Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7448 discussions

by Alex Stone

I've been working with large lscp files, over a hundred .idfs for muse 2, etc, for sometime now, and i'm pretty well knackered from manually creating large monolithic instrument config files of some sort or another, for the same sample library set, across apps. I propose the following, and i'm wiling to put some money in someone's pocket for this: A simple app (CLI is ok but GUI based preferred) that: 1) automatically converts lscp files containing single or multiple midi instrument maps into individual midnam files. 2) automatically converts muse2 .idf files into midnam files. 3) converts midnam files into muse2 .idf files Obviously needs to work properly, and create correct ( auto app checked) midnam and .idf files. App needs to be opensource (obviously), and public, for those other poor souls who have to struggle with large instrument files, particularly those using large sample libs with copious articulations per instrument, and for whom automation of these excruciating data processing tasks would save massive amounts of time and effort. Thanks, and if you're interested, contact me privately, Alex Stone.

11 years, 6 months

MusE 2.1.2 released

by Robert Jonsson

Hi all, Time for another update for general consumption, minor changes done but quite a few of them! Nearly arbitrary list of changes from the changelog: - Allow reading old drummaps for new style drumtracks - Added metronome icon in main window - Fixed moving events with keyboard in Drum editor - Added theme support, Light, Dark and Ardour so far - Added missing line draw shortcut (F) to drum editor. - Added new french translation from Yann Collette - Added: Pan and Zoom tools to editors. P + Z shortcuts. Added a Settings item for alternate behaviour. - Fixed: Pianoroll and Drum Editor 'Snap' boxes not remembering 1st or 3rd columns. - Fixed: Arranger 'Snap' was not stored or remembered. - Fixed: Accelerator buttons shift/ctrl/alt for moving/copying/cloning / restricting movement. - Fixed: Shift key restricting movement: Ignore snap setting now. - Fixed: Resize shift key ignore snap setting now. - Fixed: Draw new item shift key ignore snap setting now. - Fixed: Shift key was not snapping to vertical. - Fixed: ALL 'Speaker' related playing of notes. Works with new notes, moving notes, piano press etc. - Fixed: ALL 'Speaker' related notes now send true note-offs instead of zero-velocity note-ons. - Fixed: Drum 'Cursor' mode was playing double notes. - Fixed: New Drums 'Cursor' mode and instrument up/down movement was broken, jumping all over the place. - Added prebuilt PDF of manual (work in progress) - Improved: Shortcut listings: Added Wave/Score categories. Re-categorized several keys. Updated README.shortcuts - Improved: Right-click menus expanded. Now also shows 'Tools' menu when clicked on parts. - Added choice of new metronome with different sounds and adjustable volume. - Fixed gain adjustment with 'Other' choice in wave editor, it was reversed For more information and additional changes see the full changelog: http://lmuse.svn.sourceforge.net/viewvc/lmuse/trunk/muse2/ChangeLog?revisio… Find the download at: https://sourceforge.net/projects/lmuse/files/ MusE on! The MusE Team

11 years, 6 months

JACK latency chart

by Joel Roth

Hi all, I once saw a helpful chart showing how latency propagates through the JACK graph, through the capture and playback ports of each node. Can anyone provide a link? I want to make sure I understand correctly, that my app's latency callback should be checking capture latency at the input ports and setting playback latency on the output ports. Thanks, Joel -- Joel Roth

11 years, 6 months

JAudioLibs (JNAJack, etc.) code on GitHub

by Neil C Smith

Hi All, Just a note that I've extracted JNAJack (Java bindings for JACK) from the Praxis LIVE repository, along with other elements of the JAudioLibs code, and it's now on GitHub at https://github.com/jaudiolibs Should make it easier for people to work with (and contribute to!) the code. Downloads and other facilities are still on the Google Code site at http://code.google.com/p/java-audio-utils/ for now. Best wishes, Neil -- Neil C Smith Artist : Technologist : Adviser http://neilcsmith.net Praxis LIVE - open-source, graphical environment for rapid development of intermedia performance tools, projections and interactive spaces - http://code.google.com/p/praxis OpenEye - specialist web solutions for the cultural, education, charitable and local government sectors - http://openeye.info

11 years, 6 months

Mixing audio: Implementing pan and balance

by Tim E. Real

Hi, I need some advice, clear up some confusion: I noticed our app uses this pan formula: vol_L = volume * (1.0 - pan); vol_R = volume * (1.0 + pan); where volume is the fader value, pan is the pan knob value which ranges between -1.0 and 1.0, and vol_L and vol_R are the factors to be applied to the data when sending a mono signal to a stereo bus. When pan is center, 100% of the signal is sent to L and R. At pan extremities, the signal is boosted by 3dB. But according to [1], we should be using a Pan Law [2], where pan center is around 3dB to 6dB down and pan extremities is full signal. So I want to change how we mix mono -> stereo and use true Pan Law. I could add a Pan Law selector, seems like it might be useful for various studio acoustics. Then I noticed we use the same formula above to apply 'balance' (using the same pan knob) when sending a stereo signal to a stereo bus. But according to [3] we should be using a true balance control, not those same pan factors above. And according to [1]: "Note that mixers which have stereo input channels controlled by a single pan pot are in fact using the balance control architecture in those channels, not pan control." So I want to change how we mix stereo -> stereo and use true balance. But then I checked some other apps to see what they do. In an unofficial test I noticed that QTractor seems to do the same thing, that is, when pan is adjusted on a stereo track, one meter goes up while the other goes down. RG seems not to have stereo meters and Ardour I couldn't seem to make pan affect the meters, I will try some more. My questions: Is the pan formula above popular? What is the consensus on stereo balance - use a Pan Law, being the formula above or otherwise, or use a true balance? What should I do in the remaining case sending a stereo signal to a mono bus? If I am using a Pan Law as balance, the two signals will have already been attenuated at pan center so I could simply sum the two channels together. But if instead I use true balance, at center the two signals are 100%. So should I attenuate the signals before summing them to a mono bus? Currently as our pan formula above shows, there would be no attenuation. Thanks. Tim. [1] http://en.wikipedia.org/wiki/Panning_%28audio%29 [2] http://en.wikipedia.org/wiki/Pan_law [3] http://www.rane.com/par-b.html#balance

