Linux-audio-user February 2007

linux-audio-user@lists.linuxaudio.org

163 participants
149 discussions

[linux-audio-user] mhWaveEdit: Over/underrun in JACK driver

by Atte André Jensen

Hi I normally record my tracks in mhwaveedit, specimen and zyn is routed through ardour, and ardours master is connected to the input of mhwaveedit. However for no apparent reason, after about a minute of recording I get a bunch of these "mhWaveEdit: Over/underrun in JACK driver" and mhwaveedits peak goes wild and from there on the recording is garbage. I tried with two different projects one has hardly changed since last time (where it worked as usual), same result. I didn't change anything, didn't do a dist upgrade. Any idea what could have happened? BTW: The system is 1.6G mobile laptop (ibm) running debian/unstable and 2.6.18-rt7. The projects typically have 10-15 audio tracks in ardour + 5-10 tracks in zyn. The cpu usage when running the smaller project is about 20%. -- peace, love & harmony Atte http://www.atte.dk | quintet: http://www.anagrammer.dk | compositions: http://www.atte.dk/compositions

17 years, 2 months

[linux-audio-user] Setting hardware sample rate of VIA 8237

by Joel Roth

Hi LAU'ers, How do I set the hardware sample rate(s) of my VIA 8237 AC97 sound card? Seems like a FAQ, yet I'm having trouble finding a reference. Alsamixer does not have an item for this. .asoundrc seems to support sample rate _conversion_ but not setting the hardware rate. I think I find some reference to a some pcm_set_sample_rate function in the Alsa libraries, but would rather not write my own utility for this. Below follows some data from /proc/asound/V8237/codec97#0/ac97#0-0 Thanks for any suggestions. --Joel 0-0/0: VIA Technologies VIA1617A [...] Current setup [...] ADC/DAC loopback : off Double rate slots: 10/11 Extended ID : codec=0 rev=2 LDAC SDAC CDAC DSA=0 SPDIF DRA VRA Extended status : LDAC SDAC CDAC SPDIF=3/4 SPDIF VRA PCM front DAC : 48000Hz PCM Surr DAC : 48000Hz PCM LFE DAC : 48000Hz PCM ADC : 22050Hz SPDIF Control : Consumer PCM Category=0x2 Generation=1 Rate=48kHz -- Joel Roth

17 years, 2 months

[linux-audio-user] jack_connect equivalent for MIDI ?

by Jean-Baptiste Mestelan

Hello, people :-) I am writing some scripts for automating JACK connections. I can use jack_connect for audio ports ; but is there a way to connect MIDI ports in such a way ? I cannot find any information about this, and jack_lsp apparently only lists audio ports ... Thanks for attention.

17 years, 2 months

[linux-audio-user] freesound integration in Aldrin

by Leonard Ritter

In my quest to equip Aldrin with some serious collaboration features, I spent the past week adding support for The Freesound Project. http://www.leonard-ritter.com/freesound_integration (includes screenshots and sourcecode) -- Leonard Ritter -- Freelance Art & Logic -- http://www.leonard-ritter.com

17 years, 2 months

[linux-audio-user] Anyone have any opinion on CMS?

by oktyabr

Hiya, I didn't want to post here until I actually had something to show off a bit but basically the story goes like this... I had some webspace I wasn't using so I registered www.linux-studio.org (nothing meaningful there at the moment) with the idea of setting up a... (save the rotten vegetables for a minute!)... ... forum. Specifically I want to put up a sort of blog on the front page with easy access to forums from there. Since combining a blog software with a forum can be tricky to get up as well as admin I thought I would look at an integrated solution in the form of one or another CMS (Content Management System) that can provide both in a small, efficient package. So if anyone out there has experience with CMS and can either recommend or recommend against any particular package that would be well suited for such a task I would be very happy to hear your opinions! There are a ton of them out there and it is quite time consuming to try them all and compare. I've been using the excellent resources at http://www.opensourcecms.com to try some of them out... it lets you try any one you like without installing it on your server first. I'm currently leaning a bit towards Geeklog and Jupiter CMS and an IT guy I asked recommended e107 but lots more testing to do before I know which is the right one for sure. Furthermore I know I'm not the only one on this list that has entertained the idea of a linux forum dealing with audio software. If any of you read this and might be interested in putting in a hand to try and build such a monster please feel free to email me directly if you like. Thanks for all the fish, Jon Hoskins

17 years, 2 months

[linux-audio-user] Shaving CPU cycles

by Ken Restivo

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've run into an 80% DSP problem on my machine. This poor little PC doesn't have enough CPU power to run all the softsynths and LADSPA plugins I tend to use. When JACK gets to 80% DSP usage, all hell breaks loose. If I up the latency, then my CPU usage goes down, but the system is unplayable (too much latency). So. What kinds of things would be best to try to squeeze some life out of this underpowered 1.66Ghz Core Duo? I have a list, but I'm not sure which would be the first, second, third order improvements and I'd like to try to conserve time, especially if the end result will end up being that I have to sell this PC and get a new one anyway. Do I have the priorities right? 1) Try jackdmp instead of jackd 2) Try DRM or some kind of accelerated graphics for Xorg 3) Blindly chase "latest and greatest" versions of things like kernel 2.4.20, latest jackd, svn freebob, etc. 4) Try messing around with PCI bus latency I'll try as many of those as you-all think might be worth pursuing, before moving on to hardware solutions: 5) Replace the CPU on this box with a 2.16Ghz Core Duo. 6) Get rid of the machine completely and get a new PC with 2Ghz+ Intel or AMD dual CPU, and PCI audio card not firewire. - -ken -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFFwG6We8HF+6xeOIcRApf6AKCmlNbp8oWhGP2THi+va2VqfKWGiACgqitk NWSBmNwyws/atURFGYO+FyE= =MouT -----END PGP SIGNATURE-----

