Linux-audio-dev July 2006

linux-audio-dev@lists.linuxaudio.org

76 participants
53 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] Re: Akai's MPC4000 Sampler/Workstation Open Source Project

by Juhana Sadeharju

>From: Renich Bon ?iri? <renich(a)woralelandia.com> > And what you wanted to say? The mail had no content. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

17 years, 9 months

[linux-audio-dev] C++ wrapper for libsndfile

by Erik de Castro Lopo

Hi all, Thanks to suggestions from people here I now have a relatively complete C++ wrapper for libsndfile: http://www.mega-nerd.com/tmp/sndfile.hh There is also a pre-release of libsndfile which includes a test for this wrapper: http://www.mega-nerd.com/tmp/libsndfile-1.0.17pre7.tar.gz C++ users, please comment. Cheers, Erik -- +-----------------------------------------------------------+ Erik de Castro Lopo +-----------------------------------------------------------+ "Even among Europe's Muslim minorities, roughly one-in-seven in France, Spain, and Great Britain feel that suicide bombings against civilian targets can at least sometimes be justified to defend Islam against its enemies." -- http://pewglobal.org/reports/display.php?ReportID=253

17 years, 9 months

[linux-audio-dev] [ANN] das_watchdog V0.2.4 and jack_capture V0.3.7

by kjetil＠ccrma.stanford.edu

New source location: http://www.notam02.no/~kjetism/src/ (Sorry, I have temporarily lost access to both my previously used upload directories) das_watchdog ************************************************************************* Whenever a program locks up the machine, das_watchdog will temporarily sets all realtime process to non-realtime for 8 seconds. You will get an xmessage window up on the screen whenever that happens. Changes 0.2.3->0.2.4 -------------------- *Test if the xmessage program found during the make process is a valid executable. If not, search the $PATH instead. This should fix it for Gentoo when the pro-audio overlay is updated to at least this version. *Various modifications for the High Res Timer, which should be used instead of setting the timer interrupt process to SCHED_FIFO/99. jack_capture ************************************************************************* jack_capture is a small program to capture whatever sound is going out to your speakers into a file without having to patch jack connections, fiddle around with fileformats, or set options on the argument line. This is the program I always wanted to have for jack, but no one made. So here it is. Changes 0.3.1 -> 0.3.7: ----------------------- *Fixed potentional buffer underrun error. *Fixed potentional ringbuffer size allocation miscalculation. *Better way to set leading zeros in filename. Thanks to Melanie. *Better underrun handling. Thanks to Dmitry Baikov. *Added support for jack buffer size change. *Removed some unnecessary code and comments *Beautified code a bit. *Fixed a bug in the reconnection code. *Beautified code a lot. *Changed bufsize argument to accept seconds instead of frames. Default buffer size is 60 seconds. *Improved documentation and help option. *Beautified source a bit. *Fixed bug in ringbuffer size allocation. *Fixed so that more than one instance of jack_capture can run at once.

17 years, 9 months

[linux-audio-dev] light C++ set for WAV

by Andrew Gaydenko

I mean some minimal C++ class set like: WavFile, WavHeader, WavFrame with few apparent methods (open/close, read/write frame(s)). Suggestions?

17 years, 9 months

[linux-audio-dev] Popular LADSPA plug-ins to depend on?

by Jono Bacon

Hi all, I am working on LADSPA support in Jokosher and we want Jokosher to depend on particular plug-ins in different parts of the application. Specifically, I would like to see a powerful compressor and equalizer as part of the application. So, my question to you all is which compressor and equalizer do we depend on? Importantly, the chosen plug-ins need to exhibit the following qualities: * packages for all major distributions (Ubuntu, Debian, Red Hat, Fedora, Gentoo, SuSE etc.) * well maintained * very high quality audio quality Cheers, Jono

17 years, 9 months

[linux-audio-dev] Basic MIDI question

by Lee Revell

I've never been a MIDI expert but I'm now having to learn. I have a question about this excerpt of a MIDI file viewed with hexedit. 00001BB0 22 80 3D 35 31 80 3A 39 0E 80 37 31 03 80 31 1F ".=51.:9..71..1. 00001BC0 81 0C 90 30 5B 00 90 3C 79 81 70 90 39 73 00 90 ...0[..<y.p.9s.. 00001BD0 36 69 4B 80 36 43 0A 80 3C 26 01 80 30 44 0A 80 6iK.6C..<&..0D.. 00001BE0 39 42 82 08 90 37 63 00 90 43 7B 81 70 90 3E 5E 9B...7c..C{.p.>^ 00001BF0 00 90 3A 66 08 80 37 30 02 80 43 32 31 80 3E 11 ..:f..70..C21.> Take the sequence "80 3D 35 31 80 3A 39 0E 80 37 31 03 80 31 1F" in the first line for example. I know that 0x80 is note-off, and 0x3D are note number and 0x35 the velocity of the note-off. But what the heck is the next byte, 0x31? The MIDI standard says note-off is one status byte followed by 2 data bytes! Lee

17 years, 9 months

[linux-audio-dev] Open MPC OS Initiative

by Renich Bon Ćirić

If anyone would like to join the OpenMPC OS dev team, welcome. http://www.woralelandia.com/openmpc/blog/

17 years, 9 months

[linux-audio-dev] Re: Language fanboys [was Re: light C++ set for WAV]

by lazzaro

On Jul 25, 2006, at 9:33 AM, Dave Robillard <drobilla(a)connect.carleton.ca> wrote: > But you don't "just get plug and play" with MIDI. It's all about > learning with MIDI. "Common things should be easy, and unusual things should be possible". The common things in MIDI are plug-and-play. Only the "unusual things" are "all about learning". NoteOn and NoteOff, sustain pedal, volume control, stereo pan, pitch-bend, mod-wheel ... these are all plug-and-play, and have been since the earliest days of MIDI. Manufacturers who make controllers know to send out these commands in a stylized way, and sound designers who write patches for synths (soft and hard) know to make their synths respond in an appropriate way to these controllers. And for a lot musicians, this is enough for them to do what they want to do. This is the MIDI world Garageband lives in, for example, and the biggest problem Apple has with Garageband is that it is an entry-level program that makes most of its users so happy that they aren't interested in upgrading to semi-pro software. --- John Lazzaro http://www.cs.berkeley.edu/~lazzaro lazzaro [at] cs [dot] berkeley [dot] edu ---

17 years, 9 months

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Linux-audio-dev July 2006