Linux-audio-user July 2006

linux-audio-user@lists.linuxaudio.org

131 participants
122 discussions

[linux-audio-user] zombified jackd tascam us-122

by Devin Venable

Audio has never worked well with the us-122 on my box. When it does work, the sound is clippy/choppy. This happens even before jack makes any complaints. I've tried every conceivable jack setting. Here's my latest configuration, along with a log that ends zobmified. Should I just give up and sell the damn us-122? (my PCI audio card works fine...but is noisy; which is why I invested in the us-122) 22:08:26.920 /usr/bin/jackd -R -p256 -t10000 -dalsa -r44100 -p1024 -n2 -D -Chw:1 -Phw:1,0 22:08:26.924 JACK was started with PID=5642 (0x160a). jackd 0.100.0 Copyright 2001-2005 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. loading driver .. apparent rate = 44100 creating alsa driver ... hw:1,0|hw:1|1024|2|44100|0|0|nomon|swmeter|-| 32bit control device hw:1 configuring for 44100Hz, period = 1024 frames, buffer = 2 periods Note: audio device hw:1 doesn't support a 32bit sample format so JACK will try a 24bit format instead nperiods = 2 for capture Note: audio device hw:1,0 doesn't support a 32bit sample format so JACK will try a 24bit format instead nperiods = 2 for playback 22:08:28.985 Server configuration saved to "/home/dextron/.jackdrc". 22:08:28.986 Statistics reset. 22:08:28.988 Client activated. 22:08:28.988 Audio connection change. 22:08:28.989 Audio connection graph change. 22:08:39.970 Audio connection graph change. 22:08:39.996 Audio connection change. 22:08:39.998 Audio connection graph change. 22:08:40.184 MIDI connection graph change. 22:08:40.201 MIDI connection change. 22:10:31.593 Audio connection graph change. 22:10:31.632 MIDI connection graph change. 22:10:31.750 Audio connection change. 22:10:31.752 MIDI connection change. jackd watchdog: timeout - killing jackd zombified - calling shutdown handler 22:10:37.271 Shutdown notification. 22:10:37.273 Client deactivated. 22:10:37.275 JACK was stopped successfully. cannot send request type 7 to server cannot read result for request type 7 from server (Broken pipe) cannot send request type 7 to server cannot read result for request type 7 from server (Broken pipe) Other interesting factors: I'm running 64 bit Ubuntu Dapper, dual core processor, NVIDIA nForce 4 chipset based K8N Neo4 motherboard

18 years, 6 months

[linux-audio-user] [ANN] jack_capture V0.3.1, das_watchdog V0.2.2 and Mammut V0.22

by Kjetil S. Matheussen

http://ccrma.stanford.edu/~kjetil/src/ 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 0.2.4 -> 0.3.1: ----------------------- *Reduced CPU usage a lot because of better disk handling. (25% -> 1%) *Make sure the rest of the recorded file is not garbage in case of an overrun. *Added the port argument, which can be specified many times and accepts both input and output port names (including regexp expressions). This makes jack_capture to completely replace jackrec. *Rewrote buffer handling. Silence is now inserted when underruns occure. Previously, the file became shorter than the recording in case of underrun. It can still happen though, but much more seldom, and a warning about that will be printed to the terminal. *Last rests of jackrec code has been rewritten. Well, all the code with substance, at least. *Nicified code a lot. *More efficient way of handling overruns. *Fixed really stupid compilation error. Thanks to Dragan Noveski for spotting it. 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 0.2.2->0.2.3 -------------------- *Fixed commandline arguments for increasetime, checktime and waittime. *Nicified source a bit Mammut ************************************************************************* Mammut will FFT your sound in one single gigantic analysis (no windows). These spectral data, where the development in time is incorporated in mysterious ways, may then be transformed by different algorithms prior to resynthesis. An interesting aspect of Mammut is its completely non-intuitive sound transformation approach. Changes 0.21->0.22 ------------------ *Added patch and instructions from Owen Green on how to make mammut compile on OSX. Thanks! (Sorry, I forgot to release this version for almost a year...)

