Linux-audio-dev April 2011

linux-audio-dev@lists.linuxaudio.org

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Faust MIDI?

by Carlo Capocasa

Hi, is it possible to send and receive MIDI with a FAUST program? Carlo

12 years, 10 months

RDF libraries, was Re: [ANN] IR: LV2 Convolution Reverb

by Chris Cannam

On 23 February 2011 22:11, David Robillard <d(a)drobilla.net> wrote: > SLV2 is now based on two new libraries: Serd (RDF syntax) and Sord (RDF > store). Both are roughly 2 thousand lines of C, solid and thoroughly > tested (about 95% code coverage, like SLV2 itself). Serd has zero > dependencies, Sord depends only on Glib (for the time being, possibly > not in the future). Can you point me at the API or code? I couldn't see it in a quick browse on your SVN server. I have a library (Dataquay, http://code.breakfastquay.com/projects/dataquay -- preparing a 1.0 release of it at the moment, so if anyone wants to try it, go for the repository rather than the old releases) which provides a Qt4 wrapper for librdf and an object-RDF mapper. It's intended for applications whose developers like the idea of RDF as an abstract data model and Turtle as a syntax, but are not particularly interested in being scalable datastores or engaging in the linked data world. For my purposes, Dataquay using librdf is fine -- I can configure it so that bloat is not an issue (and hey! I'm using Qt already) and some optional extras are welcome. But I can see the appeal of a more limited, lightweight, or at least less configuration-dependent back-end. I've considered doing LV2 as a simple example case for Dataquay, but the thought of engaging in more flamewars about LV2 and GUIs is really what has put me off so far. In other words, I like the cut of your jib here. Chris

13 years, 6 months

[ANN] International Conference on Spatial Audio, Nov 10-13 2011, in Detmold, Germany

by Jörn Nettingsmeier

hi *! sorry for the slightly off-topic post, but since spatial audio has been a frequent topic lately, i think some people here might be interested. linux or FLOSS won't be exactly in the limelight, but yours truly will make sure there are at least 2-3 boxes with your favourite OS and audio tools humming along in various places. oh, and you might come early and watch a few high-end mixing consoles boot - the startup screen will bring tears to your eyes (as will the price tag, unfortunately :) unfortunately, there will have to be an admission fee, which we haven't decided on yet. but we're trying to keep it reasonable. don't shout at me when it turns out to be a bit more costly than LAC, though... jörn *.* ICSA 2011 - International Conference on Spatial Audio November 10 - 13, Hochschule für Musik, Detmold Organizers: Verband Deutscher Tonmeister (VDT), in cooperation with Deutsche Gesellschaft für Akustik e.V. (DEGA), and European Acoustics Association (EAA). Contact/Chair: Prof. Dr.-Ing. Malte Kob Erich-Thienhaus-Institut Neustadt 22, 52756 Detmold Mail: icsa2011attonmeister.de Phone: +49-(0)5231-975-644 Fax: +49-(0)5231-975-689 Summary: The International Conference on Spatial Audio 2011 takes place from November 10 to 13 at Detmold University of Music. This expert‘s summit will examine current systems for multichannel audio reproduction and complementing recording techniques, and discuss their respective strengths and weaknesses. Wavefield synthesis systems, a higher-order Ambisonics array, as well as 5.1/7.1 installations in diverse acoustic environments will be available for comparative listening tests during the conference. Structured plenary talks, paper and poster sessions will revisit fundamentals and present latest research. A series of workshops will be dedicated to practical implementations of spatial sound capture and playback methods, and their esthetic and psychoacoustical implications for music perception. Concerts that include music specially arranged for the conference will let you experience various spatial sound systems in "live" conditions. Call for papers and music: Your contributions are welcome, either as presentations, posters, or workshops. Submissions will undergo a review process, and accepted contributions will be published in the conference proceedings. The conference language is English. We are planning structured sessions on the following topics: * Multichannel stereo * Wave field synthesis * Higher-order Ambisonics / spherical acoustics * 3D systems * Binaural techniques An additional session will be dedicated to related miscellaneous contributions, such as hybrid systems and perception/evaluation of spatial music reproduction.

