Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7520 discussions

[linux-audio-dev] Softsynth latency control [Jack?]

by Florian Schmidt

Hi, i was thinking about latencies of softsynths and how cubase vst handles this when playing back recorded midi tracks as opposed to playing the softsynth directly. I'm talking about the vst-softsynths here. not the ones that get controlled via usual midi ports. It seems that during playback of a prerecorded midi track cubase vst knows the latency of the softsynth and and arranges for that so that the softsynth is really tight [no offset to other recorded audio material]. This is, of course, only possible because the midi signal if routed vst-internally is handled different than midi that gets sent out. I suppose that todays available audio/midi-sequencer support midi in a form that it is synchronized to the latency of the audio signal that is sent out. So, if the audio signal takes 2*128/44100s [2 buffers a 128 frames] to reach the ear of the user, then the prerecorded midi signal is delayed this exact amount of time, because midi signals is supposed to be of zero latency. And for external hardware synthesizers this is pretty much true. This way prerecorded audio and midi tracks are nicly synchronised during playback. Now, for the case of a sofstsynth this scenario doesn't fit, because the synth is not zero latency. This means that a "tight" midi track is audible with the softsynths output latency. Now, Jack is really an audio server but it could also be used to communicate the latency of the softsynth to the Audio/Midi Sequencer. I know there's calls to get the output latency of physical output ports. But it would be nice to have calls to ask for the physical output latency of clients [this value can be different for different clients - maybe with different output devices]. This way, the audio/midi-sequencer could ask jack for a list of clients and offer this choice to the user who then can select which midi tracks correspond to which softsynth. The Sequencer could then send the midi events a tad bit earlier [the output latency of the softsynth]. What do you think? I have no insight into the implementation details of jack, so i don't know how possible this scenario is and if it fits with design decisions. Regards Florian Schmidt

22 years, 1 month

[linux-audio-dev] CheeseTracker 0.9.0 released!

by Juan Linietsky

Cheesetracker is a portable Impulse Tracker clone. It supports all the main Impulse Tracker and FastTracker/SoundTracker features, plus many more. It is licensed under the GNU Public License. It runs under Linux/BSDs, MacOSX(QT/Mac or Qt/X11), or Win32(Cygwin). It can be obtained at http://cheesetronic.sf.net For those unfamiliar with trackers, this is basically an all-in-one sequencer/sampler/sample editor/mixer/fx processor bundle, wich provides fast and flexible means for professional grade music composing. Including in the release are tutorials and documentation. Volunteers to help to better documment it would be highly appreciated. The ChangeLog for this release follows: v0.9.0 ------ -Removed sample mode (Scream Tracker 3 mode) as It's obsolete and not needed for backwards compatibility. -Instruments are now layered and can perform up to 4 simultaneous voices with individual parameters each. -Added an effect buffers system. Instruments are now routed to custom buffers (each with individual effect chains), which can also re-route to other buffers. This allows to create very complex effect routes for realtime processing. -Effect buffers are "process on demand", which means they are smart enough to notice when they are doing nothing, thus disabling themselves. -Added a few internal effects: Amplifier, Clipping/Distortion, Recursive Delay Line, Stereo Enhancer, Chorus and Reverb. -Added a LADSPA effect source plugin. LADSPA plugins can be added to the chains. -Created new file formats that save all the new features: .CT (CheeseTracker Module) .CI (Cheesetracker Instrument) and .CS (CheeseTracker Sample) -Added preview to the sample file selection box, just hilite a file and use your keyboard to play notes (/ and * work in there too). -Readded JACK Driver (Kasper Souren) -Added RTAUDIO driver, allows for porting to Win32/ASIO and OSX/CoreAudio -Fixed some big endian compatibility issues. CheeseTracker should work fine again on big endian machines. -MacOSX port and build system/build fixes courtesy of Benjamin Reed -Fixed tons and tons of bugs. AND NOW PLEASE READ: How fast CheeseTracker reaches version 1.0 depends on YOU. The focus of this version is STABILITY. Because of this, I need to receive as many bug reports as I can, both program and build system. If you find a bug, I'd be enormously grateful if you submit it. Even if it is an obvious bug to you, chances that other people will find and report the same bug are much smaller than what you may think. If you dont report a bug that annoys you, the chances of it reappearing in the next version will allways be higher. Planned for 1.0.0: -=-=-=-=-=-=-=-=- -Rock Solid stability -WAV exporting -A hopefully working Windows port. This depends mainly on the Qt-Win32 project. If you are a good Windows programmer and would like to see CheeseTracker working in there sooner, please give those guys a hand! Enjoy!! Juan Linietsky

22 years, 2 months

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

by vanDongen/Gilcher

On Monday 27 October 2003 15:08, Benno Senoner wrote: >Assume I press C2 with velocity 50 pedal up, the C2-pedalup (associated >to velocity 50) sample sounds. >Now I press the sustain pedal and press C2 with velocity 100. >What should the sampler do ? Quickly fade out the C2-pedalup >(velocity-50) note and trigger the >C2-pedaldown (velocity 100) note ? >And of course when you release the pedal all sustained notes will get a >note-off. I am a piano player. I would expect the C2 note to be replaced with the next attack on that note. If the pedal is pressed, the damper stays up, so the next attack will be the hammer hitting the strings, and the sounding note is not faded out before. So I think you should fade out the previous note after or on the attack of the next one. Maybe even a lot later. There is an adding resonance effect if you keep hitting the same note with the pedal down, but that is probably the sympathetic vibrations from the other strings. If the pedal is not pressed the damper returns to the string as the key is released to play the next attack. You will get a note off message then anyway. Note that releasing the pedal should only send note-off to those notes that are sounding, but whose key is not pressed. If you have keys pressed down, the dampers will be up, regardless of the position of the sustain-pedal. So these should continue. I have no idea if digital piano's actually do this correctly, but it is the way a accoustic piano works. Gerard electronic & acoustic musics-- http://www.xs4all.nl/~gml

22 years, 2 months

[linux-audio-dev] LADCCA mailing list

by Bob Ham

Hi all, There's now a mailing list specifically for LADCCA discussion, ladcca-devel. You can subscribe from here: http://mail.nongnu.org/mailman/listinfo/ladcca-devel Bob -- Bob Ham <rah(a)bash.sh> "At some point, keystroke recorders got installed on several machines at Valve. Our speculation is that these were done via a buffer overflow in Outlook's preview pane." -- Gabe Newell on the Half-Life 2 source leak

22 years, 2 months

[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, 2 months

[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, 2 months

[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, 2 months

[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, 2 months

[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, 2 months

[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, 2 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev