2ork.music.vt.edu/main/?page_id=56 Complete changelog since 11/25/2010 is available here: http://l2ork.music.vt.edu/data/pd/Changelog Best wishes, Ico

14 years, 8 months

Fwd: Devs needed for opensource virtual analog softsynth idea

by Sascha Schneider

Alsa Modular, is this one still under development?? I never have tested it and never saw a UI that I could use to controll it with external USB controllers. I'll check that .... regards, saschas 2011/1/2 Julien Claassen <julien(a)c-lab.de>: > Hello Sascha! > I'm not good at coding at all, but I think a more useable framework for a > softsynth, if you like to build it with an existing one, might be bristol. > Bristol is a synth emulator. It has a couple of synths already. But it might > not suffer, having a new filter or different oscillator in it, if Nick is OK > with that. The synths it emulates, are basically built from the components > (filters, oscs, etc.), that are in the engine. Then they are connected in a > particular way and get a GUI/CLI put on top of them. Bristol has, what I > would call MIDI learning. You can easily assing MIDI controls to controls of > the currently loaded synth and I think you can save them as well. Have a > look at his site: > http://bristol.sf.net > The sweet thing about using this would be, that you have to implement the > new components and then there is an API - so I believe - for relatively > easily constructing the connections and the <UIs. I know only of the textUI, > which is very clever and helpful! > Kindly yours > julien > > -------- > Music was my first love and it will be my last (John Miles) > > ======== FIND MY WEB-PROJECT AT: ======== > http://ltsb.sourceforge.net > the Linux TextBased Studio guide > ======= AND MY PERSONAL PAGES AT: ======= > http://www.juliencoder.de >

14 years, 8 months

pd-l2ork 20110105 snapshot now available

by Ivica Ico Bukvic

Please pardon cross-posting. Changelog: *fixed segfault when doing to-front/back without any object being selected *added enabling/disabling menu options when appropriate (e.g. cannot cut/paste if nothing is selected) *bound relevant canvas actions to the menu actions so that they can be invoked only if the menu actions are enabled (e.g. cut/copy/paste etc.) *requires further testing *relevant menu options reflect changes across all Pd windows (e.g. duplicate) *added special behavior for the cut/copy/paste menu in the root window *added previous window to the Window menu Latest snapshot is available from the usual place: http://l2ork.music.vt.edu/main/?page_id=56 Complete changelog since 11/25/2010 is available here: http://l2ork.music.vt.edu/data/pd/Changelog Best wishes, Ico

14 years, 8 months

usb tools ?

by Dave Phillips

Greetings, A fellow Fender Mustang 1 user is working on native Linux software that can be used with the amplifier. Apparently he needs some tools like usbsnoop to find out what's happening to & from the amp's USB port. He wrote that usbsnoop and snoopyusb weren't able to do the job, so I thought I'd ask if there are other similar tools that might be more up-to-date. Suggestions ? Best, dp

14 years, 8 months

Time-Stretch

by Harry Van Haaren

Hey all, I'm looking for an open-source time-stretching library, suitable for RT work. I've googled and come up with the following list, which I can't choose from: -Soundtouch : http://www.surina.net/soundtouch/index.html -ClearScale / DspDimension: http://www.clearscale.org/ -SecretRabbitCode / libsamplerate : http://www.mega-nerd.com/SRC/ -LibResample : https://ccrma.stanford.edu/~jos/resample/ -MFFM timescale: http://mffmtimescale.sourceforge.net/ -LibZita-Resampler: http://www.kokkinizita.net/linuxaudio/zita-resampler/resampler.html I'm attepmting to write a sample player, that will time-stretch long loops to stay in sync with JACK transport. I've deduced this is what I need to do: - Calculate the amount of samples I need to change the buffer length by - Take the buffer, resample it to a larger / smaller amount of samples. - Playback the samples as I normally would, except that there's more, and hence the audio will stay in sync. Design choices: 1. should I re-sample the whole buffer, and then playback from that buffer? this approach might cause a lot of CPU strain once the rate changes, as the *whole* buffer would be re-sampled at the same time. Once done, the CPU has no strain from resampling. 2. Resample "nframes" amount of samples from the buffer, and play them back? Less sudden CPU overhead, more constant CPU usage. The other problem I'm faced with is that some libraries mention that they do *not* support "dynamically variable resampling ratios". Would I need this? I think I do, as to get the beat "on-time" sometimes I'd need to add 200 samples, while in other cases 210 samples might suit better... I'm aware that Ardour uses SoundTouch, but I'm not sure is that the best lib for real-time use. Mixxx is using LibSamplerate AFAIK, and does so quite well, but might not be as suitable as Soundtouch.. http://www.surina.net/soundtouch/faq.html#sample_processing These kind of paragraphs are what stop me from just choosing one and going with it... Choices, Advice, Laughter, Help, etc all welcome! :-) Cheers, -Harry

14 years, 8 months

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Linux-audio-dev