Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7501 discussions

by Florian Schmidt

This is totally alphabetagamma = unfinished and unpolished software. Release early - release often is my guide here ;) Here's the summary: ProcessPP is the "realisation" of a rather silly idea: A livecoding environment for C++. As C++ is not interpreted, but compiled, this indeed does sound like a really silly idea. But the twist is that while “true” livecoding cannot be done, it can be at least made as easy as possible to try out new code snippets. ProcessPP consists of three distinct parts - procespp: A host application that can load shared objects and execute the code contained in them - qprocesspp: A Qt GUI application which serves as an editor for hacking away functions and to compile them easily into a shared object. Then it tells processpp to load and run the shared object - libprocesspp: A shared library which is the “glue” between processpp and user generated shared object files. Right now it only containes code for the shared object to find out stuff like samplerate, buffersize, number of input/output channels, etc.. For illustrative purposes here are the steps involved in making noise: - File->Open from Template (choose process.template.cc) - Add some code in the inner loop to make noise - Process->Run - Hear the noise!!! If the above don’t work, take a look at the log in the lower half of the main window. Often some slight tweaks to the config file (~/.config/Ugh!/process++.conf) need to be made. Get it here: http://tapas.affenbande.org/wordpress/?page_id=91 Regards, Flo -- Palimm Palimm! http://tapas.affenbande.org

16 years, 9 months

[OT]: how to cast this damn pointer right?

by Julien Claassen

Hello! Sorry, for this, but my skills left me to unskilled... AGAIN... :-( I'm working with pthreads and now I like to hand over mains argv to the thread run_me. int main(int argc, char *argv[]); void *run_me(void *param); How to properly use param as a char * [] inside run_me? Thanks forany help! Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

16 years, 9 months

[ANN] jack_capture V0.9.30 and Rollendurcmesserzeitsammler V0.0.7

by Kjetil S. Matheussen

Download from http://www.notam02.no/arkiv/src/?M=D jack_capture ============ jack_capture is a program for recording soundfiles with jack. Its default operation is to capture whatever sound is going out to your speakers into a file. (But it can do a number of other operations as well...) Note: This version includes Hermann Meyer's jack_capture_gui2 program. "jack_capture_gui2" is a nice graphical frontend for for jack_capture with lots of options. Many thanks to Herman for the contribution. Changes 0.9.23 -> 0.9.30: *Added Hermann Meyer's jack_capture_gui2 program. jack_capture_gui2 is a nice graphical frontend for for jack_capture with lots of options. Many thanks to Herman for the contribution. *Don't exit in case port is not found. *Print runtime warning and error messages on top of the console to avoid printing the console meter yet another time. (it's much prettier also) *Fixed a bug that could cause (and especially after the switch from calloc to my_calloc apparantely) segfault when specifying --port more than once. Thanks to Peder Hedlund for spotting the bug. *Print error instead of segfaulting when a specified jack port does not exist. *Removed -g option and changed -O0 to -O2. (Oops, don't know how long that's been there) *Make sure the stop semaphore is initialized before it might be called. *Changed the --recording-time / -d option to record exactly the correct number of frames. (The format for the option is still in seconds though). This fixes the problem where the wall clock and the soundcard clock drifts apart. *Always increase the buffer size with 2 seconds when more than than half the buffer is used, unless maximum buffer size is reached. *Added the --maxbufsize / -MB option which sets maximum buffer size. Default value is 40 seconds. *Decreased the default buffer size from 20 to 10 seconds. *Changed internal data representation from lockless ringbuffer to lockless lifo and fifo stacks. Unmodified lifo/fifo code taken from midishare. (Copyright Grame 1999-2005) Rollendurchmesserzeitsammler v0.0.7 ------------------------------------ The Audio Rollendurchmesserzeitsammler is a conservative garbage collector especially made for running inside an audio DSP thread. 0.0.5 -> 0.0.7 * Cleaned up source a bit. * Fixed a bug in "tar_entering_audio_thread" which caused it to return false if currently copying a different heap. * Cleaned up the critical section handling between the DSP thread and the sweep thread. (it was really messy)

16 years, 9 months

Writing a library?

