Linux-audio-user June 2004

linux-audio-user@lists.linuxaudio.org

140 participants
148 discussions

[linux-audio-user] Copying one channel from a Stereo File with sox

by Martin McCormick

I am confused. The man page for sox lists what appears to be an effect called pick in which you provide it with a channel number from 1 to 4 or a -l or -r flag to select either left or right audio to feed to the output file. After getting complaints from sox, I listed the -help output and got: sox: Known effects: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass map mask pan phaser pitch polyphase rate resample reverb reverse silence speed stat stretch swap synth trim vibro vol no pick. The manual contains the following passage: phaser gain-in gain-out delay decay speed < -s | -t > pick [ -1 | -2 | -3 | -4 | -l | -r ] pitch shift [ width interpole fade ] polyphase [ -w < nut / ham > ] Is this something peculiar to the Debian distribution for sox or what? Am I just misunderstanding the instructions and trying to use pick in the wrong way? Thank you. Martin McCormick WB5AGZ Stillwater, OK OSU Information Technology Division Network Operations Group

20 years, 9 months

[linux-audio-user] List Archive and replying

by Marv Smoak

Question about the archive and replying to a message. I'm running rh8, Planetccrma vintage March, 03. KDE is my friend; KMail, my mail program. I get the list emails and have no problems replying to a message if it's in my lau mailbox. When I'm viewing a message in the archive, and click on Reply, the message to which I'm replying is not quoted in the compose-message screen. I'm sure I can solve this by cut/paste manually. Is there a way to setup the mail program to generate the quote of the the replied-to message automatically as when the original message is in my mailbox? Or is this a limitation of the archive setup? Yes; I'm lazy, Marv

20 years, 9 months

[linux-audio-user] Rosegarden not accepting any more input-- bug?

by Mark Wilson

I was editing an .rg file I had originally started in old Rosegarden 2.1, which a) for some reason wasn't recognized as an rg file by rosegarden-4, and b) would nevertheless allow me to import the MIDI that had been created from the original, BUT I got to a certain point in one of the tracks, and it would not let me insert any more notes or rests. The piece is in 6/8, but there I was stuck with a 5/8 measure. I've not fully investigated possible work-arounds, as it was past my bedtime, but I'm thinking maybe I can create the rest of the track as a different file, and splice it in at a certain point. Will RG let me do this? Thanks, Mark ===== -- Seek professional help! Ask a librarian. __________________________________ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail

20 years, 9 months

[linux-audio-user] [ANN] swh-plugins 0.4.4

by Steve Harris

http://plugin.org.uk/releases/0.4.4/ * Fixes to build on recent gcc * New limiter for beter performance in JAMin * Denomal fixes from Tim Blechmann (general and specifially the flanger) * Many other things I've forgotten :( This release is largely untested, I released it because there are a number of important outstanding fixes. It may break some plugins, so beware. - Steve

20 years, 9 months

[linux-audio-user] Re: Sample Rates / Sample Sizes

by Juhana Sadeharju

>From: Jan Depner <eviltwin69(a)cableone.net> > >I also hear a difference though when I record 24/44.1 and 24/96. How about 24/48 vs. 24/96? Some cards are optimized to 48 kHz and if one goes to use 44.1 kHz the performance may not be optimal. It even could be that in some cards the performance is optimized for 96 kHz instead of 48 kHz. One could start the tests by comparing if an audio bandlimited to 15 kHz comes through the system the same way independent of what sampling rate was used. What would be the limit frequency where differences starts? Is that frequency hearable? A/B tests on the listener. If yes, it would naturally explane why one hears the difference. If all these aspects are already tested, I would like to hear where. Perhaps www.aes.org would be a good place to start searching. Juhana

20 years, 9 months

[linux-audio-user] Re: Sample Rates / Sample Sizes

by Greg Murphy

> On Thu, Jun 24, 2004 at 07:51:04 -0700, Mark Knecht wrote: > >>I'm sure you're right Rick. I was looking at some technical dither stuff >>yesterday. The noise floor different dithering plugins insert is not >>flat, or the same, over frequency, so that could certainly effect >>different types of music differently. Yes, POW-R algorithms seeem to be pretty popular and many engineers will select a particular POW-R type to suit the source material. For me (and my crap ears) the difference in dither types is hard to hear and it was only when I had the chance to use decent monitoring systems (Dunlavy /B&W/ Levinson) that I (think) I heard a difference. > but it does > raise the question of how to best import a 24-bit track into a 16-bit > session. I've have to do this once or twice. Possibly I should have > dithered the track. I'd guess the normal conversions just strip the > bottom 8-bits? Yep, dither beats truncation. I'm not a Pro Tools user so I this information is secondhand but I think that if there is a chain of plugins, data is maintained as float and then converted to the final form (16, 24 or whatever) at the end of the chain. I think that the same goes for most of the Windows DAWs. Somebody correct me if I'm wrong though. So, there should be no need for dither apart from the final word length reduction from 24 to 16 bit. Also, remember that dither is cumulative and so the noise floor can rise. Now, question about LADSPA and DAWs. Is it standard practice to dither from float (or fixed) to integer post-processing? Also, do any (or all) of the Linux DAWs act as described above i.e. maintaining a float datatype if there is >1 plugin in a chain? I suppose that this would require a standard "internal" format of some sort. Greg I'm also new to Linux audio (although not to Linux) so perhaps ___________________________________________________________ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself http://uk.messenger.yahoo.com

20 years, 9 months

[linux-audio-user] about 2.6.x woes [nptl?]