17 years, 2 months

Re: [linux-audio-user] Shaving CPU cycles

by Sampo Savolainen

Quoting Ken Restivo <ken(a)restivo.org>: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > I've run into an 80% DSP problem on my machine. > > This poor little PC doesn't have enough CPU power to run all the > softsynths and LADSPA plugins I tend to use. When JACK gets to 80% DSP > usage, all hell breaks loose. If I up the latency, then my CPU usage goes > down, but the system is unplayable (too much latency). > > So. What kinds of things would be best to try to squeeze some life out of > this underpowered 1.66Ghz Core Duo? I have a list, but I'm not sure which > would be the first, second, third order improvements and I'd like to try > to conserve time, especially if the end result will end up being that I > have to sell this PC and get a new one anyway. Do I have the priorities > right? There's a lot of things to do. "underpowered" is not a word I would use for a core duo processor. My list of things to do first would be: 1) SSE. Make sure all synths and ladpsa plugins are compiled with SSE support. (This is rarely done in distributions because SSE is not supported by all currently used x86 processors, we're getting quite close to that though) 2) Sharing. Instead of running separate reverb plugins on each track, create reverb buses for a small number of different types of reverbs. Usually there are two: one long reverb and a short one. Then use sends to send appropriate amount of the signal from each track to the effects bus. You can share other effects too, reverb is a natural choice for "sharing" as many mixes sound actually better with a coherent reverb space instead of having multiple varying reverbs. 3) Freezing. This means that in for example ardour, you can pre-render the effect of the plugins on a track. This can lower the DSP usage very much if you have tracks with heavy processing. > 1) Try jackdmp instead of jackd That might help. > 2) Try DRM or some kind of accelerated graphics for Xorg Accelerated graphics is a good idea. > 3) Blindly chase "latest and greatest" versions of things like kernel > 2.4.20, latest jackd, svn freebob, etc. I expect you don't mean 2.4. "Blindly chasing" is never a good idea, but there has been a lot of work towards better realtime performance in the latest kernels. I strongly urge you to try a newer kernel if you are running < 2.6.17, especially if it's without ingos' RT patches . > 4) Try messing around with PCI bus latency That might help a bit. But keep in mind that DSP usage of 80% is really high. It doesn't leave your computer much headroom for other tasks, or even plugins which might periodically use more CPU cycles than normally (=which is in fact a sign that a plugin is not "academically" real time safe). In my opinion 80% is about the maximum of DSP usage you can expect to be stable to work with. Remember that with the rest of the time, the CPU, operating system and the software need to do tasks like update GUI windows, run the disks, complete network traffic. Without time to complete these tasks, your computer will be unresponsive (altough it might still will be processing audio, though ;) ). Sampo

17 years, 2 months

[linux-audio-user] jack transport slider

by Robin Gareus

Here's some glade-gtk application to implement a slider for jack transport: http://gjacktransport.sf.net/ to sum it up: * slider to control jack-transport * set slider start and end points in various units * memory: remember slider start/end points * auto-zoom and reposition slider * LASH support there are some open-loops: - missing/faulty drop-frame TC for videoframe units. - insane but intuitive zoom behaviour - no config-file, simple CLA parser,.. see TODO file none of which bugs me at the moment. It works fine on debian boxes. I'll wrap it up for a binary release and LAA. - feedback welcome. #robin

17 years, 2 months

[linux-audio-user] Reminder::Linux Audio Conference #5 (LAC2007): Call for Papers - Music

by nescivi

Dear all, this is the second call for papers for the 5th Linux Audio Developers Conference (LAC2007). This is a reminder since some people might not have received the last call or might just have forgotten about the deadlines by now (08 Jan 2007 : Deadline for submission of papers, worshops, tutorials, demos, hands on demos and music). The conference is organized by the TU-Berlin in cooperation with people of the Linux Audio Developers mailing list, the music festival Inventionen 2007 and the Humboldt University of Berlin. The LAC2007 is taking place at the TU-Berlin, Germany from the 22nd - 25th of March 2007. We have introduced some new tracks. Besides the category for papers, demos and workshops, calls for tutorials and hands on demos have been added. The tutorials aim is to give new (potential) users an overview of the possibilities of Linux Audio Software and how to get started. The LAC2007 provides a computer pool (LA Pool) where developers can give an introduction to their software and where participants can try out Linux Audio Software during the conference. This has been combined in the call hands on demos. Since the TU-Berlin is installing a new Wave Field Synthesis (WFS) system the call for music has been extended by a call for compositions for this system. Music that can be used for radio airplay can be submitted, and will after acceptance by the Campusradio of the TU Berlin, be played during the conference. More detailed Information can be found in the 'Call for Papers' attached to this email or on the website at: www.lac.tu-berlin.de We are looking forward to many interesting submissions for the Linux Audio Conference 2007 and hope to see you in Berlin in 2007! Please feel free to forward this email to anybody who is interested. On behalf of the LAC2007 organisation team, Marije Baalman and Simon Schampijer

17 years, 2 months

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Linux-audio-user February 2007