13 years, 7 months

Petri-Foo - a new fork of Specimen

by James Morris

Hi, I've forked Specimen primarily to provide frequency Modulation of the LFOs and to make all the LFOs and ADSRs independent so that there is no longer a single dedicated ADSR and a single dedicated LFO for ie pitch modulation, but two 'inputs' for pitch modulation for which the choice of all ADSRs and all LFOs is available. Please read the README for more information: https://github.com/jwm-art-net/Petri-Foo#readme The current state of Petri-Foo is that the LFOs and ADSRs have been made independant and are, AFAICT, working as should. The GUI is not yet up to date, but changes have been made enough to get a basic idea of what's going on. Please do read the README before commenting. I've tried to do things properly! I'm only human and only a hobbyist coder. Cheers, James. -- _ : http://jwm-art.net/ -audio/image/text/code/

13 years, 10 months

AVB applications

by Duncan Gray

This is to bring a discussion from the Jack Dev list to this more appropriate forum as suggested by Arnold Krille. First, I hear lots of people seemingly thinking that AVB (IEEE 1722) and the IEEE 1588 version of Precision Timing Protocol can be done in the kernel. It cannot and it must not. They both need hardware assist. Period. A timestamp is specified to be inserted based on the leading edge of the header immediately after the preamble. If anyone ever makes some neighboring equipment that has done these with the required precision, then you will kill that clock network and there will be yet another good reason not to bring Linux to the workplace. The ONLY exception is that if the listener is a stream-to-disk system, then the timestamp system can simply be ignored. Such a listener will never turn on PTP, but that won't hurt, because it will just ask for the 1722 stream and the talker will spit it out without knowing that that node doesn't play PTP. The version of PTP that is used in AVB is from the 802.1AS specification. The acronym PTP is now an ambiguous one that has at least these two uses, and I have heard some other hardware-assisted networked timing schemes called PTP. IEEE 1588 specifies an epoch-based struct with 48 bits of seconds (This gives 8.9 million years before a "y2k" hits IEEE 1588) and a 32-bit number that specifies nano-seconds. the 802.1AS sub-spec also uses this epoch-based timestamp. PTP maintains one suite of transactions to keep itself timed. This is blind to AVB. AVB creates Word Clock timeframes using the PTP wall-clock that MUST be made available to the 1722 layer. IF YOU HAVE PTP, then you can synthesize predictive wallclocks using a buffer-full scheme in a PTP-capable NIC. That NIC has to be configured to pay out the frames per the 802.1Qav forwarding and scheduling spec. This is how streamers will deliver streams that are well-timed, low-jitter streams. There are fruit companies doing this as we speak with new NICs that have been enabled from Broadcom and Marvell (and any host of others.) It is possible to fake a GrandMaster clock using kernel-timed calculations. The Best Master Clock Algorithm (BMCA) of a two-node system will be forced to accept such a sloppy clock and the slave will achieve lock, but with jitter that will fail a normally specified PTP system. Noisy environment listeners will not hear this, but clean listening will reveal the various artifacts of such jitter. You can just make a leaky-bucket PLL at a receiver and use the DPLL frequency to inform SRC. This hack will be un-noticed by the average media-player person, but not by the critical listener. When the 1722 timestamp is constructed, it is a complex assembly from the 802.1AS timestamp. The 802.1AS timestamp is a two-part thing, as specified above with its first part being simply seconds. This will not roll over in the lifetime of Linux, our species or even our continents, let alone a recording session. The second part is specified to roll over at decimal one billion-1 = 999999999 = 0x 3B9ACBFF. The timestamp in IEEE 1722 rolls over at unsigned long = 4294967295 = 0xFFFFFFFF, which is 4.294967296 seconds. I apologize for quoting "weeks between rollover" in the previous thread. IEEE 1588 and IEEE 1722 are Ethernet-Only protocols, do not shoe-horn them into IP. I have heard lots of people say that AVB is just some thing for consumers. go to http://grouper.ieee.org/groups/1722/ and hover over some of the names to find where they work. It was, in fact, designed FIRST to very easily accommodate Pro Audio: Reliability. Multiple-Node Synchronization without the need for Sample Rate Conversion. Low-Jitter. Unlimited (at least not limited by the protocol, only the bandwidth) channel count. And then it would be a trivial subset to get two - or 5.1, 7.1 any surround count - channels to go from my CD player to any media player over some LAN. (However, as of two autumns ago, they were still kvetching over Wi-Fi.) Finally, Yes, the CLOCK_REALTIME can be very simply pasted from a good PTP instance.