18 years, 6 months

[linux-audio-user] 64bit vs 32bit (was: Re: AMD64 question: update)

by Roberto Gordo Saez

On 7/11/06, Arnold Krille <arnold.krille(a)gmail.com> wrote: > Well, I _know_ that 64bit is faster: Here at work I have dual-Xeon's > with 2.8GHz, my laptop is a Turion64 with 1.8GHz. The same computation > (evaluating scientific data from time-of-flight/lifetime experiments) > on the same data-files is significant slower on the work-pc's (almost > factor 2). As my threads are one thread for computation and others for > loading and saving the data, it is really a comparison of two single > processors, where the faster one does even the IO-work itself... Personally, I would like to suggest everyone to actually make some serious tests on 64bit hardware with some intensive sound applications (or other programs that are main tasks for the computer). Latter, you can decide which one you want for everyday use. Surprisingly for myself, I've found that 64bit are NOT always much faster, and sometimes are actually slower. It is true that most applications may improve performance when running on 64bit. But exceptions does exist, mainly because the smaller data chunks on code with 4 byte pointers (32bit), which improves cache usage, and also because there are still lots of applications heavily optimized for 32bit x86 code, whose performance is degraded when running on 64bit. I've made tests mainly on big server applications, I have not made tests on audio software, so I suggest everyone to dedicate some time testing your own apps. It is simple: make two partitions, one of them with a 32bit OS, and the other optimized for 64 bit. Do not use a chroot nor mixed environment (32bit libs on 64bit system), because the results can be slightly different. There is one case in that 64bit is always recommended: if you want to use big resource hungry applications that handle more than 3GB of RAM per process. Linux does support more RAM even on 32bit machines (using PAE extension), but the ugly limit on memory per process on 32bit mode is still there. On 64bit mode processes can access up to 1TB. And yes, they actually exists. I've seen some applications easy allocating more than 50GB of RAM per process! But not on my main desktop computer, it is still a Pentium II :-) I'm interested on this, so if anyone has made some real performance tests with commonly used audio or other multimedia apps on 32bit and 64bit, it would be great to know results.

18 years, 6 months

[linux-audio-user] S/PDIF capture with m-audio 2496

by Rob Jackson

Could anyone please give me a step-by-step guide to setting up my m-audio 2496 to capture from the S/PDIF? I am trying to record into either Audacity or Ardour, but cannot 'find' any sound. I can route data using Envy24Ctrl so that the digital mix bars on the L/H side are active, but cannot capture sound using applications. I'm guessing therefore that alsa is set up incorrectly. Using Audacity, under Edit->Preferences, only /dev/dsp is visible (I cannot select any other interface). Recording just produces a flat line. In Ardour (via jack), I cannot get anything to appear on the v/u bars despite selecting alsa_capture1 & 2. The interface works fine in winXP, so it's something I'm doing wrong. So, can anyone help me to get the sound from my interface to audio applications? Thanks, Rob Jackson

18 years, 6 months

[linux-audio-user] [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)!

18 years, 6 months

[linux-audio-user] LS sources?

by Petter Sundlöf

Anyone know where I can acquire the most recent LS sources? The website is down, and the SF CVS match I found contained only "linuxsampler" (no libgig, liblscp, qsampler) and the file structure was odd.

18 years, 6 months

[linux-audio-user] problems building meterbridge

by Dragan Noveski

hi to the list, i removed the debian meterbridge package from my system, cause i will get rid of the appis depending on jack.deb, making me unable to properli install svn-jack. so last night i compiled h2, zynadd, specimen, jack-rack, everything good, but runing make on meterbridge i get following error: nowhiskey@murija2:~/software/nove/meterbridge/meterbridge-0.9.2$ make Making all in src make[1]: Entering directory `/home/nowhiskey/software/nove/meterbridge/meterbridge-0.9.2/src' make all-am make[2]: Entering directory `/home/nowhiskey/software/nove/meterbridge/meterbridge-0.9.2/src' source='vu_meters.c' object='vu_meters.o' libtool=no \ depfile='.deps/vu_meters.Po' tmpdepfile='.deps/vu_meters.TPo' \ depmode=gcc3 /bin/sh ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I. -I. -Wall -g -I/usr/include/SDL -D_REENTRANT -I/usr/local/include -DPKG_DATA_DIR=\"/usr/local/share/meterbridge\" -c `test -f 'vu_meters.c' || echo './'`vu_meters.c vu_meters.c:11: error: static declaration of 'buf_rect' follows non-static declaration main.h:11: error: previous declaration of 'buf_rect' was here make[2]: *** [vu_meters.o] Fehler 1 make[2]: Leaving directory `/home/nowhiskey/software/nove/meterbridge/meterbridge-0.9.2/src' make[1]: *** [all] Fehler 2 make[1]: Leaving directory `/home/nowhiskey/software/nove/meterbridge/meterbridge-0.9.2/src' make: *** [all-recursive] Fehler 1 for me as newbee, it does not look like missing dependency, but like a code error?! does anybody out there has an idea how to solve this problem. cheers, doc

