Linux-audio-dev

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7454 discussions

audio architecture wiki/docs/e-book idea?

by Iain Duncan

Hey everyone, especially those who have been helping me with me architecture questions. I'm wondering whether some of you would be interested in helping in a simple advisory/editorial capacity if I were to try to write up a wiki or e-book of some kind on real time audio architecture. It seems to me like there is a lot of knowledge here on it, and not very many (if any) good starting points for such things. I've found decent references on the dsp side of audio coding, but not seen anything on 'how to put together a real time audio app from a-z' kind of thing. I find writing docs helps me clarify things in my head, I'd be interested in doing some writing if I know that people who know what they are doing would be interested in advising and correcting. I figured if I put it online it might be a good source of publicity for your work and we could link back to projects ( ardour, etc, ) It would take a while of course, but might also help people new to these lists and give us all some thing to point at and say: there's a good write up on that here -> thoughts? iain

12 years, 10 months

Blog post on Polyphonic Faust synth

by Timo Westkämper

Hi. I wrote this blog post on a Faust based polyphonic virtual analog synth prototype : http://westkamper.wordpress.com/2011/11/09/polyphonic-faust/ This might be of interested to some of you Cheers, Timo

12 years, 10 months

Guitarix release guitarix2-0.20.0

by Andreas Degert

Guitarix release guitarix2-0.20.0 After much code shuffling, refactoring and testing we are happy to release "whizzing abacus" guitarix2-0.20.0. Thanks to all testers and special thanks to rosea grammostola for his patience. Guitarix is a tube amplifier simulation for jack, with effect modules and an additional stereo effect chain. Please refer to our project page for more information: http://guitarix.sourceforge.net/ new features and bugfixes in short: * important bugfix: convolver (cabinet, presence) in 0.19.0 only worked when samplerate was 48000Hz * avoid connect already connected ports * always save state on exit (in earlier versions state was not saved when presets where selected) * separation of engine and UI if you always wanted to use Guitarix headless or embedded, now there's a chance :-) * now its possible to set the reference pitch of the tuner * remember currently selected preset in guitarix state file * display selected preset / factory preset in window title and patch info window * updated factory settings from funkmuscle * reworked guitarix operation under a jack session manager * error popup window in addition to existing logging facility * command line option to set instance name (determines jack client names and state file to use) * some more work to support localization * upgraded to zita-convolver version 3 * other smaller changes and clean-ups download site: http://sourceforge.net/projects/guitarix/ please report bugs and suggestions in our forum: http://sourceforge.net/apps/phpbb/guitarix/ here you can find a couple of examples produced by guitarix users: http://sourceforge.net/apps/phpbb/guitarix/viewtopic.php?f=11&t=83 have fun _________________________________________________________________________ For extra Impulse Responses, guitarix uses the zita-convolver library, and, for resampling we use zita-resampler, both written by Fons Adriaensen. http://kokkinizita.linuxaudio.org/linuxaudio/index.html We use the marvellous faust compiler to build the amp and effects and will say thanks to : Julius Smith http://ccrma.stanford.edu/realsimple/faust/ : Albert Graef http://q-lang.sourceforge.net/examples.html#Faust : Yann Orlary http://faust.grame.fr/ ________________________________________________________________________ guitarix development team

12 years, 11 months

timing the processing of queues between engine and ui threads?

by Iain Duncan

Further to the conversation about Python to C++ ( with many helpful responses, thanks everyone! ). For my particular case, no drop outs is critical, and I really really want to be able to run multiple UIs on lots of cheap machines talking to the same engine over something (osc I expect). So I'm ok with the fact that user input and requests for user interface updates may lag, as the queue is likely to be really busy sometimes. I'm imagining: Engine thread, which owns all the data actually getting played ( sequences, wave tables, mixer/synth/effect controls, the works ) - gets called once per sample by audio subsystem ( rtaudio at the moment ) - does it's audio processing, sends out audio - loop by Magical Mystery Number 1: - get message off input queue describing change to a data point ( sequence or sample data ) - updates data point - loop by mystery number 2: - get message off 2nd UI queue requesting the state of a data point - sends back a message with that data to the requester done GUI thread - keeps it's own copy of whatever data is pertinent to that particular gui at that point - sends a bunch of requests if user changes the view - sends messages data change requests according to user actions Here's my question, how do I determine the magical mystery numbers? I need to make sure engine callback is always done in time, no matter how many messages are in the queue, which could be very high if someone is dropping in a sample of audio. By making the data point messages very simple, I hope that I'll have a pretty good idea of how long one takes to process. It's just a lock-get-write to simple data structure. But how much audio processing has happened before that point will be variable. Anyone have suggestions on that? Is the system clock accurate enough to check the time and see how much a sample period is left and make some safe calculation with headroom left over there? It is totally ok for the queue and the inputs to lag if the audio number crunching goes through a spike. suggestions most welcome. (including 'that design sucks and here's why') thanks iain

12 years, 11 months

Kobo II: Another song WIP

by David Olofson

Figured this might be of some interest to someone around here - and either way, it's all done on Linux, and it will be released on Linux. ;-) (Original announcement post at the end!) ------------------------------------------------------------------------- The whole game engine will probably go Free/Open Source eventually; older versions of parts of it already are. The synth engine will be JACKified and open sourced as soon as I get around to it! Going to support JACK in the game as well, as I use it on my devsystem all the time anyway. No idea if anyone will ever understand or care for this strange beast of a sound engine, but anyway... :-D For your amusement, here's the "lead synth" used for the theme and some other melodic features in the song: CuteSaw(P V=1) { !P0 sp, +P sp, w saw, p P, a 0, !er .05, !vib 0, !tvib .01 .rt wg (V + a - .001) { sp (vib * 6 + rand .01) 12 { -sp vib, +p (P - p * .8), +a (V - a * er), d 5 } 12 { +sp vib, +p (P - p * .8), +a (V - a * er), d 5 } +vib (tvib - vib * .1) } a 0, d 5 1(NV) { V NV if NV { vib .005, tvib .005, er .05, wake rt } else { tvib .02, er .02 } } 2(NP) { P (P0 + NP), p P } } Yeah, I was in a neurotically minimalistic kind of mood when I designed that language... But, it Works For Me(TM)! Less typing ==> quicker editing. ;-) (The original version of ChipSound, with a more assembly-like scripting language, was less than 2000 lines of C code. It's slightly below 4500 lines now, compiler included.) When playing a note, a voice with its own VM is started, and set to run this script. The VM runs in unison with the voice, alternating between audio processing and code execution. Thus, timing is sub-sample accurate, allowing the implementation of hardsync, granular synthesis and the like without specific engine support. Timing commands can deal in milliseconds or musical ticks, making it easy to implement rhythm effects, or even writing the music in the same language, as I've done here. Voices (microthreads) are arranged in a tree structure, where each voice can spawn any number of sub-voices it needs. Messages can be sent to these voices (broadcast or single voice), allowing full real time control. Oscillators currently available are wavetable (mipmapping, Hermite interpolating, arbitrary waveform length sampleplayers) and "SID style" S&H noise. It's possible to use arbitrary sampled sounds and waveforms, but so far I've only been using the pre-defined sine, triangle, saw and square waveforms, and the noise generator. ------------------------------------------------------------------------- Kobo II: Another song WIP ------------------------- "Yesterday, I started playing around with a basic drum loop made from sound effects from the game. The usual text editor + ChipSound exercise, using only basic waveforms. I sort of got caught in the groove, and came up with this: [...]" Full story: http://olofsonarcade.com/2011/11/06/kobo-ii-another-song-wip/ Direct download: http://www.olofson.net/music/K2Epilogue.mp3 -- //David Olofson - Consultant, Developer, Artist, Open Source Advocate .--- Games, examples, libraries, scripting, sound, music, graphics ---. | http://consulting.olofson.net http://olofsonarcade.com | '---------------------------------------------------------------------'

12 years, 11 months

Re: [LAD] prototyping callback based architecture in python?

