Linux-audio-dev

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3 participants
7535 discussions

Re: [LAD] on the soft synth midi jitters ...

by Jeff McClintock

>> BUT never ever a licenced Windows + a bought Cubase will cause such an >> issue, assumed you didn't install a cracked Windows Office too. >clearly you have no idea who Jeff McClintock is, or you wouldn't be offensive. ;) I do use licensed software. I am quite anti-piracy and have made submissions to the government on the subject, even got a letter published in PC World magazine. Off-Topic: IMHO Piracy hurts Linux by providing a competing 'low cost' alternative to *real* free (FOSS) software.

15 years, 4 months

[LAD] question on ADAT S/MUX using snd-ice1712 and Alesis AI4

by Niels Mayer

Now that I have an ADAT-capable card ($20 ebay ice1712-based terratec ews88d) I'm curious... if I combine it with something like http://www.kellyindustries.com/computer/alesis_ai4.html ( http://www.alesis.com/ai4 ) and use the S/MUX mode built in to the AI4 across eight channels to create four 24/96 channels. //// //// //// //// In order for the AI-4 to operate at the 96 kHz samplerate it has to be run in S/MUX mode or sample split mode which means that you get 4 channels of conversion and not 8. This is standard and perfectly acceptable. The first two channels will be routed out to the ADAT lightpipe outputs 1 through 4 and the 2nd two channels with be routed out ADAT lightpipe outputs 5 through 8. //// //// //// //// Would I be able to "transparently" use these "S/MUX"d channels in a linux DAW, by simply recording/playing-back a higher channel count per track (e.g. 4 for a stereo track, 2 for a mono)? This seems like a good way to achieve "studio quality" 24/96 record&playback at a distance from the computer -- via ADAT cable -- using high quality outboard A/D and D/A or interfacing to external equipment already presenting AES/EBU format I/O. If a device like the AI4 actually does all the bit-splitting and other fu, both for input and output -- wouldn't it not matter that the actual 2-track or 4-track contents are essentially "noise" because nothing in Linux-land would understand the S/MUX format. Next question: to avoid the hack suggested above, is there some kind of ALSA plugin that would reconstitute/create synchronized pairs of S/MUX data on the same soundcard into single 24/96 streams, both for input and output? How is S/Mux handled in Linux & ALSA? Thanks. Niels http://nielsmayer.com PS: The Alesis AI4 seems nicer than the http://www.aphex.com/144.htm -- for one I won't need to make 110 ohm cables with XLR connectors to DB25. The AI4's support for S/MUX mode is especially nice-sounding -- if there was a way for it to work in linux. Any other suggestions for converting ADAT to AES/EBU (or spdif) w/ decent synchronization options for input?

15 years, 4 months

[LAD] Announcing libsndfile-1.0.23 release

by Erik de Castro Lopo

Hi all, Latest release version 1.0.23 is available here: http://www.mega-nerd.com/libsndfile/#Download Changes are: * Add version metadata to Windows DLL. * Add a missing 'inline' to sndfile.hh. * Update docs. * Minor bug fixes and improvements. Cheers, Erik -- ---------------------------------------------------------------------- Erik de Castro Lopo http://www.mega-nerd.com/

15 years, 4 months

Re: [LAD] on the soft synth midi jitters ...

by Jeff McClintock

> It could be useful to have some anecdotal evidence to quantify measures > of jitter like "annoying" and "drunk", so: > > What is your buffer-size? Hi Jens, I test with several sound cards. M-Audio, Creative Audigy, ASIO-for-all, and generic motherboard driver. I've found the jitter at settings over 30-50ms difficult for serious recording. Without ASIO drivers latency can be 100-200ms or more, which is very bad. ASIO at 5-10ms seems the best I can use on Windows without stutter, that feels nice and responsive to me. Best Regards, Jeff

15 years, 4 months

Re: [LAD] on the soft synth midi jitters ...

