Linux-audio-dev October 2003

linux-audio-dev@lists.linuxaudio.org

60 participants
67 discussions

[linux-audio-dev] Open source echo cancellation?

by Mark Rages

I am looking for a simple adaptive echo cancellation algorithm for a project I'm working on. Does anybody know of one under a GPL-compatible license? Preferably optimized for real time use? This might be found in a VoIP application, although I want it for musical purposes. I couldn't find anything at freshmeat.net. Then again, I can never find anything on that site. Regards, Mark markrages(a)mlug.missouri.edu -- To invent, you need a good imagination and a pile of junk. -Thomas Edison

22 years

[linux-audio-dev] ANNOUNCE: BEAST/BSE v0.5.5

by Tim Janik

BEAST/BSE version 0.5.5 is available for download at: ftp://beast.gtk.org/pub/beast/v0.5 or http://beast.gtk.org/beast-ftp/v0.5 BEAST (the Bedevilled Audio SysTem) is a graphical front-end to BSE (the Bedevilled Sound Engine), a library for music composition, audio synthesis, MIDI processing and sample manipulation. The project is hosted at: http://beast.gtk.org This new development series of BEAST comes with a lot of the internals redone, many new GUI features and a sound generation back-end separated from all GUI activities. The most outstanding new features are the demo song, the effect and instrument management abilities, the track editor which allowes for easy selection of synthesizers or samples as track sources, loop support in songs and unlimited Undo/Redo capabilities. Note, if you encounter problems with .bse files from previous BEAST versions, this may indicate bugs at the compatibility layer. A bug report accompanied by the problematic file can be send to the mailing list and is likely to get you a fixed file in return. Overview of Changes in BEAST/BSE 0.5.5: * New (or ported) modules: DavCanyonDelay - Canyon Echo by David A. Bartold BseMidiInput - Monophonic MIDI Keyboard input module BseBalance - Stereo panorama position module ArtsCompressor - Mono and stereo compressor [Stefan Westerfeld] * Added utility script to crop and duplicate parts [Stefan Westerfeld] * Added "Party Monster" demo song [Stefan Westerfeld] * Implemented ability to use sequencer as modulation source * Added support for external MIDI events in song tracks * Added .bse file playback facility to bsesh * Added support for C++ Plugins * Now installs bse-plugin-generator for simple creation of C++ Modules * Added manual pages for installed executables * Lots of small MIDI handling fixes * Fixed MP3 loader * Major GUI improvements * Registered MIME types for .bse files, provided .desktop file * Made search paths for various resources user configurable * Added prototype support to IDL compiler [Stefan Westerfeld] * Work around PTH poll() bug on NetBSD [Ben Collver, Tim Janik] * Support NetBSD sound device names [Ben Collver] * Added i18n infrastrukture for BEAST and BSE [Christian Neumair, Tim Janik] * Added Azerbaijani translation [Metin Amiroff] * Added Russian translation [Alexandre Prokoudine] * Added Serbian translation [Danilo Segan] * Added Swedish translation [Christian Rose] * Added German translation [Christian Neumair] * Added Czech translation [Miloslav Trmac] * Added Dutch translation [Vincent van Adrighem] * Lots of bug fixes --- ciaoTJ

22 years

[linux-audio-dev] Correct sustain pedal emulation (midi hold controller) ?

by Benno Senoner

Hi, I was wondering what's the correct way to handle the sustain pedal when implementing a MIDI sound generating module. from the MIDI specs: ----------- Hold Pedal, controller number: 64: When on, this holds (ie, sustains) notes that are playing, even if the musician releases the notes. (ie, The Note Off effect is postponed until the musician switches the Hold Pedal off). If a MultiTimbral device, then each Part usually has its own Hold Pedal setting. Note: When on, this also postpones any All Notes Off controller message on the same channel. Value Range: 0 (to 63) is off. 127 (to 64) is on. -------------- My question is about ".... holds (ie, sustains) notes that are playing, even if the musician releases the notes." Assume I play a chord, press the hold pedal, which causes the notes to be sustained. When I play new notes those are sustained too. So far so good. The question arises when I press the same key two times. Assume no sustain pedal for now. When I press C2 I hear the note. When I release it the sound does not vanish immediately but takes a small amout of time to decay due to the release envelope. If after releasing C2 I immediately press C2 again I hear two C2 notes for a brief time. Now same situation as above but with the sustain pedal pressed. You hear the first C2, release it (the corresponding note-off is postponed) and then press C2 again. In that case is it correct that you must hear two sustained C2 notes. Or must the first C2 be forced to get faded out / muted ? If not (eg you hear two sustained C2 notes), how far can this go ? Can there be 3, 4 etc sustained notes on the same key too ? While I am not a piano player,common sense says me thatpiano has only one string per key so IMHO it would sound unnatural to play two notes on the same key. As you might have guessed I ask this stuff because we want to add support of sustain in linuxsampler. thanks for your infos. PS: a new CVS repository for linuxsampler is up: cvs.linuxsampler.org interested developers and users please check it out and give us feedback via our mailing list. (subscription infos at http://www.linuxsampler.org ). cheers, Benno http://www.linuxsampler.org

22 years

[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

22 years

[linux-audio-dev] Roland CR80 - any good?

by Frank Neumann

Hi list, in a local music store in the "clearance corner" I found a Roland CR80 Rhythm Composer, built in 1991. It doesn't seem to have much similarity with the classic CR78, but at 50 Euro, it sounded like a nice bargain. Can anyone comment on it - sound quality, stability? MusicMachines etc don't have much info on it, and I even can't find samples to download anywhere.. Thanks, Frank

22 years

[linux-audio-dev] LADCCA without JACK ?

by "horsh

Hi, please excuse my stupidity, but: Why can't ladccad daemon start up without jack server being running? >From my point of view it would be right for ladccad to bring jack up. So allowing for example multiple jack sessions with different jack settings. horsh

22 years

[linux-audio-dev] Fw: Open letter to Steve Dekorte

by Erik de Castro Lopo

Hi all, I just thought some you you might be interested in how a small mnority of the MacOSX world sees the fruits of our hard work. Regards, Erik Begin forwarded message: Date: Sun, 26 Oct 2003 18:20:39 +1100 From: Erik de Castro Lopo <nospam(a)mega-nerd.com> To: undisclosed-recipients: ; Newsgroups: comp.sys.mac.apps,gnu.misc.discuss Subject: Open letter to Steve Dekorte Dear Mr Dekorte, This has been emailed directly to you as well as being posted to Usenet in the groups gnu.misc.discuss and comp.sys.mac.apps. I am writing to you in regard to your shareware application for MacOSX available here: http://www.dekorte.com/Software/OSX/SoundConverter/ Please also note that I am not charging you with contravening anyone's software license. I am however charging you with behaviour that is both morally repugnant and deceitful. When I download the tarball you provide (for which you charge US$10 for a full license) I find the following files: SoundConverter.app/Contents/Resources/ffmpeg SoundConverter.app/Contents/Resources/macconverter SoundConverter.app/Contents/Resources/mppdec SoundConverter.app/Contents/Resources/qt_export SoundConverter.app/Contents/Resources/ringtonetools SoundConverter.app/Contents/Resources/scm2wav SoundConverter.app/Contents/Resources/sndfile-convert SoundConverter.app/Contents/Resources/sox Here is some information about these programs: Program Size Author Licence -------------------------------------------------------------- ffmpeg 1.46M Fabrice Bellard LGPL macconverter 17k ? ? mppdec 85k Frank Klemm GPL qt_export 56k David Van Brink Lootware??? ringtonetools 94k Michael Kohn Non-comm. use only scm2wav 15k Christoph Leuzinger MIT License sndfile-convert 795k Erik de Castro Lopo GPL/LGPL sox 3.12M various LGPL Now compare this with the only part of this tarball actually written by you, the SoundConverter binary which weighs in at 399k. From the looks of this, your contribution to the total is significantly less than 10%. I would also argue that your contribution (a couple of hours with the MacOSX GUI builder) is far less that the amount of time and effort put in by the other people whose work you are using. How can you possibly justify pocketing US$10 per license for work to which you have contributed well less than 10% of the total time and effort. Furthermore, on the web site listed above, you display a credit for the guy who designed the icon, while none of the people who wrote the actual code get any credit whatsoever. Interestingly, many of the licenses above (GPL, LGPL, MIT etc) were developed to foster openness in the field of software development, much like the openness of scientific research. In scientific research circles, it is considered important to credit the people whose work yours builds on. In this case, you have failed miserably to do so. If you were a researcher you would be charged with academic misconduct and fraud. Now many people might think that you are just doing what Redhat, Suse and the other Linux distributors are doing; bundling up other people's software and selling it. However, I see a big difference. Redhat and Suse make huge contributions to the Free Software world; Redhat supporting GCC and GNU libc and Suse suporting KDE and ALSA. In light of all this, I am curious to know, what is your contribution? If you aren't contributing I suggest that you take all three of following three: a) Immediately, release the source code to SoundConverter under a suitable free license. b) Donate all the money you have collected so far to the Free Software Foundation or a recognised charity of your choice. c) Add some credits on your webpage to the people who did the vast majority of the work. I look forward to reading your response in one of the above public newsgroups. Regards, Erik -- +-----------------------------------------------------------+ Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid) +-----------------------------------------------------------+ "He who writes the code gets to choose his license, and nobody else gets to complain" -- Linus Torvalds

