Linux-audio-user

linux-audio-user@lists.linuxaudio.org

3 participants
20662 discussions

by LM

I posted some links related to sound samples to the FreePats mailing list and Carlos suggested I share them here as well. Was recently researching a multimedia topic and happened to run across some sound sample related web sites. Thought I'd share what I found in case others might be interested. I found a cross-platform soundfont editor for the sf2 format. Looks like it's built using Qt. http://sourceforge.net/projects/polyphone/ Thousands of free sound samples specially recorded by Philharmonia Orchestra players licensed under a Creative Commons Attribution-ShareAlike 3.0 Unported License: http://www.philharmonia.co.uk/explore/make_music/ The LinuxSampler project was founded in 2002 with the goal of producing a free, streaming capable open source pure software audio sampler with high stability: http://www.linuxsampler.org/about.html This project aims to create a full orchestral sample library in the .gig format for use with the linuxsampler: http://sourceforge.net/projects/openorchestra/ List of sample libraries: http://bb.linuxsampler.org/viewtopic.php?f=8&t=11 musescore soundfonts http://musescore.org/en/handbook/soundfont http://www.linuxmusicians.com/ linuxmusician samples: http://www.linuxmusicians.com/viewforum.php?f=50&sid=bd65ce094deb26d5c09255… Recommended Sound Libraries and Where to Get Them... http://www.remastersys.com/forums/index.php?topic=1740.0 http://www.bandshed.net/sounds/sfz/ Web site to share Free music: http://freemusicpush.blogspot.com/ Best wishes. Laura http://www.distasis.com/recipes/music.htm

11 years, 8 months

Common Music - Importing audio files, and pointers to learning material?

by Angel de Vicente

Hi, I'm completely new to Common Music (http://sourceforge.net/projects/commonmusic/), so I'm hoping somebody can give me a hand here. Before I go to the effort of starting to learn it, I wanted to know if it is possible to somehow import audio files with it and treat them as objects (say for example that I have a number of noise sounds and that I want to generate some music by randomly playing them for a length of time)? A quick look at the Common Music webpage and some of the examples there didn't clear this up for me. At the same time, what would be the best way to start learning how to use it? From what I read, the book http://www.amazon.com/Notes-Metalevel-Introduction-Computer-Composition/dp/… is a very good starting point, but apparently all the examples, etc. are for an old version of Common Music (CM2), so I'm not sure if whatever one can learn from that book will be of much use with the current version of CM3. Any pointers welcome. Thanks a lot, -- Ángel de Vicente http://www.iac.es/galeria/angelv/ --------------------------------------------------------------------------------------------- ADVERTENCIA: Sobre la privacidad y cumplimiento de la Ley de Protecci�n de Datos, acceda a http://www.iac.es/disclaimer.php WARNING: For more information on privacy and fulfilment of the Law concerning the Protection of Data, consult http://www.iac.es/disclaimer.php?lang=en

11 years, 8 months

ps3 controller with bloetooth

by Atte

Hi I have "some success" with getting input from a ps3 controller over usb: This shows up in lsusb: atte@skagen:~$ lsusb | grep Sony Bus 003 Device 013: ID 054c:0268 Sony Corp. Batoh Device / PlayStation 3 Controller "cat /dev/input/js0" (as regular user) show the expected garbage when moving the controller around or touching buttons. I also get input in chuck opened with Hid.openJoystick(0), so I'm pretty sure the controller is recognized and sending stuff into the system. Now, how do I get it running over bluetooth (wireless). Google led me to lot's of outdated info, so I'm hoping someone here with hands-on experience on a recent system could provide a few starting points, hints, links or clues. Thanks in advance! -- Atte http://atte.dk http://modlys.dk

11 years, 8 months

[OT] Free Course online about the Higgs Boson

by Grekim Jennings

Hi dear all. Just wanted to let you know about this amazing course that the University of Edimburg has online during 7 weeks, and there's still days to sign in: https://www.futurelearn.com/courses/higgs As maybe you've notice, I'm quite eclectic and curious about anything, and Science (Physics in this case) is one of my favourite matters, so I'm in the course. It's now in the thick of the 2nd week and I tell you that it is amazing the quality and also the simplicity they've achieved given the complexity of the subjects, with a lot of instructional videos and some articles and texts, and even Mr. Higgs is there himself. Kindest Regards. -- Carlos sanchiavedraz * Musix GNU+Linux http://www.musix.es Thank you! This looks fascinating! Grekim

11 years, 9 months

Low latency audio interface for the Beagle Bone Black?

by Rafael Vega

Hi, has anyone tried a usb sound card with the BBB? I want to connect my guitar to it and run some PD patches so something with low latency would be niiice :) Also wondering how hard it would be to directly connect some ADCs and DACs. Has anyone tried? -- Rafael Vega email.rafa(a)gmail.com