13 years, 10 months

FW: Frequency Space Editors in Linux

by Ivica Ico Bukvic

Hi Mike, Forwarding this to the list where users and developers might be able to assist. Hope this helps! Best wishes, Ico -----Original Message----- From: Michael Van [mailto:m_vanwagner@yahoo.com] Sent: Saturday, April 09, 2011 12:44 PM To: ico(a)linuxaudio.org Subject: Frequency Space Editors in Linux Hello Linux Audio, I just wanted to find out if anyone know of any Linux programs that do sophisticated noise removal from recordings, like the frequency space editing process of Windows programs, Adobe Audition or Cool Edit. I wondered if there is a plugin for Audacity that might do it. I need to use something other than standard noise sample removal plugins because the crackle is only present when the music is playing, not present during quiet stretches. Thanks, Mike

13 years, 11 months

Decode and Info about media

by Alfs Kurmis

Hello Experts. Is here library with C #include headers for so what ? How can i gotta info about Wav, mp3, ogg, m4a, wma [wmv] ... files SampleRate, Channels, PlayLen( in samples ), bytesPerSample , bitrate, VBR ... , and after i gotta this info , decode em to RAW stream ? At least i need everything about Wav, mp3, ogg. Info about sound file should not be printf()'ed to stdout , but must be filled in info structure. Decoder library should have functions - loadFile, Play, SeekToSample , Stop. At least possibility - play file from sample or second.part . Or should be used 2 libs, one for gotta info and one for decoding ? Any C example and pointer welcomed. Tnx in advance @ all. Alfs Kurmis ----

13 years, 11 months

LAC: Anyone driving to Dublin after the Linux Sound Night(s)?

by rosea.grammostola

Hi, I might be staying in Dublin during the LAC. The last train on friday and saturday is leaving on 23:10. The concerts might not be ended at that time. Is there someone who is driving from Maynooth to Dublin after the sound night and who can take someone with him? Regards, \r

13 years, 11 months

LAC Accomodation

by Harry Van Haaren

Hey guys, I was just looking at rates for acccomodation for the LAC, I was about to book a single person in the GlenRoyal hotel, but if somebody doesn't mind sharing with me and has booked I can chip in 1/2 the cost? I'm told I don't snore :) Cheers, -Harry

13 years, 11 months

Realtime Audio on Android

by Patrick Shirkey

Hi, I'm working on a cross platform mobile OS media player and doing some research into Realtime audio playback on Android at the moment. It seems there is a fairly small amount of info online and a lot less fully fledged code examples than available for the iPhone platform. For example I have already implemented my demo app in objective-C but it looks like it is going to be significantly more effort to do the same for Android. I have found a couple of interesting posts and articles about the current state things. Maybe people round here would like to comment on the situation? Here's what I have found so far: 1: There are three main API's - mediaPlayer, SoundPool, AutdioTrack with openSL ES in the works for release sometime in the near future. 2: All of the Android API's do not provide true low latency support and have varying bugs that need to be worked around with native code. The pd team have made some progress on the workarounds based on jack style callback wrapper classes. 3: There is no standard library for decoding audio file formats at the lower level. Although several options exist including native java implementations for mp3 (JLayer), and custom code from developers like Ivan Memruk from mindtherobot.com 4: There is no native midi or OSC support and it looks like there is no LV2 or ladspa port at this point either. There are very few open source demo apps available which provide a complete solution for media playback and recording at low latency. It seems that the pd implementation is the only one. Overview: http://www.wiseandroid.com/post/2010/07/13/Intro-to-the-three-Android-Audio… http://mindtherobot.com/blog/555/android-audio-problems-hidden-limitations-… <http://mindtherobot.com/blog/555/android-audio-problems-hidden-limitations-…>pensl-es/ <http://mindtherobot.com/blog/555/android-audio-problems-hidden-limitations-…> Sample code: http://mobiledeviceprogramming.blogspot.com/2010/10/fun-with-android-audiot… http://mindtherobot.com/blog/580/android-audio-play-a-wav-file-on-an-audiot… http://mindtherobot.com/blog/624/android-audio-play-an-mp3-file-on-an-audio… http://blog.pocketjourney.com/2008/04/04/tutorial-custom-media-streaming-fo… Workarounds / Middleware: http://gitorious.org/pdlib http://www.typhon4android.org/ Cheers. -- Patrick Shirkey Boost Hardware Ltd.

13 years, 11 months

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Linux-audio-dev April 2011