Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7536 discussions

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

17 years, 5 months

[LAD] 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 !

17 years, 5 months

[LAD] 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

17 years, 5 months

[LAD] 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 !

17 years, 5 months

[LAD] 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).

17 years, 5 months

Re: [LAD] [LAU] Ubuntu Studio 8.10, -rt and 2.6.27

by Cory K.

Hartmut Noack wrote: > Cory K. schrieb: > > > * Shipping the -generic kernel with this 8.10 release of Ubuntu > > Studio and let people compile their own -rt kernel. > > This could be done in any Distro so there would not be a real > Ubuntustudio anymore. I don't agree completely since our focus is not *just* audio, though we know that's our biggest user segment. > The major strength of UBuntustudio is its > near-perfect integration of a audiosystem with a friendly desktop-distro. > I can run VMWare and NVIDIA-Drivers easily with the UBuntu rt kernel > would be a major p.i.t.a. do make stuff like that run with a self made > kernel. These could also be the very things you lose. > > * Ship a out-of-sync 2.6.26-rt kernel, hoping for a Stable Update > > Release in Intrepid with .27 later. > > This would be perfectly acceptable for me :-) > > > best regs > HZN > > BTW: what is so extremely important in .27? new hardwaresupport not > achievable with .26? Much better suspend/resume support along with many bug-fixes and device support. -Cory K.

17 years, 5 months

[LAD] Ubuntu Studio 8.10, -rt and 2.6.27

by Cory K.

Hello all. This is going out to a couple of lists so I can get as wide an opinion as I can. I'll correct any inaccuracies as this discussion progresses. Quick intro: I'm Cory K. Lean on Ubuntu Studio. Hi :) So here's the pickle we're in as I understand it. The way the kernels are now managed in Ubuntu has changed radically in this release. Thus causing our kernel guy *much* more work that ever before. We've had to work very hard to get upstream -rt to support the .26 kernel but, now mainline Ubuntu has moved to 2.6.27. Which upstream -rt doesn't *look* to support yet. So because of these, and other issues that don't matter to the question I have we're looking at these options: * Shipping the -generic kernel with this 8.10 release of Ubuntu Studio and let people compile their own -rt kernel. With a latter PPA release of -rt for testing as upstream support happens. * Ship a out-of-sync 2.6.26-rt kernel, hoping for a Stable Update Release in Intrepid with .27 later. Of some combination of those. Thoughts on what to do? What do users want? (please be mature) -Cory K.

17 years, 5 months

[LAD] [ANN] Qtractor 0.2.1 (fainty diva) released!