11 years, 9 months

Re: [LAU] [LAD] JACK latency API clarifications

by Lieven Moors

On Fri, Feb 21, 2014 at 08:34:16PM +0100, Jörn Nettingsmeier wrote: > On 02/21/2014 07:52 PM, Lieven Moors wrote: > >>it was part of the API very early on, then we decided we didn't want to > >>impose the possibility of change on clients. as time goes on, it becomes > >>clear (to me at least) that we should have implemented it. > > > >What would be use cases for changing the sample rate dynamically? > > > having wired up a complex signal graph, which for the most part depends on > the studio, not on the project at hand, and then having to deal with > different projects in different sample rates. > > say your studio involves three monitoring setups, one main stereo, one > nearfield, and one surround, you are using jack to do EQ on those things, in > my case there's an ambisonic decoder in the loop as well. that means the > jack graph is already quite elaborated. in that case, it would be nice to > leave it running while switching from, say, a cd project at 44k1 to a tv > thing at 48k. > > as it is now, i have decided to do _everything_ at 48k (i have no second > thoughts about a final resampling step), but if a client brings material at, > say, 96k, i have to downsample first. sometimes i wish for an easy way to > reclock a graph. obviously, nobody expects this to be gapless. fading > everthing down and then taking a few seconds to reclock everything would be > fine. > > but then, many pieces of software in my chain would need changes. for > instance, an important piece of dsp for me is jconvolver, as it sits in > front of all my speakers. > of course, the impulse responses i use for EQ and room correction only make > sense for a given sample rate - it would have to be changed to swap one set > of IRs for another during a reclocking call, and of course that needs to be > configured and the user actually needs to provide those different IRs. > Yes, I see... I got into the habbit of using the same sample rate for all my projects as well. And I can remember a few times I wished to change the sample rate on the fly. Now I wonder if this would be difficult to implement. Do many clients expect the sample rate to remain stable? Aren't most clients checking for the sample rate in the process callback anyway? Of course, clients depending on samples or IR's would have to play back at the wrong rate... lieven

11 years, 9 months

[OT] Free Course online about the Higgs Boson

by Carlos sanchiavedraz

11 years, 9 months

Cutting vorbis files and resetting their times?

by Ken Restivo

I have a feeling I've asked this before, but I don't remember if there ever was an answer. I'm trying to do the following: 1) Cut an Ogg Vorbis file into chunks 2) Reset the start time of each chunk back to zero, and have its end time be the end time of the clip from zero. Thing #1 is very easy; oggz-chop does the job well. But thing #2 seems un-possible due to perhaps some design flaw in vorbis? The start times of vorbis files cut up with oggz-chop or similar tools is broken: it shows a start time of whatever was the time of the clip in the original file. This causes certain players (including Airtime) to lose control of their bladder: they either refuse to play the file or play silence for X number of hours until the start time of the clip. Yeah yeah, I know, I could just convert the file to WAV, then re-encode it. But... I do not want to do that. First of all, it reduces the quality. Secondly, there's something just upsetting my OCD nature, about not being able to do this without re-encoding. Any clues? I don't mind writing some C (or whatever) and wading through docs, if I had some expert advice on how to approach the problem (or at least confirmation that it is indeed possible). It's almost like I'd have to have something that reads the blocks one by one, then calculates the new time, and writes the block out with the new time? Is that a sensible way to do it? -ken

11 years, 9 months

Audio and Bluetooth

by Clifford Dunn

Hi List, Since I discovered that there is a great deal of interference happening when I have both wifi and USB audio running, it certainly has been a drastic improvement to turn that off when I'm working. I'm currently building a foot pedal controller. I have an Arduino Diecimila that is going to transmit the simple on/off as well as continuous controller info. My thought is to make it a bluetooth device and have wireless communication with my computer. Since I currently have no other bluetooth devices to test it out, does anyone have any information on how much bluetooth has the potential to interfere with my audio? Is it better to simply keep it as a bluetooth device? Thanks again! Cliff Clifford Dunn Flutist/Composer http://www.myspace.com/clifforddunn http://www.youtube.com/user/beatleboy07 https://www.soundcloud.com/clifford-dunn

11 years, 9 months

Jconvolver: Can't initialise engine.

by F. Silvain

Hey hey, with jconvolver I get exactly that warning: Can't initialise engine. I haven't found any means to get further output. I use jconvolver 0.9.2, just compiled again to make sure, that some updates didn't cause confusion. Compiler is gcc 4.8.2. Any advise is precious. Ta-ta ---- Ffanci * Internet: http://freeshell.de/~silvain

11 years, 9 months

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Linux-audio-user