Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

4 participants
7502 discussions

[linux-audio-dev] Happy LAC2007!

by nescivi

Happy new year to you all! I just wanted to remind you all that the deadline (8th of January) for the LAC2007 is getting nearer, so now is the time to still write that paper, mail that cd, or make that proposition for that tutorial. Detailed info: http://www.kgw.tu-berlin.de/~lac2007/index.shtml Sincerely, On behalf of the Linux Audio Conference 2007 Orga Team, Marije Baalman

18 years, 6 months

[linux-audio-dev] 2007..

by Fons Adriaensen

Hello all, First of all, my best wishes for 2007 to all Linux Audio Developers ! 2007 will be a special year for me. As some of you already know, I said goodbey at Alcatel Space two months ago, and starting 8 Jan 2007 I'll be working at LAE - Laboratorio di Acustica ed Elettroacustica - <http://www.laegroup.org/> in Parma, Italy. LAE is an acoustics and electro-acoustics research and consultancy lab operated by the university of Parma and by three companies active in the area of acoustics and audio. My activities in LAD will continue of course, after maybe a short break while the dust of moving to Italy settles. Looking forward to meet you all at LAC2007 in Berlin ! -- FA Lascia la spina, cogli la rosa.

18 years, 6 months

[linux-audio-dev] Re: [Alsa-user] alsa, oss and jack

by Dominique Michel

Le Sun, 31 Dec 2006 02:21:20 -0500, Lee Revell <rlrevell(a)joe-job.com> a écrit : > On Sat, 2006-12-30 at 18:06 +0100, Dominique Michel wrote: > > The result was at the sound from flash 9 > > (some kind of worst case scenario) when listening to > > http://concerts.wolfgangsvault.com/ConcertDetail.aspx?id=2229|3385 was > > hopping all the time, when it's hopping without any .asoundrc only > > when I open or close a windows or when I shift from a virual desktop > > to another one. > > Did you install the latest Flash 9 beta? Earlier versions did not work > well for me. > > Lee > Thank you Bill and Lee. It is the last beta from portage in gentoo, that is 9.0.21.78 and it is the last version (installer 112006). After restarting not only alsa but the whole system and running flash on the same website with just 2 open windows in firefox, the sound work much better. The processor load was very high on the first test. And I am able to record from audacity without any .asoundrc file. Is it possible to run flash with jack? My wm is currently fvwm-crystal. I try to run arts with konqueror and arts with jack as sound server. I can see arts in the connection windows of qjackctl, the sound is here but already hoping with a processor load of 17% and I don't have flash in qjackctl. So, is it possible to run flash 9 with jack and firefox? Cheers, Dominique

18 years, 6 months

[linux-audio-dev] ALSA performance issues?

by Mario Lang

Hi. I just tried to get sound working on a LinkSys NSLU2 (ARM) running Debian Etch using a USB sound card. What I discovered I find pretty strange, and would like to know some more details. Apparently, all ALSA native clients don't manage to play sound click-free, they actually have underruns all the time. However, if I use OSS (via the snd-pcm-oss module) and set libao to use the oss driver, I get perfect playback with about 20% CPU use at maxiumum by the user-space app playing/decoding files. This is a bit strange, isn't it. OSS-emulation actually uses the same kernel code to access the actual hardware. I am just very vaguely familiar with the whole ALSA architecture, but it feels as if the problem here actually lies inside the user-space alsa libraries? Did anyone ever see such an effect, and investigate more closely what is the reason for the difference? -- CYa, Mario | Debian Developer <URL:http://debian.org/> .''`. | Get my public key via finger mlang(a)db.debian.org : :' : | 1024D/7FC1A0854909BCCDBE6C102DDFFC022A6B113E44 `. `' `- <URL:http://delysid.org/> <URL:http://www.staff.tugraz.at/mlang/>

18 years, 6 months

[linux-audio-dev] nohid boot option ?

by Dave Phillips

Greetings: My ancient 800 MHz machine has dead PS/2 ports so I plugged my mouse and keyboard into its available USB ports. Alas, the keyboard disappears after selecting my kernel image (Demudi) in grub. I'm away from the machine right now so I don't have the exact text of the error message. However, I do recall that it refers to a USB-related IRQ error. I tried to install the 32-bit version of 64Studio, but the error occurs during the configuration process: As soon as I press Enter at the boot prompt the keyboard is no longer present to the system. :( I Googled for some more information regarding the problem and found that it is in fact a known issue. However, none of the proposed solutions worked for me. I also sent messages to the 64Studio dev list and to Free and Daniel, but I've heard nothing back from them or the list (I think my email to the list isn't getting there). One thing does work: Dynebolic includes a nohid boot parameter that does the trick ('linux nohid'), so I can still boot into the machine and access my data. So, my question is: What's Dynebolic doing that the other distros aren't ? I'd surely like to boot directly into my system, and I'd like to install 32-bit 64Studio, but I can't do either until I solve this problem. Any suggestions ? Best, dp

