Linux-audio-dev July 2003

linux-audio-dev@lists.linuxaudio.org

84 participants
89 discussions

[linux-audio-dev] Synchronizing sound to MIDI events?

by Denis de Leeuw Duarte

Hello, (I'm new to this list, so hi everyone!) I'm rather stuck on the following: I'm writing an app that uses JACK for its audio output. I now want to control this app using midi but I have trouble figuring out how to synchronize the rendered sound to the incoming events. The events, midi notes for example, come in with timestamps in one thread. Another thread (the one entered by process()) renders the audio. In order to render properly, it would need to calculate the exact sample at which the incoming note should begin to take effect in the rendered output stream. If you have an evenly spaced font, here's a graphical representation of the problem: |...e.....e|e....e....|...ee...e.|.....e.e.e|....e...e.| midi events |..........|...rrr....|.rr.......|......rrr.|....rrrr..| rendering |..........|..........|ssssssssss|ssssssssss|ssssssssss| sound Here, the e's represent midi events (but could be gui events just as well). The r's in the second bar represent the calls to the process function of my app. During this time, the audio that will be played back during the next cycle will be rendered. The s'es in the third bar represent the actual sound as it was rendered during the previous block. The vertical bars represent blocks of time equivalent to the buffer size. The best I can think of now is that I have to record midi events during the first block, process these into audio during the second block (because I want to take into account all events that occured during the first block) so it can be played back during the third. Now, all is fine, but time in the event-bar is measured in seconds and fractions thereof, but time in the third bar is measured in samples. How can I translate the time recorden in the events (seconds) to time in samples? How can I know at which exact time relative to the current playback time my process() method was called? If I just measure time at the start of my application I'm afraid things will drift. Is that correct? How have other people solved this problem? Hope somebody can help! Regards, Denis _________________________________________________________________ Help STOP SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail

19 years, 7 months

[linux-audio-dev] Call for participation: MMA GMPI working group

by RonKuper＠Cakewalk.com

The Generalized Music Plug-In Interface (GMPI) working group of the MIDI Manufacturer's Association (MMA) is seeking the input of music and audio software developers, to help define the technical requirements of GMPI. The objective of the GMPI working group is to create a unified cross-platform music plug-in interface. This new interface is hoped to provide an alternative choice to the multitude of plug-in interfaces that exist today. Among the many benefits of standardization are increased choice for customers, lower cost for music plug-in vendors and a secure future for valuable market-enabling technology. Like MIDI, GMPI will be license free and royalty free. Phase 1 of the GMPI working group's effort is to determine what is required of GMPI: What sorts of capabilities are needed to support existing products and customers? What are the emerging new directions that must be addressed? Phase 1 is open to any music software developer and is not limited to MMA members. It will last a minimum of three months, to be extended if deemed necessary by the MMA. Discussions will be held on an email reflector, with possible meetings at major industry gatherings such as AES, NAMM and Musik Messe. Following the collection of requirements in Phase 1, the members of the MMA will meet to discuss and evaluate proposals, in accordance with existing MMA procedures for developing standards. There will be one or more periods for public comment prior to adoption by MMA members. If you are a developer with a serious interest in the design of this specification, and are not currently a member of the MMA, we urge you to consider joining. Fees are not prohibitively high even for a small commercial developer. Your fees will pay for administration, legal fees and marketing. Please visit http://www.midi.org for more information about membership. To participate, please email gmpi-request(a)freelists.org with the word "subscribe" in the subject line. Please also provide your name, company name (if any) and a brief description of your personal or corporate domain of interest. We look forward to hearing from you. Sincerely, Ron Kuper GMPI Working Group Chair

19 years, 7 months

[linux-audio-dev] sc synth definition file format?

by Dr.Graef＠t-online.de

Hi, I'm currently embarking on a project to make an interface between Q, a functional programming language (http://www.musikwissenschaft.uni-mainz.de/~ag/q/), and SuperCollider. I think the OSC interface will be fairly straightforward to do, but I haven't been able to find any documentation (besides the sc sources, which I haven't grokked yet ;-) on the format of the synth definition file. Does anyone here know more about this? Many thanks in advance, Albert -- Dr. Albert Gr"af Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikwissenschaft.uni-mainz.de/~ag

19 years, 9 months

[linux-audio-dev] Denormal numbers

by Steve Harris

Several people have asked me what denormal numbers are over the last few weeks, well heres a much better description than my rambling head scratching: http://www.ecs.soton.ac.uk/~swh/denormal.ps Its an extract from David Goldberg's article, "What Every Computer Scientist Should Know about Floating-Point Arithmetic". - Steve

19 years, 9 months

[linux-audio-dev] Cool Linux-based DJing tool

by Frank Neumann

Hi list, I just found this nice link at the Debian Weekly News: "Debian as musical Instrument. James Patten and Ben Recht developed a composition and performance instrument for electronic music which tracks the positions of objects on a tabletop surface and converts their motion into music. One can pull sounds from a giant set of samples, cut between drum loops to create new beats, and apply digital processing all at the same time on the same table. Knoppix is used as operating system." Homepage is at http://web.media.mit.edu/~jpatten/audiopad/ which also has a very nice 20MB Quicktime .mov file explaining how it is controlled and how it works (mplayer can play this .mov without problems once you have the codecs installed). "Sweet!" :-) Greetings, Frank

