Linux-audio-dev June 2006

linux-audio-dev@lists.linuxaudio.org

70 participants
59 discussions

[linux-audio-dev] LAC-Konzertreport auf SWR2 15.5. 23h (German only)

by Michael Iber

Hi lists, my little report on the lac-concerts and some other subjects of the lac in Karlsruhe will be broadcasted tomorrow on SWR2 JetztMusik Magazin at 23h. Michael

17 years, 1 month

[linux-audio-dev] Re: [linux-audio-user] Recording ADAT inputs on RME hammerfall 9636/52

by Stephen Cameron

Well, mixed results tonight. I was able to get some sound to go across the ADAT cables from the PC to the AW4416. But not good sound. On the bright side, I think I more or less understand connecting things up with jack, ecasound, and so on. On the bad side, so far it's not working too well. I monitored things with "jackmeter" and this meter registered peaks near 0dB for the stuff I was playing with ecasound, and pretty high levels for the most part. On the AW4416, the levels were registering between -30dB and -48dB. I guess I don't understand how ADAT works. I was under the impression the signal going across the cables was digital -- and so to get a reduction in levels like that, I would expect some digital numbers would have to go from being big numbers to being small numbers, which seems unlikely thing to happen to numbers encoded as pulses going down a cable. So I conclude I don't know how ADAT works, except it's not as I imagined it did. Oh, and besides a drastic loss of signal level, the signal was distorted strangely. Hard to describe. This may be due to xruns... I haven't got things to work without xruns yet, but that shouldn't cause a drop in levels, right? Just kind of choppiness, dropouts, crappy sound, right? Transfering from the AW4416 to the PC did not work at all. on capture_1 and capture_2, I got very low level white noise apparently. Are those the s/pdif ports? On the other channels input was dead silence. I tried both ADAT ports on the RME board, with similar results on each. I tried swapping the two ADAT cables in case one of the cables was bad... this did not seem to make a difference. Maybe the RME just transmits harder than the Yamaha, so it's signal makes it across (just barely, crossing the finish line at -48dB) while the yamaha's signal dies. I did change the RME's frequency to 44.1kHz in qjackctl's setup window. Maybe there are some clues in here: [root@zuul R15]# cat /proc/asound/R15/rme9652 RME Digi9636 (Rev 1.5) (Card #2) Buffers: capture f6a00000 playback f6400000 IRQ: 10 Registers bus: 0xea000000 VM: 0xf88a2000 Control register: 48029 Latency: 1024 samples (2 periods of 4096 bytes) Hardware pointer (frames): 1024 Passthru: no Clock mode: autosync Pref. sync source: ADAT1 ADAT1 Input source: ADAT1 optical IEC958 input: Internal IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 Dolby: off IEC958 sample rate: error flag set ADAT Sample rate: 44100Hz ADAT1: No Lock ADAT2: Sync ADAT3: No Lock Timecode signal: no Punch Status: 1: off 2: off 3: off 4: off 5: off 6: off 7: off 8: off 9: off 10: off 11: off 12: off 13: off 14: off 15: off 16: off 17: off 18: off __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