by Rui Nuno Capela

Howdy! Vacation time is now gone and the rentrée is upon us, at least on this hemisphere side of the globe ;) Silly season, beloved August, summertime it is. And before it's too late, I'm proud to announce that a new alpha has summoned to the challenge: Qtractor 0.2.1 (fainty diva) has been released! Release highlights: - Gradient eye-candy (NEW) - DSSI multi-instance plug-in issue fixed - Clip waveform/piano-roll rendered while recording (NEW) - MIDI pitch-bend, omni-channel filtering and recording fixes - Play-head positioning in the thumb-view (NEW) - And many few more (see change-log below:) For the clueless: Qtractor is an audio/MIDI multi-track sequencer application written in C++ with the Qt4 framework [1]. Target platform is Linux, where the Jack Audio Connection Kit (JACK) for audio [2] and the Advanced Linux Sound Architecture (ALSA) for MIDI [3] are the main infrastructures to evolve as a fairly-featured Linux desktop audio workstation GUI, specially dedicated to the personal home-studio. Author: Rui Nuno Capela <rncbc at rncbc dot org> Weblog: http://wwww.rncbc.org Website: http://qtractor.sourceforge.net Project page: http://sourceforge.net/projects/qtractor License: Qtractor is free, open-source software, distributed under the terms of the GNU General Public License (GPL) [15] version 2 or later. Features: - Multi-track audio and MIDI sequencing and recording. - Developed on pure Qt4 C++ application framework [1] (no Qt3 nor KDE dependencies). - Uses JACK [2] for audio and ALSA [3] sequencer for MIDI as multimedia infrastructures. - Traditional multi-track tape recorder control paradigm. - Audio file formats support: OGG (via libvorbis [6]), MP3 (via libmad [7], playback only), WAV, FLAC, AIFF and many, many more (via linsndfile [4]). - Standard MIDI files support (format 0 and 1). - Non-destructive, non-linear editing. - Unlimited number of tracks per session/project. - Unlimited number of overlapping clips per track. - XML encoded session/project description files (SDI). - Point-and-click, multi-select, drag-and-drop interaction (drag, move, drop, cut, copy, paste, delete, split) - Unlimited undo/redo. - Built-in mixer and monitor controls. - Built-in connection patchbay control and persistence (a-la QjackCtl [13]). - LADSPA [5], DSSI [11] and native VSTi [12] plug-ins support. - Unlimited number of plug-ins per track or bus. - Plug-in presets, programs and chunk/configurations support. - Audio/MIDI clip fade-in/out (linear, quadratic, cubic). - Audio clip time-stretching (WSOLA-like or via librubberband [9]), pitch-shifting (via librubberband [9]) and seemless sample-rate conversion (via libsamplerate [8]). - Audio/MIDI track export (mix-down, merge). - Audio/MIDI metronome bar/beat clicks. - MIDI clip editor (piano roll). - MIDI instrument definitions (a-la Cakewalk(tm) [15]) - JACK transport sync master. - MMC control surface enabled. - Configurable keyboard shortcuts. Requirements: - mandatory libqt4 [1] libjack [2] libasound [3] libsndfile [4] LADSPA [5] - optional libvorbis [6] libmad [7] libsamplerate [8] librubberband [9] liblo [10] DSSI [11] VST-SDK [12] Download: http://downloads.sourceforge.net/qtractor/qtractor-0.2.1.tar.gz Change-log: 0.2.1 2008-08-30 The Faint Diva - Gradient eye-candy now featured for clips, tracks and mixer strips widget backgrounds, disabled on configure time (--disable-gradient). - MIDI pitch-bend/wheel events are now captured/recorded properly; in fact, there was this ancient bug, now squashed, as all MIDI clips weren't being stored at all if there weren't a single note event captured. - MIDI channel translation is finally in effect on monitored tracks, specially the ones set in omni-channel input mode. - MIDI open files dialog gets a few more file filter types now. - Playhead position is now shown, updated and can be also set on the main toolbar session thumb-view (shift-click to set the playhead). - The floating selection, as shown for a clipboard pasting operation, has been corrected regarding time scale (horizontal zoom) changes, while in the main track-view. Also fixed final position snapping precision (spotted in bug #2055846). - Current tempo and snap-per-beat setting now survive session cycling. - DSSI plugins implementing run_multiple_synths (eg. fluidsynth-dssi) are now formally supported according to the DSSI spec (hopefully); note that previously one could only have one DSSI plugin instance loaded in session in any given time, otherwise a sudden crash was in sure demand when either plugin got activated (fixes bug #2048198). - Audio plugin outputs now overrides each other when the number of output ports does not match the number of mix-down audio channels and thus fixing a nasty crash bug in the process. - All custom font sizes are now set relative to default global font. - Changing loop points by dragging on the time rulers is now mapped as undoable/redoable commands as it ought to be ever since. - Drop-span option (View/Options.../Drop multiple audio files into the same track) now takes into effect the current session snap-per-beat setting when concatenating multiple audio clips. - All plugins get their default bank/program selected on insertion. - Make record armed tracks muted for playback, a needed regression to let both audio and MIDI tracks behave consistently regarding input monitor switching through output. - Fixed a pretty old and shameless bug regarding MIDI clip recording, in which cases the queue start time offset was not taken into account, with captured events being recorded with erratic long delays. - Almost complete refactoring of the clumsy audio peak file classes, possibly making it even clumsier but straight neverthless, with the noble and final aim to build it in-flight and draw the waveforms while recording is rolling. - Recording clips get their initial name labels drawn while fade-in/out handles are dropped as found pretty useless while recording. - Escape key now closes connections and mixer widgets as found usual. Disclaimer: Qtractor is (still) in some alpha stage of development, although already functional it surely has a plenty of lurking bugs. It also lacks some features that some might find critical in the application field. All is (still) not lost, as the following are among the ones planned to be worked on in the times to come (not in any particular order): - Automation and MIDI controller mapping. - Session/project templates. - Puch-in/out and loop recording (takes). - Audio clip cross-fading. - Audio effect send/return aux. inserts. - Clip locking, muting, normalize (gain). - LV2 plug-in support. - JACK MIDI support. - Tempo and key signature map. - Paste-repeat command. - MIDI SysEx manager. - MIDI event list editor. - MIDI editor (piano roll) draw mode. - MIDI groove/swing quantize. - OSC interface. - Scripting. - Etc. etc. References: [1] Qt4 (core, gui, xml), C++ class library and tools for crossplatform development and internationalization http://www.trolltech.org/products/qt/ [2] JACK Audio Connection Kit http://jackaudio.org/ [3] ALSA, Advanced Linux Sound Architecture http://www.alsa-project.org/ [4] libsndfile, C library for reading and writing files containing sampled sound http://www.mega-nerd.com/libsndfile/ [5] LADSPA, Linux Audio Developer's Simple Plugin API http://www.ladspa.org/ [6] libvorbis (enc, file), Ogg Vorbis audio compression http://xiph.org/vorbis/ [7] libmad, High-quality MPEG audio decoder http://www.underbit.com/products/mad/ [8] libsamplerate, The secret rabbit code, C library for audio sample rate conversion http://www.mega-nerd.com/SRC/ [9] librubberband, Rubber Band Audio Time Stretcher, an audio time-stretching and pitch-shifting library http://breakfastquay.com/rubberband/ [10] liblo, Lightweight OSC implementation (needed for DSSI GUI support) http://liblo.sourceforge.net/ [11] DSSI, an API for soft synth plugins with custom user interfaces http://dssi.sourceforge.net/ [12] VST SDK, Steinberg's Virtual Studio Technology (see README.VST) http://www.steinberg.net/ [13] QjackCtl - JACK Qt GUI Interface http://qjackctl.sourceforge.net [14] GNU General Public License http://www.gnu.org/copyleft/gpl.html [15] Cakewalk (by Roland), powerful and easy-to-use tools for Windows-based music and sound production http://www.cakewalk.com/ Cheers && Enjoy. -- rncbc aka Rui Nuno Capela rncbc at rncbc dot org http://www.rncbc.org

