Linux-audio-dev

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[linux-audio-dev] ANNC: Audiality 0.1.0 Alpha Release

by David Olofson

Audiality 0.1.0 Alpha Release ----------------------------- Audiality is an audio engine originally designed for playing music and sound effects in games. Scalability has been a major priority; the current version will run on very low end Pentium systems, but also scales to audio quality comparable to that of "real" hardware synths. (That is not to say that Audiality can compete with the feature set of those synths... Yet.) The engine builds as a shared or static library, or both if possible. (Not sure if it builds for Win32, but it did not long ago...) Two simple test programs (one for the scripting engine and one for the whole thing) and a bunch of demo sounds and songs are included. That minimalistic plugin API, and the timestamped event system are in there. Should be easy enough to extract for anyone who wants either or both for something else. Both are LGPL, just like *most* of the rest of the engine. (Read on...) Licensing terms: I intend to release all of the code under the LGPL. However, there are still traces of the GPL code that "inspired" the creation of this engine. (I didn't really mean to turn those four little files it this beast... *heh*) That code was written by Masanao Izumo (maintainer of Timidity++), whom I have not been able to contact. Therefore, the engine as a whole has to be considered GPLed, until this issue can be resolved one way or another. Feel free to find Masanao for me, contribute LGPLed versions of the offending files, or whatever. This might all sound silly to you, but I can't say for sure how much of Masanao's code and design is still in there, and I don't feel like stealing and sublicensing stuff - be it 5 lines of code. No big deal - it's still Free, and I'm not planning on making money on Audiality as a product anyway. The motivation for LGPL is just that I want people to also be able to use Audiality for what it was originally intended: Game audio. The files (temporary location): http://www.seaside.se/~sea1154a/audiality-0.1.0.tar.bz2 (359 kB) http://www.seaside.se/~sea1154a/audiality-0.1.0.tar.gz (467 kB) To install: * Unpack into whatever place you want sources * In the top dir: * ./configure * make * As root: (or add options for user install) * make install * In the test subdir: * ./configure * make * ./eeltest (does the output make sense? ;-) * ./atest (enjoy! :-) If the lib won't compile for reasons related to ALSA, I'm afraid you have to './configure --disable-alsa', since the ALSA rawmidi support is for 0.5 only... Sorry. (I'd happily accept a patch - including audio support, as I'm using ALSA 0.9 myself now! ;-) Requirements: * SDL 1.2 (http://www.libsdl.org) * Autoconf 2.55 * Automake 1.7.1 * Libtool 1.4.3 (Though you can probably build with slightly older versions of autotools. I just grabbed the latest tarballs from the GNU site.) Some working features (or so I hope!): * Voice mixer with * 2 stereo sends with independent bus selection * Smooth ramping of sends - no zipper noise * Scalable resampling quality: * Nearest sample * Linear interpolation * Adaptive oversampling * Cubic interpolation * Native support for multiple waveform formats: * 8 bit integer * 16 bit integer * Mono as well as stereo * Internal mixer with * Up to 16 busses * Up to 8 insert effects per bus * Full send array... * ...that can send to other busses * ...before the first insert * ...after each insert * Two way kewl insert effects (not really): * Feedback delay "reverb" * Basic limiter * Off-line waveform rendering package: * Driven by a simple scripting language * Can be driven through a handy C API. * Supports various sample formats: * 8 bit integer * 16 bit integer * 32 bit float * Mono as well as stereo * 7 modulation targets, each with: * 1 envelope generator * 1 LFO * Oscillator operator with: * 6 mixing/combination modes: * ADD * MUL * FM * FM_ADD * SYNC * SYNC_ADD * 13 waveforms/tone synthesisers * SINE (with FM if you like) * HALFSINE (half wave rectified) * RECTSINE (full wave rectified) * PULSE (PWM capable) * TRIANGLE (mod => saw-tri-saw) * SINEMORPH (recursive FM) * BLMORPH (sin, saw, square) * BLCROSS (sin, saw, square) * NOISE (SID style; NG + S&H) * SPECTRUM (additive Df) * ASPECTRUM (multiplicative Df) * HSPECTRUM (harmonic) * AHSPECTRUM (pseudo harmonic) * Basic filters: * Low pass, 6 dB/oct * High pass, 6 dB/oct * Basic resonant filters (oversampled): * Low pass, 12 dB/oct * High pass, 12 dB/oct * Band pass, 12 dB/oct * Band reject/notch, 12 dB/oct * Band boost/peak, 12 dB/oct * Real time MIDI input (ALSA 0.5 or OSS) * MIDI file loader/player * Audio output through * SDL (and whatever that supports) * OSS * OSS/polling (only for broken debuggers...) * WAV and "RAW" audio file loading. (SDL) The main goal for the future is to extend the scalability of Audiality well into the range of serious home and professional studio use. The idea is to provide total control, a multitude of *useful* features, tools for fast and effective creation of original sounds, and excellent audio quality. I'm serious about music, and I'm not going back to programming hardware synths with little LCD displays or crappy editor software. Audiality is meant to replace the hardware synths I have, and eliminate the need for future purchases in that area. (I'd much rather spend the $$$ on audio interfaces, CPU power and perhaps, good sample libraries.) It's a tough goal, but I have been known to produce hundreds of lines of working code a day (*), when strongly motivated. :-) (*) Calculated from Audiality statistics during the period when most of it was written. Around 300 lines/day, IIRC. Some TODOs: * Separate audio/MIDI drivers from the core! * Implement a serious API for the engine library! * Audio I/O through JACK, ALSA and possibly LADSPA. * ALSA sequencer support. * OSC? DMIDI? Whatever works. * Hosting LADSPA plugins as inserts. * Support multichannel I/O directly through the mixer. * Make the FX plugin API use the event system. * Add support for float32 processing and plugins. * Direct-from-disk waveform playback. (EVO?) * Switch from interpreter to compiler + VM. * Support scripting in the real time engine. * Script editor with GUI panels for known constructs. * Full GUI editor that generats scripts from scratch. * Integrate the off-line and RT synths totally. * Split voice mixers into "resampler" and "send unit" * Support voice insert plugins? Maybe, maybe not... * Implement better resampling in the voice mixers Have fun! //David Olofson - Programmer, Composer, Open Source Advocate .- Coming soon from VaporWare Inc...------------------------. | The Return of Audiality! Real, working software. Really! | | Real time and off-line synthesis, scripting, MIDI, LGPL...| `-----------------------------------> (Public Release RSN) -' .- M A I A -------------------------------------------------. | The Multimedia Application Integration Architecture | `----------------------------> http://www.linuxdj.com/maia -' --- http://olofson.net --- http://www.reologica.se ---