by Jeff McClintock

> I'm also a bit puzzled by people complaining about jitter. I don't have > any exceptional kit, but in reality I can't say I've ever noticed it. > Latency yes, but that's easily corrected with a bit of post record > nudging. Cubase is particularly bad when playing a soft-synth live, esp with larger audio buffer sizes, because even though VST supports sample-accurate MIDI, all note-ons are sent with timestamp of zero (the exact start of the buffer). It's like trying to play drunk, like glue in the keys, I keep looking at my fingers thinking "did my finger slip off that note?". Playing a pre-recorded MIDI tract is different, timestamps are then honoured. Why did Steinberg implement it like this?, I think it's a misguided attempt at reducing latency. It's doesn't, the worst case notes are still delayed exactly one 'block' period. There's no upside. It's far better to have small latency and no jitter because your brain will compensate very accurately for consistent latency, you will instinctively hit the keys a fraction early. All will sound fine. Jitter is baked-in timing error, once it's in your tracks you can't get it out. Latency can always be compensated for and eliminated later. The right way is to timestamp the MIDI, send it to the synth delayed by one block period. Since audio is already buffered with the same delay, you will get perfect audio/MIDI sync. IMHO - After writing my own plugin standard, sample-accurate MIDI is no more difficult to support than block-quantized MIDI. Jeff McClintock > Message: 8 > Date: Tue, 5 Oct 2010 21:22:23 +0100 > From: Folderol <folderol(a)ukfsn.org> > Subject: Re: [LAD] on the soft synth midi jitters ... > To: linux-audio-dev(a)lists.linuxaudio.org > Message-ID: <20101005212223.5a7fbb61@debian> > Content-Type: text/plain; charset=US-ASCII > > On Tue, 5 Oct 2010 22:00:11 +0200 > fons(a)kokkinizita.net wrote: > > > On Tue, Oct 05, 2010 at 02:50:10PM +0200, David Olofson wrote: > > > > > Not only that. As long as the "fragment" initialization overhead can > be kept > > > low, smaller fragments (within reasonable limits) can also improve > throughput > > > as a result of smaller memory footprint. > > > > 'Fragment initialisation' should be little more than > > ensuring you have the right pointers into the in/out > > buffers. > > > > > Depending on the design, a synthesizer with a large number of voices > playing > > > can have a rather large memory footprint (intermediate buffers etc), > which can > > > be significantly reduced by doing the processing in smaller fragments. > > > > > Obviously, this depends a lot on the design and what hardware you're > running > > > on, but you can be pretty certain that no modern CPU likes the > occasional > > > short bursts of accesses scattered over a large memory area - > especially not > > > when other application code keeps pushing your synth code and data out > of the > > > cache between the audio callbacks. > > > > Very true. The 'bigger' the app (voices for a synth, channels for > > a mixer or daw) the more this will impact the performance. Designing > > the audio code for a fairly small basic period size will pay off. > > As will some simple optimisations of buffer use. > > > > There are other possible issues, such as using FFT operations. > > Calling a large FFT every N frames may have little impact on > > the average load, but it could have a big one on the worst case > > in a period, and in the end that's what counts. > > > > Zyn/Yoshimi uses FFTs for some of its algorithms IIRC. Getting > > the note-on timing more accurate could help to distribute those > > FFT calls more evenly over Jack periods, if the input is 'human'. > > Big chords generated by a sequencer or algorithmically will still > > start at the same period, maybe they should be 'dispersed'... > > > > Ciao, > > I'm all in favour of a bit of dispersal. > > When I started out with a Yamaha SY22 and Acorn Archimedes it was all > too easy to stuff too much down the pipe at once. However, doing some > experimenting, I was surprised at how much you could delay or advance > Note-On events undetectably although it depended to some extent on the > ADSR envelope. > > I don't need to do that any more, but old habits die hard, so if I'm > copy-pasting tracks I tend to be deliberately a bit sloppy. > > I'm also a bit puzzled by people complaining about jitter. I don't have > any exceptional kit, but in reality I can't say I've ever noticed it. > Latency yes, but that's easily corrected with a bit of post record > nudging. > > -- > Will J Godfrey > http://www.musically.me.uk > Say you have a poem and I have a tune. > Exchange them and we can both have a poem, a tune, and a song.

15 years, 4 months

[LAD] Looking for Electromyography (EMG) sensors that are easily accessed under linux

by Nathanael Anderson

Here is an example of Electromyography* *sensors (emg) http://www.biometricsltd.com/analysisemg.htm I'd like to be able to control a sequencer with muscle movements, I'd write some code to process the inputs and convert them to midi, but need to find some inexpensive emg's to use that I can read data from under Linux. Anyone have any reconsiderations? Thanks Nathanael

15 years, 4 months

[LAD] Announce : libsndfile 1.0.22

by Erik de Castro Lopo

Hi all, Latest libsndfile is now available here: http://www.mega-nerd.com/libsndfile/#Download Changes are: * Couple of fixes for SDS file writer. * Fixes arising from static analysis. * Handle FLAC files with ID3 meta data at start of file. * Handle FLAC files which report zero length. * Other minor bug fixes and improvements. Cheers, Erik -- ---------------------------------------------------------------------- Erik de Castro Lopo http://www.mega-nerd.com/