17 years, 4 months

[linux-audio-dev] realtimeness: pthread_cond_signal vs. pipe write

by Stefan Westerfeld

Hi! I am trying to notify a high priority (nice -20 or nice -19) thread from a realtime thread (from a jack callback to be precise). Of course I want the realtime thread to not block, but I want the high priority thread to react as soon as possible. For the first implementation, I used a condition, and pthread_cond_signal. But as I can't aquire a mutex within the realtime thread, at least not always (at most with trylock()), this results in racy code. Also, it doesn't integrate well with poll(), which the high priority process should be able to use. So basically I want to switch to a pipe write() instead of pthread_cond_signal. Now the question is: am I producing priority inversion, then? I've seen that the kernel code of pipe write has code like mutex_lock(PIPE_MUTEX(*inode)); (in fs/pipe.c, linux-2.6.16.16), so can this mutex cause priority inversion? However, on the other hand, I examined the code of glibc-2.3.6/ntpl/sysdeps/pthread, and found in pthread_cond_signal.c: __pthread_cond_signal (cond) pthread_cond_t *cond; { /* Make sure we are alone. */ lll_mutex_lock (cond->__data.__lock); /* Are there any waiters to be woken? */ if (cond->__data.__total_seq > cond->__data.__wakeup_seq) { /* Yes. Mark one of them as woken. */ ++cond->__data.__wakeup_seq; ++cond->__data.__futex; /* Wake one. */ lll_futex_wake (&cond->__data.__futex, 1); } /* We are done. */ lll_mutex_unlock (cond->__data.__lock); return 0; } So it looks like there is some kind of low level mutex locking involved even when signalling a condition. Can't this also lead to priority inversion? So what is the best way to achieve that some lower priority thread will definitely wakeup from a realtime priority thread, without causing priority inversion, under recent linux kernels? Cu... Stefan -- Stefan Westerfeld, Hamburg/Germany, http://space.twc.de/~stefan

17 years, 9 months

[linux-audio-dev] crossplatform atomics

by Maarten de Boer

Hello, I am looking for a cross-platform implementation of an atomic integer. Under Linux, a build an c++ class "atomic" around asm/atomic.h, (which I can use as if it where an int), but I'd like to have a solution that also works on Windows XP and Mac OS X. Thanks for any suggestions, Maarten

17 years, 9 months

Re: [linux-audio-dev] fst, VST 2.0, kontakt

by Forest Bond

> > In any case, Linux Sampler and Kontakt are > > different sorts of samplers at this point. Linux Sampler is more in the spirit > > of Gigasampler, and Kontakt is more like an emulation of a hardware sampler (but > > vastly more flexible). > > The key words there are "at this point." As noted, the *long-term* > potential of Linux Sampler is probably still much greater than Kontakt-in-wine. Agreed. Maybe time is best spent improving linux sampler. Then again, good VST has benefits, although I do see how it may hurt the long term goals of free software. I'm just not sure if the best way to help free software is to limit it's ability to integrate with proprietary software. > > > > Noted. However, I remain unconvinced that Linux Sampler is intended to be a > > drop-in replacement for something like Kontakt. > > I don't think it's intended to be a drop-in replacement for Kontakt; I > think it's just intended to be cool. I can appreciate that. -Forest

17 years, 9 months

[linux-audio-dev] IR FFT smoothing

by Andrew Gaydenko

Hi! Trying to understand some magic with Ipmulse Response, FFT and so on, I'm playing with scipy, matplotlib (and python as new language for me as a bonus). Concrete IR was got with help of DRC (rec_imp, saying more strictly). Currently I have got this frequency response graph: http://gaydenko.com/mix/needSmooth.png And the question is: what is common way to smooth the result? Some offtopic apps has something like "1/24, ..., 1/3 octave smoothing". What does it mean? Andrew

17 years, 9 months

[linux-audio-dev] What valgrind says

by james＠dis-dot-dat.net

Hi peeps. I've just been running an app through valgrind and I'm getting a few of these: ==11955== Syscall param write(buf) points to uninitialised byte(s) ==11955== at 0x4D51BDB: (within /lib64/libpthread-2.4.so) ==11955== by 0x4B27CDD: (within /usr/lib64/libjack.so.0.0.23) ==11955== by 0x4B29434: jack_deactivate (in /usr/lib64/libjack.so.0.0.23) ==11955== by 0x4B2945F: jack_client_close (in /usr/lib64/libjack.so.0.0.23) ==11955== by 0x40402F: jackpart_close() (jackPart.cc:86) ==11955== by 0x40235F: mainLoop() (sampleplayer.cpp:438) ==11955== by 0x40237B: main (sampleplayer.cpp:444) ==11955== Address 0x7FEFFE934 is on thread 1's stack I can't see anything wrong on my part, so I thought I'd check with people who know more than I before I tear my program apart. I also get them coming from jack_port_register, jack_activate and jack_connect. Any ideas? James

