Linux-audio-user September 2008

linux-audio-user@lists.linuxaudio.org

124 participants
125 discussions

by Grammostola Rosea

Hi, Are there also specialized Bsd or Solaris (unix) distro's for audio like you have 64studio for linux for example? Are there people here on the list who uses BSD or Solaris or ... for audio/ music? Do 'they' have there own list or is this list also for 'them'? Regards,

17 years

convert sd2 to wav

by Atte André Jensen

Hi Is there a tool that'll allow me to convert a file in sd2 format to wav in linux? I tried sox, audacity and mhwaveedit, but this is a brand new install, so maybe I need to install some codeces... I asked google but it didn't give me anything useful. Any input appreciated. -- Atte http://atte.dk http://modlys.dk

17 years

PulseAudio and Alsa

by Stephen Doonan

I recently installed Fedora 9, then Ubuntu Hardy Heron and the UbuntuStudio packages. In both cases, a new soundserver (new to me), PulseAudio, was installed by default and has caused me some confusion and frustration. What is the feeling among Linux audio users about PulseAudio? Is it a good development? Will it eventually replace ALSA, or does it instead simply add an additional level or layer of abstraction and indirection to audio in Linux. Will the simplicity it seems to aim for add yet more complexity to the issues surrounding configuration and troubleshooting of audio in Linux, especially for those who use the specialized audio applications that many Linux-audio-users use? How are users responding to the development of PulseAudio and its inclusion in Linux distribution installations? Remove it afterward and revert to pure ALSA, disabling Gnome and KDE specific soundservers? Learning to live with PulseAudio and how to make it work well with Ardour, Jamin, Rosegarden, Muse, Qtractor, Qjackctl, etc.? :-) -Steve (Steve Doonan, Portales, New Mexico US)

17 years

mix lof with ecasound

by becks

Hi all, i have many .lof[1] files, and i want a final mix (audacity can open .lof file, but i need one batch script) ex: $ cat ab.lof window file "a0.wav" offset 0.000000 file "b0.wav" offset 396.000000 file "a1".wav offset 0.000000 file "b1.wav" offset 396.000000 i can make a mix with something like: $ ecasound -a:1 -i a0.wav -a:2 -i a1.wav -a:all -o output.wav my problem is offset, i think b0 (and b1) start after 396 sec i can use ecasound to combine a0+a1+b0+b1 together? thanks [1]: lof format: http://audacityteam.org/wiki/index.php?title=How_to_import_playlists -- http://msound.org

17 years

Standard Sample Dump with amidi

by Mirko Horstmann

Hi there, I'm currently trying to send samples to a semi-ancient synthesizer (a Yamaha SY99), which has a sample RAM of 512kB built in. The way I do it, the synth seems to receive "something" but doesn't really acknowledge that it has received the whole sample and is acting strange afterwards (won't make a sound until I turn off and on the machine and then there seems to be only garbage in the memory). I understand that the intersection between the sets of SY99 users and Linux users is probably quite small and that I may have to ask on the SY99 list at Yahoo what the synth's behaviour means. But I couldn't find any definite answers as to whether what I do on the Linux side is any good: I save the sample as mono, 16bit, 48000kHz (which the synth is supposed to be able to handle) and in ".sds" format (using Sweep, which uses libsndfile). The beginning few bytes of that file seem to be compliant with a midi sysex dump. Then I use the following command to send it to the synth ("hw:0,0" being the midi port of my M-Audio Audiophile): amidi -p hw:0,0 -s sample.sds Is this supposed to work at all or am I missing something? Thanks, Mirko

17 years

[Hardware] Small linux box / sampler?

by Jano

Hello everyone in this my first message, hope you can give me some advice in this semi-OT question. I have a clavinova keyboard which sounds are growing tiresome on me. I have often used my laptop with ubuntu studio and a regular midi-to-usb cable to use qsynth with soundfonts from my collection grown over the years. This works quite well, but since this laptop is not always there, and tinkering with cables is always a bit of a chore, I'm thinking of some more definitive solution. One possibility is to have a (preferably cheap) silent small linux box always on the keyboard. I guess that after the initial setup it should work as fine as the laptop (although I intend not to have a monitor there, so the aspect of presets should receive a bit more thinking). One candidate could be this one, or something along these lines. I should think it would be powerful enough? I'm not really sure about the CPU demands of qsynth. http://www.tranquilpc-shop.co.uk/acatalog/T7_Ubuntu_PC.html Another possibility is some basic stand-alone sampler. I confess not to know the current hardware that could serve for this, I've always been a PC man. I'd prefer one capable of using soundfonts, or at least some format to which sf can be converted, since I'm quite used to my collection. And of course any other option I haven't think of. Thanks in advance for any comments, Alex.

17 years

Problem compiling SooperLooper

by Joe Bain

Hi All, I'm trying to compile SooperLooper 1.6.10 on Ubuntu (Hardy) and I'm having some problems. I'm getting lots of errors like: "/sooperlooper-1.6.10/src/jack_audio_driver.cpp:329: undefined reference to `jack_port_get_buffer'" Basically lots of 'undefined references' to jack things. I have jack and the libjack-dev package installed so I'm a bit stuck on what could be the problem. Any help is appreciated. Joe

17 years

Ardour2- Ranges, Playlists, and Selecting Them

by The Other

08/31/08 Hello All, I've just spent over 4 hours googling for Ardour2 tutorials and reading through the Ardour2 manual trying to figure out how to cut off the beginning and ending silence on a recording. I still haven't figured out how to set the beginning and ending points of a range so I could delete the range. Playlists are a complete mystery to me. My solution so far is to export the recording to WAV file, normalize it with ecasound, thanks to Julien Claasen for his email on how to do that, because I wasn't able to find a way in Ardour2 to normalize less than the maximum amount, then import the file to Audacity and chop off the beginning and ending silence. This process took less than 2 minutes. No doubt I missed finding a "Cookbook" tutorial for Ardour2 that says 1. Here's the task you want to do. 2. Here's how you do it. Is such a tutorial available? I tried going through Ben Powers Ardour2 Tutorial at 10.1 Snapping, but I was not having any luck getting red and blue bars to appear consistently, and any time I tried to delete, the entire recording was deleted. Thank goodness you can Exit Ardour2 without saving any changes. I was doing that repeatedly. And as stated above, I never got anywhere trying to set a region, even when going through the menus to set region beginning and set region ending. I was still deleting the entire recording. Other suggestions on where to look? Thanks All, Stephen.

17 years

audio player with playbakc with predefined times

by Vedran Vucic

Hello, I Need audio player with possibility to play audio files at predefined times. thanks, veki

17 years

64bit vs. 32bit

by Crypto

Hi, I am considering to migrate my system from 32bit to 64bit. What are Your experiences concerning 64bit for audio applications? Any issues not yet resolved? Anything that does not work at all? What are the REAL advantages of 64bit vs. 32bit? Thanks for any hint, Crypto.

17 years

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Linux-audio-user September 2008