Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

4 participants
7502 discussions

by Iain Duncan

Hey folks, what is the easiest way to deal with midi input in a jack app? I'm confused by the difference in jack midi and alsa midi, because I have two midi inputs, one is a usb input, so it appears at a low level as an alsa device, but the other is the midi input on a firewire unit, and it appears as a jack midi device. I'd like to make sure that whatever I do is easy to port to other systems. Does it make sense to use portmidi or rtmidi to get input or should I stick to the jack api entirely? thanks Iain

13 years, 6 months

Scopes & RMS

by Harry van Haaren

Hey all, I've been writing a scope the last while, and I'm intrested in how other people have approached plotting the data. Currently I'm taking every 50th sample, and drawing a line from the previous sample to there, and so on. Not particulary neat. So is there a resample, or smoothing of the samples, or how does one plot a "smooth" waveform like Ardour / QTractor? My other question is about RMS, is calculating it in its literal sense best? Or perhaps only taking every 5th sample? Resampling the signal from 44.1 or 48k to say 11025? I'm not really sure which way to go. Cheers, -Harry

13 years, 6 months

Replaygain for video?

by Philipp

Hi there, I could use some advise. You may or may not heard of replaygain. It's reasonably widely used in consumer audio, but sometimes I wish it was available for video as well. By this I mean I wish it was available for the audio part of the video. Well, I need a programming project for a university course and this is just one of my ideas that I want to propose to my teacher and prospective teammates. In order to do this I'd like to narrow it down a bit further and especially want to find out whether I have the right idea of how it can be achieved. Scanning/tagging Since replaygain works on whole audio files I think I need to extract the whole audio track from the container. How easily this can be achieved I don't know. After that, the scanning process should work as with any audio file. Afterwards the calculated replaygain values have to be added to the metadata of the file. I have no idea how hard it is to add new metadata fields to video formats. Playback Video players need to be aware of those tags, read the metadata and scale the playback volume accordingly. This is probably not hard per se, but there are many players out there. However, I plan to start with a single player, even with a single file format, and go from there. Question 1: Is there anything better than replaygain that should be used instead? Question 2: Which player would be easiest to hack to add such functionality? Could it be a gstreamer plugin? mplayer? Question 3: How much work would it be? The project should be done in C++ if possible, otherwise C. Group size 2-4 Students, all rather new at C/C++ and rather inexperienced in general. Other ideas I have are in short: - A CLI (using readline) connection manager for jack audio/midi and alsa midi that can handle large numbers of ports. More detailed ideas exist thanks to Julien Claasen. - A simple but hopefully sane mplayer GUI - A new GUI for ecasound Another problem I might have is that most students in the course are Windows users, not sure whether I can go solo. Thanks for any advice, regards, Philipp

13 years, 6 months

Debugging environment for lv2 development.

by email.rafa＠gmail.com

Hi everyone. I'm starting to write a simple filter and I want to expose it as an lv2 plugin. My development environment is very simple right now: vim editor, gcc compiler, package the lv2 manually (will write a script for that in a day or two) and then load the plugin in ardour to test it. I've found this setup to be a bit unconfortable because once I load the plugin in ardour I don't know how to get debug information from it (print statements or breakpoints with gdb). What does a lv2 development environment typically looks like? what are you guys using? Thanks! -- Rafael Vega email.rafa(a)gmail.com