18 years, 6 months

[linux-audio-dev] plugin loaders

by Anthony Green

I understand that LADSPA and friends specifically exclude any functionality around how to find and load plugins, but it seems that a lot can be gained by introducing some standards in this area. As a package of audio apps/plugins for a Linux distro, here are two of the problems I see: 1. Applications are often hard-coded to look in /usr/lib/ladspa (for instance), when many systems may require that libraries live somewhere else (like /usr/lib64/ladspa for x86-64, or /usr/lib32/ladspa for n32-ABI MIPS Linux). I've had to patch a lot of apps for x86-64 Fedora. 2. We build binaries for the lowest common denominator, so the plugins you'll find in Fedora, for instance, don't take advantage of SSE hardware or instruction scheduling for different processors. This can make a huge difference. What would be nice is if we could distribute an RPM containing a plurality of plugin builds, and then have the application load the plugin matching the capabilities the execution platform. Has there been any discussion around creating a plugin locator/loader library? It would be nice if one could be written and then widely adopted by app writers. (I'm not volunteering!) AG

18 years, 6 months

[linux-audio-dev] Re: LADSPA_Data audio values

by Ben Loftis

> Message: 3 > Date: Thu, 21 Dec 2006 21:19:04 +0000 > From: iain(a)g7iii.net (Iain Young) > Subject: [linux-audio-dev] LADSPA_Data audio values > To: linux-audio-dev(a)music.columbia.edu > Message-ID: <20061221211903.GA14158(a)g7iii.net> > Content-Type: text/plain; charset="us-ascii" > > Hi All, > > I'm trying to write a LADSPA plugin to do the following: > > Given n inputs (lets say 4), and one output, It should present the > highest "priority" input on the output. Input 1 is the lowest priority, > with Input 4 being the highest. > > In order to write this in code, within the runPriomux function, I do > this: > > y=-1; > for (i=0; i<=3; i++) > { > pfBuffer=*(pfInput[i]++); > if (pfBuffer > -1.0f) > y=i; > } > if (y!=-1) > *(pfOutput++)=pfBuffer; > > Hi Iain, I don't know your application, but this looks wrong. If you are trying to choose between various microphones, and only output the microphone with the highest priority that has audio on it, (i.e. an automixer), then you're going to have to do a bit more work than this. Here are the immediate problems I see (Chris Cannam already indicated the first 2) 1) 0.0f indicates silence, not -1.0f 2) A microphone input will have some noise on it, you'll need an adjustable threshold value which separates the "on" state from the "off" 3) You can't do this on a sample-by-sample basis, you need some time averaging: 3a) you need to implement a rectified, running average value of all 4 inputs 3b) you'll need to mix the values of all four buffers into the output, with a smoothly changing level control on each input controlled by the logic 4) Make sure the "decay" of the envelope is kinda long ...this will keep a speaker from getting cut off if someody else coughs or whatever There are other subtleties that apply to an automixer, but this is enough to get started. -Ben Loftis

18 years, 6 months

[linux-audio-dev] [ANN] das_watchdog V0.2.5 and jack_capture V0.9.3

by Kjetil Svalastog Matheussen

Download from http://www.notam02.no/arkiv/src/ das_watchdog 0.2.5 ================== 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 0.2.4->0.2.5 -------------------- *Let the test thread run with SCHED_FIFO priority using the lowest priority. Should hopefully stop all the unnecessary reports. (This change has been tested quite thoroughly) jack_capture v0.9.3 =================== jack_capture is a program for recording soundfiles with jack. Its default operation is to capture whatever sound is going out to your speakers into a file. This is the program I always wanted to have for jack, but no one made. So here it is. Note: Anyone using 0.9.2 should upgrade to 0.9.3! 0.9.2 will most likely hang during startup. :-( (I'm going to start testing my software before releasing from now on, this one was very embarrasing.) Distros: If there is a system for doing so, you should mark 0.9.2 as unusable. Changes 0.3.9 -> 0.9.3 ----------------------- *Fixed horrible deadlock in 0.9.2. Bug found by Ken Restivo. *Fix for a potensional deadlock. *Added the --silent/-s argument. *Some smaller fixes. *If recording to wav (the default) and the the 4GB limitation is reached, automatically close the file and continue writing to a new file with an autogenerated name. *Added the --version/-v argument. *Changed default number of zeros in the autogenerated filename to 1. *Better error output. *Autogenerate code to check if various formats are supported by sndlibfile. *Bumped version up to 0.9. jack_capture should reach a 1.0 state quite soon. *Changed the name of --recording-time to --duration to match -d.