15 years, 4 months

[LAD] [ANN] Qtractor 0.4.7 - The Furious Desertrix is out!

by Rui Nuno Capela

It's TYOQA (The Year Of Qtractor Automation:) what else? But wait, there's three months to go yet. Meanwhile, the foundations have already been laid and one can now tell that a rocky milestone is ready to get bumped. Ouch! Qtractor 0.4.7 (furious desertrix) is out! Release highlights: - MIDI learn/controller mapping for all plugin parameters (NEW) - Extended Clip fade-in/out WYSIWYG curves (NEW) - MIDI resolution overflow (FIX) - MIDI tempo standard base on quarter-note (FIX) - Extended MIDI controller mapping for mixer/tracks (NEW) - Audio metronome gain control (NEW) - Mute/solo tracks while looping (FIX) - MIDI Clock support (NEW) - Audio clip import while looping (FIX) - MIDI track bank-select/program-change transparency (FIX) - VeSTige headers included for native VST plugin support (NEW) - JACK transport sync support (FIX) - Clip tempo-adjust tool (NEW) - Audio tracks auto-monitoring (FIX) - Transport back/forward stops on loop points (NEW) - MIDI tracks redundant mute/solo (FIX) See also: http://www.rncbc.org/drupal/node/235 Website: http://qtractor.sourceforge.net Project page: http://sourceforge.net/projects/qtractor Downloads: - source tarball: http://downloads.sourceforge.net/qtractor/qtractor-0.4.7.tar.gz - source package (openSUSE 11.3): http://downloads.sourceforge.net/qtractor/qtractor-0.4.7-1.rncbc.suse113.sr… - binary packages (openSUSE 11.3): http://downloads.sourceforge.net/qtractor/qtractor-0.4.7-1.rncbc.suse113.i5… http://downloads.sourceforge.net/qtractor/qtractor-0.4.7-1.rncbc.suse113.x8… - user manual (nevermind outdated): http://downloads.sourceforge.net/qtractor/qtractor-0.3.0-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: - While moving multi-selected MIDI events around the clip editor (aka piano-roll), with help of keyboard arrow keys, that is, was not clear which one was the so-called "anchor" event, the one which positioning gets honored for snap-to-beat business. Not anymore: the anchor event now defaults to the earliest in time or the one the user's last point(-click)ed. - MIDI control observer pattern implementation has sneaked in, making it ready for the so-called and long-awaited "MIDI Learn" feature and arbitrary MIDI controller assignment, for plugin parameters in particular. - MMC DEFERRED PLAY doesn't cause transport state to stop if currently rolling (mitigating bug #3067264). - Audio clip merge processing might have been skipping a few initial frame blocks, now fixed. - Clip selection and plugin parameter hash optimization. - Anti-glitch audio clip macro fade-in/out fixed again. - New clip fade-in/out slopes (curves) are introduced, partially adapted and refactored from those easing equations of Robert Penner's fame. - Clip fade-in/out non-linear slopes are now shown as actual WYSIWYG curves. - Escape key now closes generic plugin widgets as ever found usual elsewhere. - Picking nits: unselect current track when clicking on any gray empty area, also accessible from a new menu item: Track/Navigate/None. - A nasty and deadly MIDI resolution overflow has been finally fixed, allowing for long MIDI sequences (1h+) to load correctly on 32bit machines from now on (was perfectly fine on 64bit though). - MIDI editor selection hash optimization in face of reasonably huge event sequences. - MIDI controller mapping finally refactored to support some other MIDI event types than just CC (0xBn) ones. - Nitpicking fix: corrected main track-list (left pane) display when no track is currently selected. - libX11 is now being added explicitly to the build link phase, as seen necessary on some bleeding-edge distros eg. Fedora 13, Debian 6. (fixing bug #3050944). - New audio metronome bar and beat sample gain options. - Progressively, the observer pattern is being finally introduced, targeting all potentially automation controls and widgets as plain ground-zero for the (ultra-)long overdue automation feature. - MIDI controller mapping of still non-existing tracks were being implicitly assigned to the last, highly numbered, existing track. Now fixed. - Moving from old deprecated Qt3'ish custom event post handling into regular asynchronous signal/slot strategy. - Muting/soloing tracks while playback is looping was leaving current audio clip out-of-sync whenever that same track is later un-muted on any other preceding clip. Now hopefully fixed. - MIDI Clock support makes its first appearance. - All tempo (BPM) calculations are now compliant to the MIDI conventional equivalence between beat and quarter note (1/4, crotchet) as common standard time division. - Automatic audio time-stretch option is not enabled by default anymore. - Standard warning Apply button is now only shown when dismissing dialog changes are actually valid. - Make sure non-dedicated metronome and player buses are properly reset and reopen when changing regular audio buses (hopefully fixing bug item #3021645 - Crash after changing audio bus). - Hopefully, an outrageously old bug got squashed away, which was causing random impromptu crashes, most often when importing audio clips while looping and play-head is any near the loop end point. - General standard dialog buttons layout is now in place. - Fixed main track view off-limits play-head positioning. - Main tool-bar Time and Tempo spin-boxes, may now have their colors correct, as for most non-Qt based theme engines (ie. Gnome). Green text on black background has been and still is the the intended aspect design ;) - MIDI file import and internal sequence representation has been changed to be inclusive on all bank-select (CC#0,32) and program-change events which were previously discarded while honoring MIDI track properties. Interleaved SysEx events are now also preserved on their original sequence positions instead of squashing a duplicate into the MIDI bus SysEx setup. - Attempt to include the VeSTige header by default, as for minimal VST plugin support. - JACK transport support has been slightly rewritten, in fact the sync callback is now in effect for repositioning. - The MIDI clip editor (piano roll) widget won't be flagged as a tool window anymore. - A tempo adjustment tool is making inroads from the menu, as Edit/Clip/Tempo... (factory shortcut: F7). - Audio tracks auto-monitoring is now effective on playback. - Make sure to ask whether a dirty MIDI clip should be saved, upon resizing or stretching its edges (fixes bug #3017723). - Backward and Forward transport commands are now taking additional stops on loop points. - Attempt to optimize track solo/mute redundant transactions, in special regard to MIDI track events which were being duplicated on soloing and temporarily muted on unsoloing. Cheers && Enjoy (be happy!) -- rncbc aka Rui Nuno capela rncbc(a)rncbc.org