by Kjetil Matheussen

Paul Davis: > On Wed, Nov 2, 2011 at 12:23 AM, Iain Duncan <iainduncanlists(a)gmail.com> > wrote: >> Hi, I'm working on an a project that I intend to do using the STK in a >> callback style, but am hoping I can prototype the architecture in python >> until I've figured out the various components and their responsibilities >> and >> dependencies. Does anyone know of any kind of python library ( or >> method? ) >> that would let me simulate the way callback based STK apps using RTAudio >> work? IE I want to have a python master callable that gets called once >> per >> audio sample and has a way of sending out it's results to the audio >> subsystem. >> >> I've found a bunch of python audio libs but it doesn't seem like they >> work >> that way, maybe I'm missing something? > > the obvious choices would be the python wrappers for JACK and/or > PortAudio. > > note, however, that the chances of using Python for per-sample > processing at low latency without intense dropouts are low. > Actually, since Python has an incremental garbage collector, it should in theory only require replacing the memory allocator with a real time memory allocator to make Python pretty hard real time capable. I think. I also think I remember someone using Python for real time sample by sample signal processing in Pd... Efficiency is another story though

12 years, 11 months

Communicating between python UI and C++ engine for real time audio?

by Iain Duncan

I looked into this about five years ago, but didn't get too far. Wondering if anyone on here has experience splitting apps up into: - realtime low latency engine in C++ using per sample callback audio ( either RTAudio or Jack or PortAudio ) - user interfaces and algo composition routines in python that either communicate with the engine by shared memory or over a queue Basically I want to be able to do the gui and data transforming code in Python whenever possible and allow plugins to be written to work on the data in Python I'm hoping to arrive at some sort of design that ultimately lets the engine act as an audio server with multiple user interface clients, possibly not even on the same machines, but definitely not on the same cores. If anyone has tips, war stories, suggestions on reading or projects to look at, I'd love to hear them. Thanks Iain

12 years, 11 months

Isolating predefined part of audio stream

by JohnLM

Hi! Well the subject is more technical term of "cutting my favourite show from radio stream". :D I guess the usual is to simply define the start and end times and trim the data to this range. Pretty straightforward, but in my case nor the start time, nor the show length is strictly known. My intention is to analyse the data (which isn't necessarily streaming) to detect signature of show in some way. Now the case-specific details: The show I'm interested in is readily recognizable on waveform already. (see this pic: http://home.venta.lv/s9_smedin_j/files/waveform.png ) The show has an "intro", so I can detect that by "sliding" correlator. The naive algorithm would be to use one or more sample chunks of this "intro" and to detect correlation spike when "sliding" and testing over some range of the larger audio data. I already started implementation of this, but haven't tested it yet. Maybe there is good implementation of this? Does phase shift between sampling positions in test chunk and data skews results significantly? The end (or duration) of show is a bit more complicated since it is highly variable. But as can be seen in picture, the end is quite visible on the waveform and the show has "spikey" waveform since it's all talking. But I have only very vague ideas how to detect this with a program. Any ideas on this? Maybe there is good programs, scripts or LADSPA/LV2 plugins that do something like this already, which I could use? Thanks in advance, JohnLM

12 years, 11 months

prototyping (wrong) communication between python and rt threads?

by Iain Duncan

Sounds like there are a number of options for me to look at and spend some time getting the communication right, I'm wondering if anyone can give me a suggestion on how I can start experimenting with object design *deferring* the question of getting communication *right*, while I study options. I'd like to come up with a well encapsulated API, wondered if anyone has ideas for what would 'sort of work' for now while I'm writing experimental code, but still be layered properly so that when it's time to examine the threading and timing issues in detail I can. or maybe this isn't possible? For example, what kind of queuing system would one suggest for just getting started where occasional blocking is ok? Does anyone use boost queues or is strictly a roll-your-own endeavor? Is planning on sending messages with an osc protocal realistic as a starting point? thanks iain

12 years, 11 months

prototyping callback based architecture in python?

by Iain Duncan

Hi, I'm working on an a project that I intend to do using the STK in a callback style, but am hoping I can prototype the architecture in python until I've figured out the various components and their responsibilities and dependencies. Does anyone know of any kind of python library ( or method? ) that would let me simulate the way callback based STK apps using RTAudio work? IE I want to have a python master callable that gets called once per audio sample and has a way of sending out it's results to the audio subsystem. I've found a bunch of python audio libs but it doesn't seem like they work that way, maybe I'm missing something? thanks! Iain

12 years, 11 months

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Linux-audio-dev