18 years, 6 months

[linux-audio-dev] LADSPA_Data audio values

by iain＠g7iii.net

Hi All, I'm trying to write a LADSPA plugin to do the following: Given n inputs (lets say 4), and one output, It should present the highest "priority" input on the output. Input 1 is the lowest priority, with Input 4 being the highest. In order to write this in code, within the runPriomux function, I do this: y=-1; for (i=0; i<=3; i++) { pfBuffer=*(pfInput[i]++); if (pfBuffer > -1.0f) y=i; } if (y!=-1) *(pfOutput++)=pfBuffer; i is the input channel (0-3, not 1-4, but you know what I mean) y is what is supposed to be the highest priority channel with audio on it. Now..here's where Im confused. I thought the values for audio samples in LADSPA were -1.0f...1.0f (with -1.0f being infinity). This doesn't seem to work, as the above code always appears to think that the 4th input (i=3) has data on it, even if it's muted in ardour (via another bus that's single output is connected to input 4...) I've also looked at the code of meterbridge, and tried using -70.0f, but with the same problem...Anyone want to plant me a clue on what I'm doing wrong ? I've attached the entire plugin source below. It's really been ripped off of the example amp in the LADSPA SDK I'm afraid.... Thanks in advance for any help, this last bit is driving me mad, now I've worked out how to code plugins! All the Best Iain

18 years, 6 months

[linux-audio-dev] 31st International AES Conference - New Directions in High Resolution Audio

by Betty Woessner

+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 31st International AES Conference - New Directions in High Resolution Audio June 25-27, 2007 London, UK +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Submissions site now open! Please let the conference organizers know of your intent to contribute or attend. Further detail available at http://www.aes.org/events/31/ The 31st International Audio Engineering Society Conference, entitled New Directions in High Resolution Audio, will be held at Queen Mary, University of London, from June 25th-27th, 2007. This Conference is concerned with the promotion and delivery of high resolution audio, by maintaining quality throughout the recording and playback chain with current and future technologies. It reflects the tremendous recent growth of high resolution audio techniques and products intended for use throughout the audio recording and playback chain. However, issues remain on how to avoid bottlenecks where quality is compromised, and how to maintain and encourage high resolution audio in an everchanging marketplace. These concerns are of interest to the audio engineering, recording and production industries, as well as to education and academia. We aim to provide a place for the exchange of news, issues and results, by bringing together researchers , developers, educators, students and professional users, working in fields that contribute to high resolution audio, to present original theoretical or practical work. It also serves as a discussion forum, provides introductory and in-depth information in specific domains, and showcases current products. The 31st AES Conference solicits contributions to the field of high resolution audio, including, but not limited to, the following domains and topics: High resolution recording issues * Bandwidth, sampling rate, dynamic resolution and spatial representation. * Analog and digital recording equipment Processing, manipulation and preparation of high resolution signals * High quality analog design * A/D and D/A conversion * Format and sample rate interconversions * High-resolution signal processing, 1 to n bit paradigms * Spatial audio, virtual and acoustic spaces, virtual image manipulation Storage of high resolution audio * Current and future storage technology for high-density audio data * Overview of high resolution consumer formats * Copy protection * Access speed, data reliability, bit packing and compression * Professional archival formats and future-proofing Electronic delivery of high resolution audio * Network (wired and wireless) systems/delivery of high quality audio content * Current and future formats for streaming and file delivery of high resolution audio * Bandwidth * Digital radio and broadcast applications including multichannel audio Maintaining quality at playback * Evolving technology for multiformat players * PC-based playback: configuration and interfacing * Converter technologies * Jitter/interface/power supplies and other related forms of noise * Digital amplification e.g., Class-D amplifiers Perception * Perceptual modeling. * Objective evaluation and subjective performance of high-resolution audio * Objective measurement and resolution issues * Subjective procedures and evaluation psychology +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ SUBMISSIONS Submissions, which will be peer-reviewed, may be in the following categories: - paper (to be presented in the main sessions) - poster or demonstration (to be presented in the poster sessions) - original composition demonstrating features of high resolution audio (to be showcased throughout the conference) - tutorial, panel or workshop proposals (see conference web site for details) Recording companies. publishers, software and hardware developers, etc., are invited to contact the programme committee regarding exhibition space. +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ DEADLINES Deadline for submissions of tutorials, panels and workshops: Feb. 2nd 2007. Deadline for submissions of papers and posters/demos: Feb. 2nd 2007. Deadline for exhibitor space: June 1st 2007. Please see the conference website for more information; http://www.aes.org/events/31/

18 years, 6 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev