Linux-audio-dev

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7501 discussions

by Gordon JC Pearce

I wanted a very simple SDR with jack inputs and outputs for a demonstration I was doing. I had a look at the DSP guts of dttsp and quisk, and sat down to code. Now, since I wanted to demonstrate how you could use LADSPA filters to clean up received audio, it occurred to me that I should implement my SDR core as a LADSPA plugin. So, I did. It "works for me". If you try it out, let me know how you get on. At 256 frames/period it sits at about 3% usage on my P4-2.8 without any other LADSPAs running - not bad, but it probably could be better. If you want to build it, get the code with: git clone git://lovesthepython.org/ladspa-sdr.git then build it with scons. You'll need to manually copy the resulting sdr.so to wherever your LADSPA plugins live. Load it up in jack-rack and add in an amplifier plugin (there's no AGC) and some sort of filter (I recommend the Glame Bandpass Filter). Performance and quality isn't exactly amazing, but for less than 300 lines of code - much of that used to set up the plugin - it's not too bad. Gordon MM0YEQ

15 years, 4 months

wanted: good online reading material for RT audio/midi software design?

by Harry Van Haaren

We seem to be fairly intrested in the same things James! I don't know if you have access to University Lecturers... if you do, go have a chat with the software engineer lecturer. I've only had positive experiences when approaching them about "totally-unrelated-to-course" projects. On the other hand, I bought a book (forget the exact name.. can find out) which showed some of the basic ObjectOrientated stuff, but at the same time, I found it to be relatively useless when trying to apply it to "music-software". (Ie: Ardour, Seq24, Dino, etc kind of programs) Spending time doing out program diagrams.. (you know the "standard" boxes approach to explaing how classes interact.) That's been my approach, I didnt really find any great resources online. If you do find any, please post back here! :-) Good luck, -Harry

15 years, 4 months

ambisonics UHJ encoder

by alex stone

Jorn, Fons, i'm looking for a ladspa UHJ encoder, and can't seem to find one. Any idea if such a beast exists? Or if there's a standalone instance or ambdec preset i can use, and route in and out of? Jorn ,i've had several browses over your web examples of using AMB plugins with Ardour, and have reflected the setup where possible in Non-Mixer. I'm using samples (ala LSampler) for noise, but i'll ask here, what's the function of using the tetraproc mike plugin over something else? I'm lost in your explanation. I'm still getting my feet wet in ambisonics, and making plenty of errors along the way, but progress seems imminent. (as it always does i guess, for the optimistic among us.) Some general questions. When i use Jconvolver standalone (my preference) and test with a *amb.conf, i get 1 input and 4 outputs WXYZ. Is this correct for 4 signals coming into 1, into the *amb.conf, or do i need to change this to reflect individual WXYZ routing, from something like a MASTER strip, or from an ambdec plugin in a channel strip? (i'm trying to get the signal chain sorted out correctly.) i.e. 4 in, 4 out. I'm using all mono ins for sound sources, and want to reflect positioning in the busses, as i have multitrack 1st violins, 2ndviolins, etc... So my 1st violins (4 monotracks) are going into a 1stviolin buss (4 ins) and in the buss signal chain, i'm adding a ladspa amb mono panner, which naturally gives me 4 outs, then the chain continues to the MASTER and jconvolver, back into a jconv buss in the mixer with the intent of finally routing that to the UHJ buss... Should i then stay "faithful" to that signal chain, and up to a UHJ encode to stereo (which i hope is in ladspa existence) maintain the 4 port stream to stay compliant with WXYZ? The intent with this is provide ambisonic positioning, and convolver tail, right up until downsizing to stereo as the last part of the signal chain. I'm finding the challenge of this interesting, and may have more questions as more of this slowly seeps into my head. Feel free to point out obvious errors, or alternative (meaning smarter) suggestions. Alex. -- www.openoctave.org midi-subscribe(a)openoctave.org development-subscribe(a)openoctave.org

15 years, 4 months

wanted: good online reading material for RT audio/midi software design?

by James Morris

Anyone know of any good reading material freely available online focused on real time audio/midi software design? As examples, of what I've found so far: (and what's wrong with them) Parts of this are helpful, but far too generalized: http://www.cs.st-andrews.ac.uk/~ifs/Books/SE7/Presentations/PDF/ch15.pdf Whereas this is far too technical, and also massively generalized: http://citeseerx.ist.psu.edu/viewdoc/download?doi=10.1.1.109.6980&rep=rep1&… Cheers, James.

15 years, 4 months

Jack port alias names

by Tim E. Real

Good day... Just coming to grips working with and learning the alias system... Under what conditions might a Jack port not have any alias names? When might I expect to encounter that situation? Because our app supports both ALSA midi and Jack midi, the app's very own ALSA ports are showing up in its list of Jack midi ports. We don't want our own ALSA ports listed in there. So to filter them out of our Jack midi ports list, I look at the port's (first) alias name and see if the app's name is in there, and filter out the port if it's a match. So far so good, but I'm worried what happens if there's no alias to work with. I can't figure out a way to determine if a non-aliased name like "system:midi_playback_4" actually belongs to our app's own ALSA ports. I don't know how or if 'alias renaming' will affect my plans. Still learning + investigating much about this system. Thanks. Tim.

15 years, 4 months

2.6.31.x vs. rme vs. alsa 1.0.22?

by Fernando Lopez-Lezcano

Hey, has anyone been seeing strange behavior from this combination? kernel 2.6.31.x rt20 + alsa 1.0.22 userland RME card (pcmcia card + multiface) hdspmixer is not doing the right thing (does not initialize the card in a way in which playback works), it does not see the hwdep interface (or something like that) and disables metering, alsamixer even segfaults when I reach the end of the controls listed. Plain weird. Smells like something changed deep in the kernel that makes alsa-lib very unhappy. Alsa-tools rebuilt from source does not make a difference. Weirdness goes away when I boot into 2.6.29.6 rt23... Is there anything in alsa-* that depends on which _kernel_ is available at compile time? -- Fernando

15 years, 4 months

Re: [LAD] [off-list] midi beat clock

by Tim E. Real

<ralf.mardorf(a)alice-dsl.net> wrote: > With my CPU [1] resampling seems to be still a problem. I'm using > Qtractor, but didn't use it's time stretch feature for audio clips, > instead I used Rubber Band as plugin to pitch down drums. This forced > the CPU to it's knees. Another issue might be trouble because of > transients. I like Rubber Band as an effect, so I'm fine with > transients, but for your needs they might be a problem. > http://www.breakfastquay.com/rubberband/technical.html Found a new one Zita-resampler. Can't wait to try it. http://www.kokkinizita.net/linuxaudio So far I'm only using SRC, currently writing classes for the others. So far so good if I don't push it with waay too many audio tracks.

15 years, 4 months

For your information

by fons＠kokkinizita.net

To all, for your information. I have received the following message from Mr. Nick Copeland. It was sent privately, but since this is the continuation of a thread on this list and the person concerned has well gone beyond any reasonable limits of decent behaviour I feel free to post it here. *** Start included message *** The last time I looked at the AMS source code I was under the impression that the Moog VCF filters were not actually yours, they were Tossavainen and Kellets. They were published eventually placed on musicdsp.org however there are a couple of things here: 1. In their original publishing they did not advocate the use of quantisation of their parameters - you implemented this as an adaptation of their algorithm. 2. In not quoting the original authors you are plagiarism them. In the source to AMS they are quoted however it is bit disingenious to imply that this was 'your' MoogVCF. In short you did not actually write the 'Moog' VCF parts and suggesting that this is how their algorithms worked is just attempting to justify quantisation to 1/16 is not deleterious to quality which is definitely not attributable to the original authors. >From what you are saying I would like to just say you are an arrogant selfpublicist however base on the claims in your email you are actually a lying plagiarst. Regards nick. *** End included message *** -- FA O tu, che porte, correndo si ? E guerra e morte !

15 years, 4 months

jack_diplomat

by alex stone

Once upon a time, jack_diplomat was alive and well. Sadly, all attempts to find it have failed, as the hosting site seems to be, extinct. Does anyone have a copy of this app on their machine that they might consider sharing? Alex. -- www.openoctave.org midi-subscribe(a)openoctave.org development-subscribe(a)openoctave.org

15 years, 4 months

patterns, GUI, JACK midi interactions..

by james＠jwm-art.net

Hi, The sequencer/arpeggiator/WYWTCI I'm working on, (eventually) will partially generate MIDI events (primarily NOTE-ON/OFF) from a pattern. The pattern consists of equi-distant intervals (ie 1/8th or 1/16th etc), where each interval has the potential to play a note or not. The note pitch and velocity is generated by something akin to window placement. Given a grid: pitch @ X, velocity @ Y, place a box within the grid using an algorithm to prevent overlapping other boxes* etc, but only when the pattern says to do so. Initially I thought I'd have this generative stuff within the JACK process callback, but soon decided not too... Here is how I think this will work: 1) The 'pattern processor' processes 1 quarter note at a time (I guess)... looking ahead, processing notes which will be playing shortly, adding the generated note data to a main event list for the sequencer (??) 2) The sequencer processes the main event list, creating the MIDI events and adding them to a jack_ringbuffer. 3) the JACK process thread/callback reads the jack_ringbuffer and outputs the MIDI events it finds there to the JACK MIDI port. Now, somewhere along the line, the GUI has to provide visual feedback at exactly (or as exactly as possible) the time any note on/off occurs. Perhaps then: 4) the JACK process thread/callback upon sending a midi event, adds to a 2nd ring buffer which is read by the GUI thread, and displays the visual feedback (ie a coloured box). Also, the 'pattern processor' will need to be notified of note off events so it can remove the box from the grid (such that new notes might replace the old). So.. 5) the Jack process thread cb upon sending a note off adds to a 3rd ringbuffer which is read by the pattern processor to notify it of the removal of a note/box. Does this sound about right? Cheers, James. * see also: http://jwm-art.net/art/text/xwinmidiarptoy http://jwm-art.net/art/text/xwinmidiarptoy.txt

15 years, 4 months

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Linux-audio-dev