17 years, 5 months

Re: [LAD] CUDA

by Jens M Andreasen

Kryz, your mail-severs seems to be down and when I checked out if you had a running WWW, I got this message: "Chwilowo nic tu nie ma" [We are not here yet] So, back to the list ... This time corrected as per second mail: On Tue, 2008-08-26 at 11:12 +0100, Krzysztof Foltman wrote: > Jens M Andreasen wrote: > > 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. > > Does it require the proprietary drivers and/or Nvidia kernel module? > Yes, and not only that. The proprietary drivers distributed with say Mandrake, Ubuntu et al won't work either. Uninstall that, change your X setup to vesa (to stop recursive nvidia installer madness) and then get your CUDA driver and compiler from: http://www.nvidia.com/object/cuda_get.html > What kind of things is the gfx card processor potentially capable of > doing? Anything like multipoint interpolation for audio resampling > purposes? Multiple delay lines in parallel? Biquads? Multichannel > recording to VRAM? Multichannel recording by itself would be a waste of perfectly good floating point clock-cycles, but anything that you can map to a wide vector (64 to 196 elements) is up for grabs. A 196 voice multi timbral synthesizer perhaps or 64 channel-strips with basic filters and compressor/noise-gate for remix. The five muladds needed for a single biquad filter times the number of bands you need to equalize on fits the optimal programming model quite well. The linear 2D interpolator is also available and even cached. Perhaps not the worlds most useful toy for audio-resampling, but could find its way into some variation of wave-table synthesis. It can be set up to wrap around at the edges, which I find kind of interesting. Random access to main (device) memory is - generally speaking - a bitch and a no-go if you cannot wrap your head around ways to load and use very wide vectors. There are some 8096 fp registers to load into though, so all is not lost. Communication, permutation and exchange of data between vector elements OTOH is then fairly straight forward and cheap by means of a smallish shared memory on chip. The more you can make your algorithm(s) look like infinitely brain-dead parallel iterations of multiply/add, the better they will make use of the hardware. The way I see it, the overall feel of your strategy should be something like "The Marching Hammers" animation (from Pink Floyd: The Wall.) > Is it possible to confine all the audio stream transfer between gfx > and audio cards to kernel layer and only implement control in user > space? (to potentially reduce xruns, won't help for control latency > but at least it's some improvement) > Yuo you mean something like DMA? Yes I would have thought so but this is apparently not always the case, Especially not on this very card that I have here. :-/ The CUDA program running on the device will have priority over X though. So no blinking lights (nor printfs) before your calculation is done. For real-time work, I reckon this as a GoodFeature (tm)! Potentially this can also hang the system if you happen to implement the infinite loop (so don't do that ...) > Would it be possible to use a high level language like FAUST to > generate CUDA code? (by adding CUDA-specific backend) > The problem would be to give Faust a good understanding of the memory model and how to keep individual vector elements away from collectively falling over each other. But I must admit that I am not too familiar with what Faust is actually doing? Would it be of any help to you with a library of common higher level functionality like FFT and BLAS? ---8<--------------------------------------------- - "CUBLAS is an implementation of BLAS (Basic Linear Algebra Subprograms) on top of the NVIDIA(r) CUDA(tm) (compute unified device architecture) driver. It allows access to the computational resources of NVIDIA GPUs. The library is self-contained at the API level, that is, no direct interaction with the CUDA driver is necessary." ------8<................................ But observe that: ... - "Currently, only a subset of the CUBLAS core functions is implemented." /j > Krzysztof >

17 years, 5 months

[LAD] Wireless Electricity vs audio quality

by Patrick Shirkey

Hi, After the latest Intel announcment that they are indeed working seriously on wireless electricity uses resonance to transmit hte energy sources I am wondering if anyone has ideas on whether that would affect audio quality as resonance is an important part of the ambience of live sound. Cheers. -- Patrick Shirkey Boost Hardware Ltd

17 years, 5 months

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Linux-audio-dev