Linux-audio-dev November 2011

linux-audio-dev@lists.linuxaudio.org

60 participants
38 discussions

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 | '---------------------------------------------------------------------'

14 years, 1 month

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

14 years, 1 month

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

14 years, 1 month

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

14 years, 1 month

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

14 years, 1 month

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

14 years, 1 month

QasMixer 0.15 released

by Sebastian H.

QasMixer version 0.15 is now available. QasMixer is an ALSA mixer with a size adaptive Qt GUI. Changes: * New simpler device selection view * New user setting: Mixer device on startup * ALSA configuration view moved to a separate application: QasConfig * Localizations moved to a separate package: qasmixer-l10n * Default fallback translation * Code merges and cleanups * Version code shortened to two numbers instead of three Homepage with more information http://xwmw.org/qasmixer Project page http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

14 years, 1 month

Kernel RT patches ?

by Parisson

Hi all ! Do anyone know where are the RT kernel patches since the kernel.org reinstall ? They used to be there: http://www.kernel.org/pub/linux/kernel/projects/rt/ Cheers, Yomguy

14 years, 1 month

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Linux-audio-dev November 2011