22 years

[linux-audio-dev] URGENT ALERT: Stop the "Broadcast Flag" at the FCC

by Seth Johnson

New Yorkers for Fair Use Action Alert: -------------------------------------- Please send a comment to the FCC AGAIN, opposing the "Broadcast Flag" Proposal Tell the FCC to Serve the Public, Not Hollywood! Okay, you folks understand this issue -- it's very important to send word to the FCC in the next few days, that you OPPOSE the Notice of Proposed Rulemaking #02-230. This rule would make it illegal for ordinary citizens to own fully functional digital television devices. We've made it easy; just follow the links below. 1) Please send in your comments to the FCC using the form provided below. Tell them that the movie industry should not have a special privilege to own fully-functional digital television devices. Read the alert below for details. 2) Please forward this alert to any other interested parties that you know of, who would understand and see the importance of this issue. 3) Volunteer to help us with this and other alerts related to your rights to flexible information technology in the future. Two roles you can take up are to become a Press Outreach Campaigner or a Commentator. Simply reply to this email to show your interest. New Yorkers for Fair Use Action Alert: -------------------------------------- Tell the FCC to Serve the Public, Not Hollywood! Send Public Comments to the FCC AGAIN to Stop the "Broadcast Flag" Please follow these links to let the FCC know that the public's rights are at stake: http://www.nyfairuse.org/action/fcc.flag/ What's Going On: The FCC is expected to decide this week that digital televisions will be required to work only according to the rules set by Hollywood, through the use of a "broadcast flag" assigned to digital TV broadcasts. As a result of the deliberations of a group called the Broadcast Protection Discussion Group, which has assiduously discounted the public's rights to use flexible information technology, Hollywood and leading technology players have devised a plan that would only allow "professionals" to have fully-functional devices for processing digital broadcast materials. Almost a year ago, you responded to our call to tell the FCC that they are to serve the public, not Hollywood. You delivered more than 4000 comments to the the FCC's public comments system in the space of the last week of their public comments period for the broadcast flag proposal. As a result of this, Congress took notice and called a hearing to question the FCC on the issue. When they asked the FCC's representative whether he believed they could make this copyright-related policy decision without stepping beyond their bounds and into Congress's jurisdiction, they answered in one word: "Yes." Now, their period of considering the proposal is drawing to an end, and they are expected to decide to mandate the broadcast flag in a matter of days, by the end of this month. It's time to demonstrate AGAIN that the public's interests take priority over the wishes of the MPAA. The idea of the broadcast flag is to implement universal content control and abolish the right of free citizens to own effective tools for employing digital content in useful ways. Hollywood and content producers must not be allowed to determine the rights of the public to use flexible information technology. The broadcast flag is theft. In the ongoing fight with old world content industries, the most essential rights and interests in a free society are those of the public. Free citizens are not mere consumers; they are not a separate group from so-called "professionals." The stakeholders in a truly just information policy in a free society are the public, not those who would reserve special rights to control public uses of information technology. Please let the FCC know that the public's rights are at stake: http://www.nyfairuse.org/action/fcc.flag/. Here is a page pulling together and summarizing the comments submitted after the last comments campaign: http://www.nyfairuse.org/bfpc/ Here is our Reply Comment: http://www.nyfairuse.org/bfpc/extdoc/NPRM%2002-231%20Reply%20Comments.pdf ---- The following link is the FCC's "Notice of Proposed Rulemaking" for the broadcast flag. http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-02-231A1.pdf

