Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7516 discussions

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

12 years, 7 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

12 years, 7 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

12 years, 7 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

12 years, 7 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

12 years, 7 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

12 years, 7 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

12 years, 7 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.

12 years, 7 months

SysV init scripts vs. systemd, a short note

by Fernando Lopez-Lezcano

Hi all, Hopefully this will be useful to others. I just wrote a short note describing my experiences when moving from a SysV init script based OpenMixer[*] system to one that uses systemd. In short, yes, it is possible, no, it was not easy (mostly because of my own ignorance of systemd, of course :-) https://ccrma.stanford.edu/~nando/plog/20130315/ Enjoy! -- Fernando [*] OpenMixer is a mixing and routing application written in SuperCollider that manages audio for our 22 channel 3D Listening Room. It has to start on boot and runs as the "openmixer" user and group, and I wanted it to not use a regular Desktop "autologin" feature but rather be a SysV init service... https://ccrma.stanford.edu/room-guides/listening-room/ https://ccrma.stanford.edu/software/openmixer/manual/

12 years, 7 months

[ANN] Qtractor 0.5.8 - The India Romeo is out, singing a serenade...

by Rui Nuno Capela

Spring is nigh. Qtractor 0.5.8 (india romeo) is out, singing a serenade... Nothing but the change-log (see below:)) Enjoy && lots of fun. Website: http://qtractor.sourceforge.net Project page: http://sourceforge.net/projects/qtractor Downloads: - source tarball: http://downloads.sourceforge.net/qtractor/qtractor-0.5.8.tar.gz - source package (openSUSE 12.3): http://downloads.sourceforge.net/qtractor/qtractor-0.5.8-5.rncbc.suse123.sr… - binary packages (openSUSE 12.3): http://downloads.sourceforge.net/qtractor/qtractor-0.5.8-5.rncbc.suse123.i5… http://downloads.sourceforge.net/qtractor/qtractor-0.5.8-5.rncbc.suse123.x8… - quick start guide & user manual: http://downloads.sourceforge.net/qtractor/qtractor-0.5.x-user-manual.pdf Weblog (upstream support): http://www.rncbc.org License: Qtractor is free, open-source software, distributed under the terms of the GNU General Public License (GPL) version 2 or later. Change-log: - Dropped old audio ramping spin-locks as its glitching reduction weren't that effective anymore. - Audio bus and track gain may now be set for amplification again, from +0dB up to +6dB, while using the Mixer strip sliders/faders (an old function found missing ever since pre-TYOQA). - Basic LV2 X11 UI support has been added through libSUIL but only really effective if plugins doesn't support the LV2 External UI extension in addition which takes precedence on any case. - Improved precision tolerance on the Tempo Map / Markers dialog. - Reinstated and fixed (old) warning and impending re-conversion on loading session files which the original sample-rate differs from current audio device engine (aka. JACK). - LV2 Plugin State/Preset name discrimination fix (after a ticket by Jiri Prochaszka aka. Anchakor, thanks:) - Linked/ref-counted audio clips must not overlap and now must have a buffer-size worth of a gap between each other. - Something fishy has been detected in the SSE (not so) optimized code from SoundTouch's-inspired WSOLA time-stretching. - Splitting clips apart is now easier than ever: a brand new entry enters the main menu scene: Edit/Split (Ctrl+Y) splits up clips according to current range/rectangular selection. - Audio clip offsets are now properly corrected when time-stretching is applied via Shift/Ctrl+dragging any of the clip edges. - One semi-colon typo was hiding proper descrimination of peak files used to draw distinct waveforms of time-stretched audio clips. - Track automation curves are now also affected by Edit/Insert/Range commands. - Finally, some visual feedback is shown while audio track export is running, in he form of a main status progress bar. - New user option: save backup versions of existing session files. - Default session directory now set to regular file's path on load. - A convenient minimum slack duration has been fixed for MIDI SysEx messages. - LV2 Time/position information is now asynchronously fed back into their parameter (control input) ports when designated. - LV2 State is now properly restored for plugins inserted on buses, probably solving the Calf Fluidsynth SoundFont information missing on buses ticket, reported by Albert Graef, thanks. - Fixed an immediate null pointer crash on creating a parentless new group while on the files organizer widget. - Preparations for future Qt5 migration. Cheers! -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

12 years, 7 months

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Linux-audio-dev