17 years, 9 months

Re: [linux-audio-dev] fst, VST 2.0, kontakt

by Forest Bond

> You are... Linux sampler doesn't support built-in effects because of what > working with audio is like under linux. JACK (jack audio connection kit) > is an application that allows one to manage connection between various > audio software. This includes linux sampler, but also plugins based on the > LADSPA sdk. You are right to say that there are no built-in effect in > Linuxsampler. It's not a lack of functionnality. It's meant to be like > this because you can connect linuxsampler's output to a effect management > rack like jack-rack (which also benefits of being jack compatible) have > any effect you like applied (and there's loads ! see > http://www.ladspa.org/). I can appreciate this modularity as much as the next fellow, but having effects support in the sampler itself is near-critical. Creating sophisticated sounds using a sampler requires it. I need to have multiple layers of samples, each filtered and modulated differently. You can't do this in post-processing. It just isn't possible. Even supposing Linux Sampler could route sounds out to effects and back in to the sampler, that would be far to complicated to be even remotely usable without giving up huge amounts of sampling felixibility. This is not about applying reverb to a drum kit. It's about having 6 samples triggered when I press a key on the keyboard, and each sample being filtered, pitch bended, saturated differently, and modulating all of those parameters using LFO's, envelopes, and midi parameters. You can't do that using "outboard" effects. -Forest

17 years, 9 months

[linux-audio-dev] [ANN] Aqualung 0.9beta5 released

by Tom Szilagyi

Aqualung: Music Player for GNU/Linux http://aqualung.sf.net Release 0.9beta5 It is our greatest pleasure to announce the fifth official beta release of Aqualung. Some features you'd rarely stumble upon in other players (at least not too many of them at once): * Gapless playback (designed for this from the ground up) * High quality decoders (eg. libMAD for mp3), many supported formats * High-quality sample rate conversion support via libsamplerate * LADSPA support * Music Store for organizing your music * And much, much more... We hope you will enjoy this release. The release ChangeLog follows below. 2006-06-30 Tom Szilagyi <tszilagyi at users dot sourceforge dot net> * Aqualung 0.9beta5 http://aqualung.sf.net This is a new milestone release after 17 months of silent development. Large parts of the program have been rewritten, refactored, fixed, etc. A multitude of new features have been added to the software, which now weighs into Open Source with about 30,000 lines of GPL'ed source code, all written by a handful of free-time developers (no, you won't need your whole hand). It won't make too much sense to precisely list every change made to the sources during this period - the list would be prohibitively lengthy. For the curious, the mailing list archive is recommended. The most important, high-level changes are summarized below. * Group CDs in the Playlist via "Album mode". Shuffle between records but play their contents in order! * Statusbars in Playlist and Music Store display statistics and other data. * Multiple Music Stores are supported - useful for separate genres, file formats or for music mounted from different file servers via NFS. * CDDB support! * iFP driver support for integrating with iRiver HW players! * Completely reworked Settings dialog, the new control center! * Embed Playlist into Main window for a more compact look! * Search facility for Music Store and Playlist. * Add support for Musepack (via libmpcdec), Monkey's Audio, Ogg Speex. * Rudimentary album art (cover display) support. * RVA-related work, improved metadata support. * Fixed a boatload of bugs concerning cyrillic filenames, etc. * MP3 improvements (file recognition, clipping, seeking...) * Better fault tolerance in Ogg Vorbis decoder. * Various GUI fixes, new command line options, etc, etc. * Improved build system for skins, icons, etc. * New skins (Ocean, Plain), new Logo (see About box)! ;-) * Better RT behaviour with Jack output. * Compiles and runs on AMD64 (thanks to Mark Knecht for testing)!

17 years, 9 months

[linux-audio-dev] Envelopes

by james＠dis-dot-dat.net

Hi peeps. I've been writing an app that requires volume envelopes. I've implemented the envelope part myself, but I was wondering if there was a library about to do it. I ask because I realised that probably everyone on this list has written something that uses envelopes, and probably written it better and with more features than I have. More generally, is there a library/toolkit of audio bits and bobs about? James

17 years, 9 months

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Linux-audio-dev June 2006