22 years

[linux-audio-dev] question about libalsaplayer's functionality (and its possible incorporation in my app)

by Ivica Bukvic

Hi all, I've tried contacting the maintainer of the project (Andy), but he never bothered to reply, so now I am taking this question to all of you out there that might have had exposure to this interesting library. Does anyone know what is the current status of the whole project anyhow? Any help on this matter is greatly appreciated! When answering any of the given questions, please include the question in your reply. Thank you very much! So here's the excerpt from the letter: I've recently decided to incorporate libalsaplayer-like functionality the upcoming version of my app RTMix. However, with having a particular feature in mind I am wondering whether libalsaplayer provides that and if not, whether you'd be willing to add it (or let me add it, although that might get messy since I am not too familiar with the inner workings of the lib). So, here it goes: Apart from all the wonderful features libalsaplayer offers, I am looking for some additional ones: I would like to be able to indefinitely loop specific "ranges" of a particular soundfile (i.e. 2.2secs-4.12secs or whatever). 1. Is it possible to do this and have music continually loop even if one changes the direction of playing so that when the alsaplayer runs out of the looping material in each direction that it just jumps back onto the other end of the loop and contiues on (kind of like a ring-buffer)? 2. Is it also possible to do this kind of looping on the whole soundfile (the gui version of alsaplayer always stops if I let it play 'til the end or the beginning, depending in which direction I am going). 3. Is it possible to define loop points by addressing a particular sample number rather than giving time in seconds? 4. How stable is alsaplayer when looping really small chunks of sound (i.e. like 5 sample loop)? 5. Is alsaplayer capable of ramping such loop points by attenuating let's say ending 20 samples (it would be cool if this number could be user-selectable) and then ramping up the beginning 20 samples in order to alleviate the "pop" that happens when looping a sound where waveforms at the beginning and the end do not align? 6. If the feature in question 5 does not exist is there a way to control output level of a player on a per-sample basis via callback so that one can implement that outside of the lib? 7. If neither 5 or 6 are possible, would you be willing to implement such functionality (i.e. a toggle_ramp( bool ); and set_ramp_length( int ); callbacks or something similar. I would greatly appreciate your feedback on this matter as that will greatly assist me in determining how to go about implementing such functionality in my app. Thank you very much! Looking forward to hearing from you. Sincerely, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico P.S. How do you implement reading of a sound faster and slower than it sounds in such a gradual fashion? Do you adjust the sampling rate of the DSP and if so does that affect other streams coming from the libalsaplayer, or is this something that is sound-specific (and if so, how)?

22 years

[linux-audio-dev] ardour 0.9beta6 release notes

by Paul Davis

Just a FYI. Those of you on ardour-dev will have parts of this list in various "latest CVS commit" messages. People using old versions of ardour should update. This new beta has lots of great stuff, with more to come. The new Mantis bug reporting/tracking system is making life way better for everyone. --p ---------------------------------------------------------------------- * additions/updates to AUTHORS * point people at ardour.sf.net/mantis for bug reporting * new region selection model - button1 press always selects - shift-button1 press adds an unselected region, unselects a selected region - more comfortable "selected region" color * new align_selection and align_relative commands * alignment now uses edit cursor * new commands to move playhead + edit cursor to region starts, ends and sync points * new zoom control interface, permitting arbitrary zoom levels (not just power-of-2) * new "add track/bus" dialog, allowing N to be added at once * mixer meters now support metering to +6dB, with new pixmap so that only >0dB is red. * initial implementation of 2-stage "clean up unusued audio files" * fix for transport lockup * fix for crashing bug when exporting at non-session frame rate * better functionality when editing AudioClocks * removed musical (BBT) time from Selection model * snap to region start/end/sync * snap to edit cursor * avoid neon when picking new colors * new transport pixmaps that are theme/skin resistant * tempo/measure lines now extend past end of session * hide editor mixer strip when clicking on current mixer track * brush mode works again ("paint" a region onto snap points) * added back "embed" functionality for referring to external soundfiles without importing * scroll playhead or edit cursors in the timeline rulers * check all X keyboard modifiers at startup, warn about duplicates * fixed crashing bug when marker was removed * allowed command-line specification of JACK client name * fixed mouse drags of torn off windows * code cleanup and redesign

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Linux-audio-dev October 2003