13 years, 6 months

Csound 5.15 release

by Victor Lazzarini

Apologies for cross-posting. ====================== We are pleased to announce the release of version 5.15. The sources are on the standard Sourceforge location (https://sourceforge.net/projects/csound/files/csound5/csound5.15/) as both zip and tar.gz Platform packages will follow shortly, and the manual on Friday. ==John ffitch ------------------------------------------------------------------------ Notes for 5.15 ============== New parser has been subjected to a great deal of work. It now has better checking of argument types and use, better diagnostics and increased functionality. We have only reached this stage in the last few days so we judge it prudent to leave the old parser as standard. We would be pleased if more users tried the new and gave the developers feedback. A major reorganisation means that there are many fewer plugins and most opcodes are in the base (about 1250 of them). A side effect of that is that leaving old plugins from an earlier release is a disaster, and so 5.15 will not load earlier plugins. The multicore system is now safe (ie maintains semantics) when zak, channels or table modification are made. New Opcodes: ftab2tab transfers between ftables and t-variables tab2pvs tsig - pvs conversion pvs2tab pvs - tsig conversion cpumeter-- not really new but now available in OSX (EXPERIMENTAL) ftresize and ftresizei allow resizing of existing tables. These will be permanent if the community feel they are useful. minmax opcodes hrtfearly, hrtfreverb opcodes New Gen and Macros: Code to allow GEN49 to be deferred [NB does not seem to work] Modified Opcodes and Gens: socksend and sockrecv no longer uses MTFU check and work on Windows mpulse changed so if next event is at negative time use the absolute value serial opcode now runs on Windows as will as Un*x out, out2, outq, outh, outo outx and out32 are now identical opcodes and will take up to as many arguments as nchnls. This replaces the current remapping of opcodes turnoff2 now polymorphic wrt S and k types (ie accepts instrumnet names) Utilities Bugs fixed: GEN42 fixed jacko: fixed a segfault removing the unused JackSessionID option doppler memory leak fixed transegr fixed in release mode when skipping most of envelope FLPack now agrees with manual max_k now agrees with manual hrtfreverb fixed atsa code now works on Windows in more cases tabmorph bug fixed fixed problem with user-defined opcodes having no outputs Various fixes to * ... */ comments System Changes: Various licence issues sorted Loris is no longer part of the Csound tree Memory leaks fixed If no score is given a dummy that runs for over 100 years is created All score processing takes place in memory without temporary files String memory now expandable and no size limitation #if #else #end now in new parser Adjustments to MIDI file precision in output On OSX move from Coreaudio to AuHAL Multicore now safe for ZAK, Channels and modifying tables New coremidi module Virtual Keyboard improved: 1) Dropdown for choosing base octave (the one that starts with the virtual key mapped to physical key Z). Default value is 5 which is backwards compatible. 2) Shift-X mappings which add two octaves to X mappings for a total of 4 octaves playable from the physical keyboard (starting from selected base octave). 3) Control-N / Control-Shift-N mappings to increment / decrement slider for control N. 4) Mouse wheel now controls sliders. tsig type for vectors tsigs and fsigs allowed as arguments in UDOs API: Minor version upped Internal: Very, very, very many! Dr Victor Lazzarini Senior Lecturer Dept. of Music NUI Maynooth Ireland tel.: +353 1 708 3545 Victor dot Lazzarini AT nuim dot ie

13 years, 6 months

how to store deferred events in a real time audio app?

by Iain Duncan

Hi, I'm sure others have tackled this and have some wisdom to share. My project is principally a monosynth step sequencer. This is nice an simple to do in real time because resolution is very limited and there can be only one note per track. So step sequenced note data is stored in simple multi-dimensional arrays, making reading and writing very easy, and messaging simple between audio and gui threads. However, I would like to add the ability for the user to send a message and have it get executed later, where later gets figured out by the engine ( ie on the top of the next 8 bar phrase ). To do this, I need some way of storing deferred events and having the engine check on each step whether there were any deferred events stored for 'now'. I can think of a few ways to do this, and all of them raise red flags for a real time thread. - I could use a hash table, hashed by time, with a linked list of all the events for that time. The engine looks up the current time and gets all the events. I don't know much about hashing so I'd prob just use Boost, is that a bad idea? - I could make a linked list of all deferred events and iterate through them checking if time is now. There wouldn't be any hashing, but maybe this list would be really big. Anyone have any suggestions for how to safely do the above or some better alternative? thanks! iain

13 years, 6 months

Re: [LAD] Scopes & RMS

by harryhaaren＠gmail.com

On , Tristan Matthews <le.businessman(a)gmail.com> wrote: > You might find some inspiration in sndpeek: > http://soundlab.cs.princeton.edu/software/sndpeek/ Definatly! Thanks, got linux-alsa compiling, but linux-jack is segfaulting in RtAudio::startStream()... debuggin atm :) Cheers, -Harry

13 years, 6 months

MediaInfo API

by Alfs Kurmis

Hi Experts. I wanna gotta some info about my (mainly) sound files. Samplerate, Bitrate, VBR, Channels, Bits, Samples, Tag info ... and put em all in my C program variables or structure. Seems that MediaInfo support many formats, and it is really good. Can somebody gimme examples how to access sound/media files with MediaInfo API ? What exact #include's must be used ? How to compile ? Evtl. what include and lib dirs should be specified in GCC command line ? gcc -I... -L... -lmediainfo What is difference between /usr/include/MediaInfo and /usr/include/MediaInfoDLL ?? If i wanna use MediaInfo API, but not MediaInfo itself, is it enough when i emerge media-libs/libmediainfo or anyway i must emerge media-libs/libmediainfo AND media-video/mediainfo ? eix mediainfo [I] media-libs/libmediainfo Available versions: 0.7.45 ~0.7.48-r1 ~0.7.49 ~0.7.50 {curl doc mms static-libs} Installed versions: 0.7.45(12:05:57 AM 12/04/2011)(-curl -doc -mms -static-libs) Homepage: http://mediainfo.sourceforge.net/ Description: MediaInfo libraries [I] media-video/mediainfo Available versions: 0.7.45 ~0.7.48 ~0.7.49 ~0.7.50 {curl mms wxwidgets} Installed versions: 0.7.45(12:08:18 AM 12/04/2011)(-curl -mms -wxwidgets) Homepage: http://mediainfo.sourceforge.net Description: MediaInfo supplies technical and tag information about media files I fond this, but it does not help :( http://mediainfo.sourceforge.net/de/Support/SDK/Quick_Start#Example Tnx in advance @ all ----