15 years, 4 months

[LAD] on the soft synth midi jitters ...

by cal

I'm writing on behalf of a friend who's having trouble dealing with midi latency in a soft synth (possibly yoshimi) ... Given a jack period of N frames, the midi latency with the original code effectively ranges from N frames to 2 * N frames, which I guess qualifies it as jittery. So far my friend has tried a few things, but there's no workable solution as yet. What seemed most promising was to break the audio generation into smaller blocks, applying pending midi events between blocks. Sadly, that drags the creation and destruction of note objects into the realtime jack process callback path. Latency improves, but the number of notes you can get away with before it all falls in a heap is significantly reduced. Getting the destruction of dead notes out of the realtime path is trivial, not so the creation of new ones. Even with a pool of pre-allocated note objects, it seems the amount of initialization code per note is still a real limiting factor on how busy things can get before it all falls apart. Such is life (for my "friend"). cheers, Cal

15 years, 4 months

[LAD] CUDA implementation for calf

by Max Tandetzky

Hallo, I am new here, so I hope this is the right place to talk about what I want to do. I want to make a CUDA implementation of the algorithms from the calf-plugins. On the front end there should be placed a button (or something else) to (de-)activate the CUDA support. I have already written a jack-program (which makes some simple changes on audio data) using CUDA. It works good and at a first sight the performance looks promisingly. I have read a part of the mailing list archive and found out that there already was a discussion about audio processing with CUDA. I know there are some reasons for not using CUDA like the duty to use the proprietary Nvida driver, the limitation that only people who have an Nvidia card will have a benefit and so on. But the CUDA implementation may show which performance can be reached and may beuseful for Nvidia users immediately. I know there is OpenCL but it is not as sophisticated as CUDA at the moment, will have less performance than CUDA and I do not have the time to learn OpenCl at the moment (but the project has to be finished soon). I heard it is not too much work to transfer existing CUDA code to OpenCl code later (assuming there is already an OpenCL implementation for all CUDA functions which were used). So I want to do this with CUDA. At the moment I have some questions: 1. Is there anybody has already done or is doing something like this? 2. Where can I get information to make any specific changes on the calf code? (I examined it a bit but it will take time to understand the structure of the program when I only have the code, especially the part for the GUI seems to be conceptualized a bit more complex.) It would be nice if I can get some help here. Regards Max Tandetzky

15 years, 4 months

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Linux-audio-dev