21 years, 9 months

[linux-audio-dev] ANN: swh-plugins 0.3.2

by Steve Harris

Sorry, to release again so soon, but I f**ked up and forgot to commit the fix for the duplicate UID. Erk! Thanks to Nathaniel Virgo for noticing, and reporting a bug in the chebstortion: for some input (eg. pure sinewaves) it produces odd clicks in the output, I've seen this elsewhere, but for the life of me I can't remeber where, or what caused it. Someone else reported a bug in the multiband EQ: freqeuncy dependent wideband distortion, but I cant reproduce it, so more reports about this would be good. To make up for it, I've just added a "Bode frequency shifter", an interesting synthesis tool, as requested by Matthias. I've wanted one for years, and didn't think they would be so simple to implement digitally. It produces some aliasing with wideband input, I can fix it later if enough people request it. Anyway, http://plugin.org.uk/releases/0.3.2/ - Steve

21 years, 9 months

[linux-audio-dev] OT: amp modeling

by Will Benton

Hello, all. This is perhaps totally OT, but with all of the recent conversations about amp-modeling, I was wondering if anyone had any pointers to how to build a physical, analog, amp-modeling stompbox or similar (or would it just be cheaper to buy a SansAmp?) Any direction to get started in this arena would be helpful. thanks, wb -- Will Benton http://www.cs.wisc.edu/~willb "YOW!! That was one of my BEST NIGHTMARES!!"

21 years, 9 months

[linux-audio-dev] dev help requested

by ljp

hello, I am writing a audio recording app for linux pdas, of the zaurus sort. I am running into a problem with switching sampling rates on the target device, and need some help to find a work around for a buggy driver, which I have no control over, as it's shipped with the device, and is not a kernel module. When testing on my x86 desktop, and various audio cards, everything works as planned. But when running on the device (arm based processor, tc35143 audio chip - heh for whatever its worth), when I request a new sampling rate, the driver/device seems to change, no errors are reported. I can ask the driver what the rate is and it reports what I requested, but the actual amount of data received is at the previous rate. I can close the app and get the actual intended rate. I have tried doing a fork, in hopes maybe the driver was holding process information. No joy. I don't want to have to resample the input, as it seems a waste of cpu, and as a musician... blasphamy. Any hints, or suggestions would be welcomed. thanks, ljp -- My cat's a debugger.... Potter, Lorn, "ljp" core member / Web Administrator Project OPIE- the Open Palmtop Integrated Environment http://opie.handhelds.org | http://www.opie.info (german) | http://www.opie.us IRC: irc.freenode.net #opie #opie.de llornkcor(a)handhelds.org

21 years, 9 months

[linux-audio-dev] Re: Intel C Compiler & RedHat 8.0 , Pentium 4 FPU performance

by Andrew Burgess

>I will check the effect of sse on my plugins as they are generally less >ram hungry, but I dont have a gcc3 machine around at the moment. Just FYI, the redhat gcc3 rpm installs the executable as gcc3 so it happily coexists with older gcc versions. You just edit the makefile to use gcc or gcc3.