by Darren Landrum

I've been looking around for a library to read and write SFZ files, which is an open sampler format released by Cakewalk: http://www.cakewalk.com/DevXchange/sfz.asp Finding none, I thought I might try my hand at writing a library for this myself, as there is no embedded wave information like with Gig files. SFZ is simply a text file to be parsed. Now, I know about writing a good header file, and its associated class, and all that, but I have no knowledge of how to write it as a dynamic library. Google searches on every possible permutation have been worthless to me as well. I would prefer to write it in C++, as that's what I know, and even then, not too well, hence why I thought I'd start with something simple like parsing a text file. If anyone has any advice, recommendations, or ideas, I'll happily listen and learn. I have yet to think too much about how the data will be stored in the class, and what methods to make available to access it, so if anyone knows any best practices there, I'd really like to know. Consider this a feeler post. I'd ultimately want this for a future project, which you can guess at by now. Thank you for the help! Regards, Darren Landrum

16 years, 9 months

CUDA

by Jens M Andreasen

I am doing some preliminary testing of CUDA for audio, Version 2 (final) has been out for a couple of days, and this is also what I am using. In order to get anything done, one will always have to do something else first, here: Get some data transferred to the board. Surprisingly this appears to be ten times harder than getting the data back, at least for very small data sets representative of a bunch of on/off events or a milliseconds worth of samples. For 1024 bytes the transfer will take about 0.2 ms. That is 20% of the available time if we use a time granularity of a single millisecond. OTOH this appears to be mostly an intial constant, (much) more data can be transferred in the exact same amount of time if that is what is needed. Now my card (8400GS) is neither the latest nor greatest and I therefore wonder if anybody with better equipment is experiencing the same phenomenon? This card does not support aynchronous transfers, which otherwise might have been the thing to use here. mvh // Jens M Andreasen

16 years, 10 months

hdspm.c compile error

by peter boehm

hello, i recently purchased a madiface with express card. trying to compile a debian-source kernel 2.6.26-4, with replaced hdspm.c and hdspm.h, following instructions on http://wiki.linuxproaudio.org/index.php/Driver:hdspm i get following compile error: ERROR: "__muldf3" [sound/pci/rme9652/snd-hdspm.ko] undefined! ERROR: "__divdf3" [sound/pci/rme9652/snd-hdspm.ko] undefined! ERROR: "__fixdfsi" [sound/pci/rme9652/snd-hdspm.ko] undefined! ERROR: "__adddf3" [sound/pci/rme9652/snd-hdspm.ko] undefined! ERROR: "__floatsidf" [sound/pci/rme9652/snd-hdspm.ko] undefined! i tried to get more info on the net about this errors but did not find usefull hints. any help would be appreciated! peter

16 years, 10 months

patchage

by Fons Adriaensen

Since returning from holidays (i.e. 10 days) I've been running patchage as my visual interface to Jack. Never managed to install it from source (there was always something missing or at least one compile error), but recently discovered the CCRMA package (thanks Fernando !). For the sort of things I'm doing, usually involving a lot of interconnected Jack clients, it provides a much clearer picture than qjackctl, and I like that a lot. Some suggestions / feature requests: - Include the window scroll offsets in .patchagerc - Make sure new apps are allways displayed in the visible part of the window. - Some way to select and connect groups of adjacent ports (I'm using 64ch soundcards, and most connections are multichannel). - Optionally collapse such sets of ports into a single visual object. Long term: - Multiple tabs, each one controlling a Jackd on a remote headless machine. This will require splitting the app into a server and client with a network connection in between. Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia O tu, che porte, correndo si ? E guerra e morte !

16 years, 10 months

Coding 96kHz 24bit flac material to 16bit/44.1 mp3

by Carl-Erik Kopseng

Hey, I thought about extending MP3FS, a user level file system that shows flac files as mp3s to user-space programs (see mp3fs.sf.net), to make it work with my 96kHz/24 bit music rips. MP3FS uses liblame and libflac on the inside, but only converts standard 44.1kHz/16bit files at the moment. Looking at the not-to-well-documented lame library I (think) that Lame only support sample rates up to 48kHz, so I would need to convert the samplerate and bitrate through the use of an other library. I finally found libsndfile today, but thought I might hear with the expertice (that's you), if it should work (if it supports what I want to do),or if I should use something else. I would use libsndflac to convert 24 bit/96kHz and 16 bit/44.1kHz flac files to 16 bit/44.1 kHz uncompressed audio, and then use liblame to convert this to mp3. Regarding the downsampling I would like to know if I would get any funny artifacts when downsampling 96kHz material to 44.1kHz (not even division). Would I be better of to convert to 48kHz for 96kHz material? I do not know much about lame, mp3 encoding, or audio development apart from the basics, so guide me into safer waters if I have drifted into unknown waters here ;) br Carl-Erik Kopseng

16 years, 10 months

Job vacancy

by Fons Adriaensen

This may be of interest to some UK LADs: (From Dave Malham on the sursound list) http://www.jobs.ac.uk/jobs/UD613/Systems_Administrator_Research_Support_Pro… -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia O tu, che porte, correndo si ? E guerra e morte !