11 years, 6 months

jackmixdesk-0.3-r3

by Ulrich-Lorenz Schlueter

Hello list, after a long long time I removed some ugly bugs from JackMixDesk. Latest source package is available from: http://sourceforge.net/projects/jackmixdesk/files/jackmixdesk/jackmixdesk-0… Regards, Uli

11 years, 6 months

DAFX13 - deadline extension, 14/04

by Victor Lazzarini

The deadline for submission of papers for DAFX13 has been extended to SUNDAY APRIL 14th More information on http://dafx13.nuim.ie/authors.html You can also follow us on twitter https://twitter.com/DAFxInfo We're looking forward to your submission! Dr Victor Lazzarini Senior Lecturer Dept. of Music NUI Maynooth Ireland tel.: +353 1 708 3545 Victor dot Lazzarini AT nuim dot ie

11 years, 6 months

DAFX13 - deadline extension.

by Victor Lazzarini

The deadline for submission of papers for DAFX13 has been extended to April 14th. More information on http://dafx13.nuim.ie/authors.html You can also follow us on twitter https://twitter.com/DAFxInfo We're looking forward to your submission! Dr Victor Lazzarini Senior Lecturer Dept. of Music NUI Maynooth Ireland tel.: +353 1 708 3545 Victor dot Lazzarini AT nuim dot ie

11 years, 6 months

Handling of expression controller in fluidsynth and linuxsampler

by Nikita Zlobin

Trying to get live strings and vocal in linuxsampler, i was suprised by fact, that this controller doesn't have effect (with all engines and banks, including suggested for fluidsynth), but with fluidsynth all ok. It is not clear now, how this controller should be handler at all. I expected, that it should be same as for velocity, but set for enter channel and variable, unlike V, specified for notes and constant). I searched both in fluidsynth and qsynth sources to see, for handling code, but found in fluidsynth only two constants (EXPRESSION_LSB and EXPRESSION_MSB) and command, setting these parameters to default value (127). IMHO, it is very necessary for linuxsampler, and who knows, what is not done for fluidsynth. Looks like LS devs left it for user (using qmidiroute, mididings or sequencer's builtint converting capabilities if available). This link only confirms that: http://bb.linuxsampler.org/viewtopic.php?f=6&t=207

11 years, 6 months

Mixing audio: Noiseless volume changes

by Tim E. Real

Hi again. Looking for any advice, tips, tricks, anecdotes etc. I want to eliminate or reduce 'zipper' noise on volume changes. So I'm looking at two techniques: Zero-crossing / zero-value signal detection, and slew-rate limiting. Code is almost done, almost ready to start testing each technique. Each technique has some advantages and disadvantages. If I use a slew-rate limiter, I figure for a sudden volume factor change from 0.0 to 1.0, if I limit the slew rate to say 0.01 per sample then after 100 samples the ramp will be done. But even with a fine ramp, this still might introduce artifacts in the audio. If I use a zero-crossing/zero-value detector and apply volume changes only at these safe points, that's a much more desirable 'perfect' system. But I stuck a time limit on waiting for a zero-cross because it's possible the signal might have a high DC offset, or contain VLF < 20Hz. (One cannot simply wait for the current data value to be 'zero' because for example with a perfect square wave signal the 'current' value will never approach zero, hence the zero-crossing detection requirement.) At some point waiting for a zero-cross, a ramp would have already finished and it may have been better to use that ramp instead. Conversely, a zero-cross might happen sooner than a ramp could finish, and we definitely want to use that zero-cross here instead of the ramp. So it means I either have to give the user a choice of the two techniques, or try to automatically switch between them - which ever one occurs first, the ramp finishing or the zero-cross, use it. But it means I have to keep two audio buffers - one for applying the ramp as the samples are processed, and one for waiting until zero-cross happens - and which ever one "finishes the race" first, that buffer "gets the nod". The zero-crossing technique has some interesting implications. For a stereo signal each channel's zero-cross will happen at different times. So I'm trying to imagine what that's going to sound like where the volume changes happen at slightly different times, if it will be noticeable, even though that is far better than 'zipper' noise. Also I'm trying to imagine how track cross-fading support would deal with zero-crossing - if it is better to use ramps in that case. What do you think? Thanks. Tim.

11 years, 6 months

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Linux-audio-dev