13 years, 6 months

Santa LACus (your Xmas Linux Audio Conference 2012 reminder)

by Bruno Ruviaro

Hi all, Just a friendly reminder that JANUARY 11 is the deadline for all submissions to the Linux Audio Conference (LAC 2012), which will take place at CCRMA (Stanford, California) in April 2012!http://lac.linuxaudio.org/2012/ Santa LACus wishes a great paper-and-music-submitting holiday to all! Ho, ho. Bruno - - - - - - - - - LAC 2012: the Linux Audio Conference - Call for Participation April 12-15, 2012 @ CCRMA, Stanford University http://lac.linuxaudio.org/2012/ [Apologies for cross-postings] [Please distribute] Online submission of papers, music, installations and workshops is now open! On the website you will find up-to-date instructions, as well as important information about deadlines, travel, lodging, and so on. Read on for more details! We invite submissions of papers addressing all areas of audio processing based on Linux and open source software. Papers can focus on technical, artistic or scientific issues and can target developers or users. We are also looking for music that has been produced or composed entirely or mostly using Linux and other Open Source music software. The Deadline for all submissions is January 11th, 2012 The Linux Audio Conference (LAC) is an international conference that brings together musicians, sound artists, software developers and researchers, working with Linux as an open, stable, professional platform for audio and media research and music production. LAC includes paper sessions, workshops, and a diverse program of electronic music. The upcoming 2012 conference will be hosted at CCRMA, Stanford University, on April 12-15. The Center for Computer Research in Music and Acoustics (CCRMA) at Stanford University is a multi-disciplinary facility where composers and researchers work together using computer-based technology both as an artistic medium and as a research tool. CCRMA has been using and developing Linux as an audio platform since 1997. http://ccrma.stanford.edu Stanford University is located in the heart of Silicon Valley, about one hour south of San Francisco, California. This is the first time LAC will take place in the United States. http://www.stanford.edu We look forward to seeing you at Stanford in April! Sincerely, The LAC 2012 Organizing Team

13 years, 6 months

ANN: pd-faust 0.1

by Albert Graef

pd-faust is my latest stab at making the integration of Pd and Faust as simple and painless as possible. For those of you who've used my utilities for Faust and Pd before, pd-faust integrates the functionality of faust2pd and pure-faust into a collection of Pd objects written in the Pure programming language. It also sports the following major improvements over faust2pd: - Reload Faust modules at runtime and have the Pd GUI of the Faust dsp regenerated automatically and instantly. - The metadata in Faust programs is interpreted to adjust the GUI layout in a faust2pd-compatible fashion. - MIDI/OSC controller mappings are provided for the 'midi' and 'osc' metadata tags in the Faust source. - Built-in MIDI sequencer and OSC recorder which syncs MIDI and OSC playback and provides an OSC-based controller automation facility for all Faust dsps in a Pd patch. So in other words it's the Swiss army knife for Faust development in Pd. ;-) If you're into Faust and Pd, I hope that you'll find it useful. Bug reports and other feedback are appreciated. A brief overview is available here: http://code.google.com/p/pure-lang/wiki/Addons#pd-faust The obligatory screenshot: http://wiki.pure-lang.googlecode.com/hg/pd-faust.png Detailed documentation (including installation information): http://docs.pure-lang.googlecode.com/hg/pd-faust.html pd-faust is compiled to a native Pd object library which can be loaded with Pd's -lib option as usual. Note that besides Pd, Faust and pd-faust itself you'll also need the Pure interpreter and a couple of Pure addon packages to build and run this software. Please check the documentation linked to above for details. All the Pure-related downloads can be found on the Pure website: http://pure-lang.googlecode.com For your convenience, here are the direct download links for the required packages from the Pure project (source tarballs): http://pure-lang.googlecode.com/files/pure-0.50.tar.gz http://pure-lang.googlecode.com/files/pd-faust-0.1.tar.gz http://pure-lang.googlecode.com/files/pd-pure-0.15.tar.gz http://pure-lang.googlecode.com/files/pure-faust-0.6.tar.gz http://pure-lang.googlecode.com/files/pure-stldict-0.2.tar.gz You'll also need a recent version of Pd (0.43 has been tested) and Faust from git (0.9.45 and 2.0.a3 are both known to work fine). Happy holidays, Albert P.S.: Sorry for the excessive cross-posting, but the nature of this project which interfaces between three different environments, each with their own communities, made this seem appropriate. -- Dr. Albert Gr"af Dept. of Music-Informatics, University of Mainz, Germany Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikinformatik.uni-mainz.de/ag

13 years, 6 months

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Linux-audio-dev