21 years, 9 months

[linux-audio-dev] ANN: swh-plugins v0.3.1

by Steve Harris

http://plugin.org.uk/releases/0.3.1/ Includes a fix for a serious problem where two plugins ended up with the same UID, 0.3.0 users should upgrade. One new plugin, "chebstortion" a synthetic distortion effect that came out of the amp modelling discussions. It doesn't really bear any resemblance to any analogue effect living or dead, but its interesting anyway. Bugfix to the analogue osc, it now doesn't (shouldn't) distort horribly. Bugfix to the chorus, it now uses less CPU and its load is constant, so it will work OK in RT systems. Now builds on Linux PPC (though still not OSX). Installs RDF metadata describing the plugins. Now has freshmeat entry, http://freshmeat.net/projects/swh-plugins/ (someone requested this, sorry, can't remeber who). - Steve

21 years, 9 months

[linux-audio-dev] finding info on sound cards

by iain

For a couple of projects I need to be able to find out information about the sound cards installed on the system, but as I need to run on lots of different systems I need it to be fairly portable. Before I start writing something to do it, I figured I'd check to see if there's anything already in progress that can do the same. What I need is to be able to get a list of sound cards on the system, and information about them (like the names), and then to get a list of the channels each sound card has and information on them (current level, name (if it has one), if it's a record source etc), and finally to be able to control the channels, change the level, change if it's the current record source. Is there a library that can do this all available anywhere, or will I have to write it myself? Would it be a useful thing to have in a library? iain -- "The 'developed' nations gave to the 'free market' the status of a god, sacrificing to it their farmers, farmlands, and communities, their forests, wetlands and praries, their ecosystems and watersheds. They had accepted universal pollution and global warming as normal costs of doing buisness." - Wendell Berry

21 years, 9 months

[linux-audio-dev] ANN: RTMix 0.52 version available

by Ivica Bukvic

Hello all! A new version of RTMix is available at the same old place: http://meowing.ccm.uc.edu/~ico/ (use html entrance for low-bandwith version -> cpu's -> rtmix download) New improvements include: *Fixed networking seg-faults in Qt3 *Finished network code *Removed all kde-libs dependencies (I overlooked a couple in the configure script itself, sorry about this) *Several minor bug-fixes *Fixed all other known seg-faulting bugs *Added a couple of networking tutorials Enjoy! Ivica Ico Bukvic, composer, multimedia sculptor, programmer, webmaster & computer consultant http://meowing.ccm.uc.edu/~ico/ ============================ "To be or not to be" - Shakespeare "To be is to do" - Socrates "To do is to be" - Sartre "Do be do be do" - Sinatra "2b || ! 2b" - ? "I am" - God

21 years, 9 months

[linux-audio-dev] Announce : Secret Rabbit Code (aka libsamplerate)

by Erik de Castro Lopo

Hi People, This is the first announcement of Secret Rabbit Code: http://www.mega-nerd.com/SRC/ Secret Rabbit Code is a library for doing Sample Rate Conversion on audio. The web page has the full spiel of what it does. The source code tarball has two demo programs: sndfile-resample - A program which can perfrom sample rate conversion on a given sound file. varispeed-play - A program which plays a given sound file in a loop. During play, the speed of the playback is continuously varied. Lots of fun on drum loops and full mixes. This currently runs only on Linux/OSS (probably also ALSA OSS emulation). Work is being done on ports to MacOSX, Solaris and Win32. At the moment the rabbit code is known to compile on Linux and MacOSX. Win32 and Solaris support is coming RealSoonNow (tm). Enjoy, Erik -- +-----------------------------------------------------------+ Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid) +-----------------------------------------------------------+ Everyone seems to assume that the current system in America is capitalism. I beg to differ. True capitalism does not involve false advertising, distribution cartels, or political lobbying for special advantages in the market. How can you call Microsoft or the RIAA capitalist, when their main business is interfering with a free market? Some of us would like to see a *return* to capitalism in this country. - Jim Flynn on Linuxtoday.com

21 years, 9 months

[linux-audio-dev] Behaviour of write() in blocking/non-blocking mode

by Greg Ward

Hi all -- I'm making my first forays into OSS DSP programming with an eye to improving the standard linuxaudiodev module included with Python. (And, of course, to using that module for my own ends!) I'm rather confused by the semantics of write() in blocking and non-blocking mode. Specifically, in the default mode (blocking, no SNDCTL_DSP_NONBLOCK ioctl() call), it appears that write() will happily write the entire buffer passed to it, and not return until almost all of the data has been played. (I assume the last dribble is in a lower-level [hardware?] buffer, so when write() returns there's still a little bit of sound to be heard.) However, in non-blocking mode, write() plays roughly 64k or 128k (depending on the hardware device used) and returns immediately. This is more like what I was expecting -- I guess the behaviour in blocking mode is a bit surprising. Is it supposed to be like this? More importantly, can I *rely* on write() in blocking mode always consuming the entire buffer passed to it? BTW, this is with Linux 2.4.20-rc1; the two audio devices are a SoundBlaster Live using the OSS emu10k1 driver, and a Stereo-Link external USB audio device using, you guessed it, the USB audio driver. Thanks -- Greg -- Greg Ward <gward(a)python.net> http://www.gerg.ca/ I hope something GOOD came in the mail today so I have a REASON to live!!

21 years, 9 months

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