by Florian Schmidt

Hi, i have tried the 2.6.7 kernel now for about two weeks and i always had massive realtime probs when running jackd. xruns would occur although the jack threads were running SCHED_FIFO. all kind of weird "subgraph timed out" messages etc. even running as root didn't better the situation one bit.. i had very good rt performance in 2.4.x with ll+preempt+caps patches. i could run 2*512 for hours w/o xruns [even compile a kernel while doing ardour stuff]. now in 2.6.x i couldn't even get 4*512 or 8*512 to work reliable at all [512 is the max period size of my soundcard]. i experienced the abovementioned symptoms.. well, i stumbled across this thread on the jackit-devel mailing list about trying to set LD_ASSUME_KERNEL=2.4.22 and run all jack stuff [server and clients with this env var set] and it worked wonders.. 2*512 works very nice again.. i suspect this is nptl related (see the jackit-devel thread for details: http://sourceforge.net/mailarchive/message.php?msg_id=8629219).. i think the LD_ASSUME_KERNEL switch somehow makes ld.so link to another libc or something w/o nptl? anyone knows details? so i just wanted to urge all the users of 2.6.x and who experience strange weird non rt operation with jack to try it. maybe it helps... i wrote myself a bash script called "realrealtime" and put that in my path: LD_ASSUME_KERNEL=2.4.22 $@ and use that to start al my jack stuff :) Any ideas about why jackd behaves so weird in the first place appreciated.. Florian Schmidt P.S.: i posted chrt output for jackd w/ and w/o realrealtime at: http://www.affenbande.org/~tapas/realrealtimejack -- Palimm Palimm!

20 years, 9 months

Re: Re: [linux-audio-user] [ANN] TAP-plugins 0.6.0 +more

by ico＠fuse.net

Has anyone tested this latest version against the Cinelerra? The last version caused Cinelerra to crash at start-up while scanning for LADSPA plug-ins. Ico

20 years, 9 months

[linux-audio-user] [ANN] TAP-plugins 0.6.0 +more

by Tom Szilagyi

TAP-plugins 0.6.0 released. TAP Reverb Editor initially released. Homepage: http://tap-plugins.sf.net OK, here we go: * New plugin: TAP Fractal Doubler. Suitable for doubling tracks containing vocals, acoustic/electric guitars, bass and just about any other instrument. The effect is created by applying small changes to the pitch and timing of the incoming signal. The changes are created by a one-dimensional random fractal line producing pink noise. Special thanks to Jan Depner for suggesting this plugin and pointing me to useful information about fractals. * New plugin: TAP Reflector. This plugin creates a psychedelic reverse audio effect. It is especially worth trying this plugin on sustained guitar and vocal tracks. Percussive sounds also create a very characteristic atmosphere when played in a backward-ish style. * New plugin: TAP Pink/Fractal Noise. This plugin came to life as a secondary product of the development of TAP Fractal Doubler. It generates pink noise by means of a one-dimensional random fractal line generated by the midpoint displacement method. Yes, it's Reverb time again! * Vastly enhanced the internal workings of TAP Reverberator. As a result, the sonic quality of reverberation got much better. (Well, at least, now it *IS* real reverberation.) If you tried it once, and found it sounded like hell, you definitely need to check it out now! * Long-long-standing denormal float (or whatever) problems causing occasional runaway CPU-usage led to the complete re-implementation of the internal DSP algorithm of the reverb using fixed-point arithmetics. This inherently fixes denormal problems. However, the option to use the previous floating-point DSP code remains as a #define which you can set before compiling. The default is to use fixed-point math. Very special thanks to Jan Depner for spending his time with repeatedly testing the reverb and sharing his insights. * A new application named TAP Reverb Editor has been written. It is a standalone JACK app, with a GTK+-2 user interface. It works and sounds the same as the LADSPA version (TAP Reverberator), but has extended features that support direct editing of Reverb Types, with immediate visual and sonic feedback. You can design new Reverb Types easily, and 'backporting' these into the LADSPA version is also a breeze. This program is available as a separate package called TAP-reverbed. * Introducing some new Reverb Types: I made these for some acoustic guitar tracks: * Ambience * Ambience (Thick) * Ambience (Thick) - HD ...and these for fun: * Cathedral * Cathedral - HD Other changes: * TAP Dynamics plugins (both Mono and Stereo) were also affected by a runaway CPU-usage issue. It was fixed in the same way as with the reverb: by converting the internals to use fixed point math by default. (The #define option to use floating point math still remains.) * Applied patch from Luke Yelavich to clean up the Makefile a bit. * Complete website/docs redesign. The documentation for TAP-plugins and the user manual for TAP Reverb Editor is now available for download as a separate package called TAP-plugins-doc. Hope you enjoy this release. Please report any problems. Tom

20 years, 9 months

[linux-audio-user] RE: New TAP plugins

by David Baron

Downloaded and actually succeded in compiling them! (What--no configure script to block the compilation, just a valid makefile!). Not so much luck with the reverb-edit. Does not like my pkg-config (it IS recent enough!) and cannot fine a jack.pc file (this may be a non-debian something) My jack IS recent enough. I suppose I could wade through the monster and take these tests out and compile the thing. Then again, jackmix will not compile and neither will gmorgan (this needs alsa include files but cannot find them--I do have them). I guess if I were make configure scripts that reflected my computer, few others would succeed in using them. Maybe the solution is to supply valid makefiles and instructions to set some env parameters before making.

20 years, 9 months

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Linux-audio-user June 2004