18 years, 6 months

[linux-audio-user] Nomad - Nord Modular Editor

by Marije Baalman

Hi, I just came across this, and thought some people here might be interested... (I remember the topic coming up some time...) sincerely, Marije ----------------------------------- Hi, our first release of Nomad - Nord Modular Editor is now available. You can download the release in the Files section: Package: nomad Release 0.2.1-pre Please note that this is only a pre release and has several bugs and limitations. However you can do the most important operations: - import patch file(s) - export patch to a file - send a patch to the synthesizer - receive a patch from the synthesizer - edit patch which exists in the synthesizer See also the file 'README.txt' in the release. Requirements: ------------- - Java Runtime Environment (JRE) 5.0 - platform: any Starting Nomad: --------------- You start Nomad with the command: java -jar nomad.jar Micro Modular users: -------------------- Since none of us developers has a Micro Modular available for testing we don't support this model at the moment. However you can try if Nomad works with this model and supply us with feedback so that we can make necessary changes to achive support. Note: Before you use Nomad with the Micro Modular you have to ... - open nomad/conf/application.xml in a text editor - change the line <entry>4</entry> to <entry>1</entry> - save the file Feedback / Reporting Problems: ------------------------------ You can supply us with feedback or report problems at the nmedit-devel mailing list: http://sourceforge.net/mailarchive/forum.php?forum_id=18303 If you want to report exceptions please add the log file 'nomad.log' to your report. Your nmedit team

18 years, 6 months

[linux-audio-user] rt-lsm

by Christoph Eckert

Hi all, I need your help. I always used the rt-lsm module with my rather dated but working kernel. For some reason (system resetup) I need to compile it anew. I never used the kernel patch but I built it as a module externally and copied it over to my modules directory. I tried to download it from http://prdownloads.sourceforge.net/realtime-lsm/realtime-lsm-0.8.6.tar.gz?d… but when trying to unpack it I get some errors and after that I only get a text file which is a patch, not the sources to build it externally. My question: * Can anyone provide the proper sources from his hard drive *or* * Can anyone provide an info how to extract the archive mentioned above properly? Any help is much appreciated, because on saturday I'll present linux audio on an Linux exhibition in Pforzheim, Germany. Thanks & best regards, ce

18 years, 6 months

[linux-audio-user] AMD Turion 64 X2 ?

by oz

Hi @ all, I have sympathy for a new notebook with AMD Turion 64 X2 dual-core cpu. It would open some new possibilities, like virtualization (pacifica, xen etc.), pure AMD64 Debian arch, other 32-bit OS' and - and that is the question - good realtime-audio performance? Do the realtime patches of Ingo Molnar work with SMP-Kernels for example? And the LSM-modules? Or is the realtime-preemption in Linux just optimized for single cpu systems? My main interest would be to use the laptop sometimes as "Linux-sound-module" triggered by an external midi-keyboard. Profit virtual-instruments (virtual-synths, VSTis) much from having two CPUs and therefore enhanced multithreading? Or is it contra-productive eventually? If using a SMP-Kernel would produce hassles - would it be a safe workaround to compile a kernel *without* SMP support and then driving one Turion core would be sufficient? Please forgive these many questions, but I just won't trust the marketing blindly for the benefits of dual-cores without asking some experienced users. Thanks for your time Oliver P.S. What about the laptop-fan? Can I expect, that it starts and stops frequently when the Turion computes "sound"? (that would be annoying)

18 years, 6 months

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Linux-audio-user July 2006