19 years, 9 months

[linux-audio-dev] [ANN] hdspmixer-1.0

by Thomas Charbonnel

Hi, I'm pleased to announce the first public release of hdspmixer. Hdspmixer is a linux clone of Totalmix, a tool to control the advanced routing possibilities of the RME Hammerfall DSP cards. You can donwload hdspmixer here : http://www.undata.org/~thomas/ Thomas

19 years, 10 months

[linux-audio-dev] ANN: Csound Debian package

by Hans Fugal

Hello Debian users and interested bystanders, I have completed the Debian csound package. Once my sponsor has looked it over, approved of it, and uploaded it, it will be in unstable. In the meantime you can find it at deb http://hans.fugal.net/debian sid/main deb-src http://hans.fugal.net/debian sid/main I use csound, but I wouldn't consider myself a power user; in particular I haven't used any of the utilities (e.g. pvanal, hetro, etc.). If you do use them, please give the manpages a look-over and let me know if you see anything wrong. -- Hans Fugal | De gustibus non disputandum est. http://hans.fugal.net/ | Debian, vim, mutt, ruby, text, gpg http://gdmxml.fugal.net/ | WindowMaker, gaim, UTF-8, RISC, JS Bach --------------------------------------------------------------------- GnuPG Fingerprint: 6940 87C5 6610 567F 1E95 CB5E FC98 E8CD E0AA D460

19 years, 10 months

Re: [linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions

by Mickael S Vardo

I'm a professional audio engineer for 15 years, and DSD did convince me. I could compare the sound quality with TI ADC/DAC. Conventional PCM techniques are unable to reproduce high frequencies correctly. And the explanation is very simple. If you record a sound at 44.1 kss, you get a theorical frequency response of 0 - 22050 Hz. BUT to describe frequencies from 11050 to 22050 Hz, you can only play with a 4-sample long period. A 22050 Hz sine could be really accurate (one sample up, one sample down every 1/22050th second), and so is 11025. But intermediary frequencies introduces temporal aliasing, some metallic feeling due to temporal quantization. This is inherent to the very low sampling rate (96 kHz is just a bit better, but no miracle), which is unable to describe waveforms at high frequencies. Bad high frequencies temporal definition means bad transients. Anyone can notice it when he _actually_ hear and compare PCM and DSD. Stop speculative talking and try to get some real demo... -- mickael

19 years, 10 months

[linux-audio-dev] Redirecting the isdncard's output to the speakers and redirecting the soundinput from the mic to the isdncard.

by Jorge

Dear Sirs, dear Madams I've programmed an utility called kisdnmonitor, it tracks the calls you make/ receive. It's for KDE as the name suggests. There's a server as well, it's called isdnserver. You can get more infos on both on: http://www.elogix.ch/linux_en.html Now I would like to add telephony support to the programs, so that you can use the computer as a "deluxe" isdn phone. So far I've figured out that you can dial through the modem emulation /dev/ttyIx. You just properly configure the device and then with the following Hayes codes: ATS18=1 ATS14=4 ATD<the phone number> you can start a call. Everything is ok so far, the problem is that, obviously, no sound will automagically come out from the speakers and nothing that is brabbled through the mic will go to the other side. Basically, what I want to do, is to stream the sound from the isdncard to the speakers and from the mic to the isdncard. Vbox (a voice call programm) is able to save samples from the isdncard's input. I should mention as well that years ago, I had a program that came with the isdncard (for windoze) called RVScom that had the same functionality, and it worked (in walkie talkie mode, I guess that my isdncard isn't full duplex capable). Since 8 bits suffice, the processor load shouldn't be extreme. Has anyone an idea? Any help would be greatly appreciated. Thank you in advance Respectfully submitted George

19 years, 10 months

[linux-audio-dev] calling all Terratec EWS88MT / envy24 control users

by Fons Adriaensen

Hello list, I hope there are some other users of the Terratec EWS88MT card out there, because I seem to be in serious trouble - the card 'works' but not in a way that makes it useful. I've got a test program that outputs the same signal (1 kHz sine at -6 dB below peak level) to all eight channels. Measuring the output levels, I get wildly different values. Using envy24control to set the DAC output levels only seems to make things worse - there is no logical relation at all between the slider position and the actual level. For each of the four pairs of channels, the two level settings interact in a more or less random way, mostly by just switching the other channel off or on. After a few minutes twiddling, all sound stops until I reboot. Similar things seem to happen with the inputs, but I haven't had the time to investigate those. So I'm wondering what's wrong - is my new card not up to standards, or is there a problem with envy24control trying to control it ? Any help / hints / tips appreciated ! -- FA

19 years, 10 months

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Linux-audio-dev July 2003