16 years, 10 months

Job offers at Barcelona Media: 3D audio developers

by Pau Arumí

Dear All, The following position may be of interest to you. Please forward to anyone interested. Apologies for double posting. ==== About the Barcelona Media Audio Group ==== Fundacio Barcelona Media Universitat Pompeu Fabra is a research centre created to foster the competitiveness of the Catalan and Spanish media and communication industry through innovative research activities and projects. BM promotes technology generation and development; research and creativity; transfer of research results to industry; promotion of the research results to society at large; training in all areas of communication; and social awareness of the communication industry in a culture of innovation. The Audio Group research embraces the whole chain of audiovisual productions, focusing specially on 3D surround sound technologies, from capturing, to postproduction, to exhibition. Two main general goals are to automatize the workflow (by automatic audio adaptation to given 3D scenes), and to make it easily adaptable to any final exhibition system (surround 5.1, 7.1, 22.2, binaural or 3D stereo, etc.). One strong line of research of the group is the reproduction of acoustic fields in 3D virtual environments, using computer simulations to predict what any source would sound like in a given virtual world. The group applies and improves the algorithms of Finite-Differences in Time-Domain for low frequencies, Ray-Tracing for high frequencies. Such technologies are then integrated in real-time, interactive multimedia systems. Audio group home page: http://www.barcelonamedia.org/linies/10/en ==== Profile ==== We are looking for one or more experienced software developers. The candidate should be self-motivated, results-oriented, and hard-working. The candidate should preferably have a degree on Computer Science, although other profiles might be taken into account. ==== Required skills ==== * Software-engineering techniques and methods for developing large software systems (design patterns, agile methodologies, version control systems, etc. ). * Real-time programming techniques including lock-free and multi-threading. * Programming languages: C, C++, Python. * Operating systems: GNU/Linux and Mac OS X. * A keen sense for the aesthetics of code, documentation, and user interfaces. Thoroughness in all aspects of software development. * The ability and willingness to interact in a team, using agile methodologies. Not required, but valuable skills: * Real-time multimedia environments (PureData, Max/MSP, Supercollider, CLAM, etc.). * Plugin architectures: LADSPA, LV2, VST, Audio Units, etc. * Knowledge of common protocols such as OSC, MIDI, etc. * Knowledge in digital signal processing and/or acoustics * 3D modeling: Blender or Maya or 3D studio * Qt * Scons ==== What we offer ==== We offer an opportunity to work in creative projects in the field of 3D audio for media productions, with applications ranging from 3D digital cinema, to sports broadcasting, and videogames. Side opportunities: perform strategic research in a new promising domain, working in a small-medium multidisciplinary team, collaborate with people from the industry and from other academic research groups, to establish contacts with the international audio research community through the attendance to international conferences... ==== How to apply ==== To apply, send email to jobs(a)barcelonamedia.org / cc: toni.mateos(a)barcelonamedia.org, pau.arumi(a)barcelonamedia.org with the subject "3D Audio Jobs" * A brief presentation letter stating your interest in the offer. * A CV * Optionally, code samples (non open-source samples will be treated as confidential) ==== More background about Barcelona Media ==== BM grew from the Communication Station set up by Universitat Pompeu Fabra in 2001. It is a member of the Catalan and Spanish network of Technology Centres, and is the only one devoted to the Media sector. BM’s trustees are representatives of the Media industry, the Catalan Government, Barcelona City and four universities. BM has an extremely strong record in European collaborative R&D and Innovation projects, both as partner and coordinator. BM is currently involved in 14 EU funded research projects in information and communication technologies with over 5 million € EC funding. BM was coordinator of an FP6 IP and 2 STREPs, including IP-RACINE which researched and developed digital cinema technologies ‘from scene to screen’. It is now co-ordinating the FP7 ICT IP 2020 3D Media, developing 3D digital cinema and home entertainment. Other directly relevant projects are IP SALERO (‘intelligent content’ objects with context-aware behaviours), SEMEDIA (Search Environments for MEDIA) and FP5 SPEED-FX (very high resolution real-time graphic interaction for digital cinema).

16 years, 10 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev