Linux-audio-dev

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7516 discussions

[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, 10 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, 10 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, 10 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, 10 months

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

by Betty Woessner

Apologies if you receive multiple copies] +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 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, 10 months

[linux-audio-dev] Audio Driver Sample Code

by oscar si

Hello: Could anyone please tell me if there are sample driver codes for PCI based sound card that are not using either alsa or oss? Thank you very much, Oscar __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 10 months

[linux-audio-dev] Daemon mode

by Patrick Shirkey

Hi, I am adding a Daemon mode to jackEQ and have most of the code in place now. I am stuck at the point where the daemon starts up and keeps going. Essentially I need the daemon mode equivalent of gtk_main() that keeps ticking over until the app is told to stop or otherwise shutdown. I have not threaded the app at this point so hopefully there is a very simple solution. Any thoughts on this are appreciated. Cheers. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com Http://lau.linuxaudio.org - The Linux Audio Users guide ======================================== "Anything your mind can see you can manifest physically, then it will become reality" - Macka B

18 years, 10 months

[linux-audio-dev] kernel 2.6.18, -19 etc

by Jens M Andreasen

Now that mingo's (et al) RT patches are coming into mainstream, what is the corporate rationale behind it and the running order of urgency? I am fishing for some information on; if it is the disk-drives, the network drivers, the usb stack or something else that I am too ignorant to have noticed? What worries me the most, is corner-cases on network, blocking multiple cpu's. FYI: I have been asked to set up a bunch of Xeon boxens to serve a VPN over WiFi, they/we are currently using kernel 2.6.17 (debian etch as of today.) So this is not stricly about audio, but I would guess all of us would like some clarification here. After all, we have followed the evolvement of these patches for quite a few years by now. -- mvh // Jens M Andreasen

18 years, 10 months

[linux-audio-dev] Grabbing sound using Roland Edirol FA-101

by Helge Fredriksen

Hello! Did any of you guys ever tried to use the OpenAL API interface towards a Firewire sound device like FA-101? I see that it appears on your lists as a device that has drivers on Linux. Best regards, Helge Fredriksen

18 years, 10 months

[linux-audio-dev] linux-sound porting & documentation proposal (Was: Attracting more Linux audio developers)

by Ivica Ico Bukvic

How about we stop the seemingly endless discussion and instead all roll-up our sleeves and actually do this? Here's what I offer on behalf of linuxaudio.org: 1) generous hosting space 2) virtually unlimited bandwidth 3) docs.linuxaudio.org and apps.linuxaudio.org domains 4) accounts to maintainers 5) server support as needed What we need volunteers to do: 1) port Dave's pages over into a legible and appealing format to apps.linuxaudio.org 2) cross-link those pages to docs.linuxaudio.org page (in addition to apps homepages) which will be a wiki with documentation templates and standardized layout 3) need to design an appealing interface for both sites (hence consider this an open call for volunteer designers) -- this cannot be emphasized enough: we do not want an ugly, plain website, but a nice inviting and user-oriented resource. 4) create a generic wikipedia entry which gives a summary, philosophy, and notable achievements of the linux audio scene and provides critical links (hence it would be used as a portal rather than an exhaustive resource for a moving target which would never fly with the wikipedia editors anyhow) On a side note, here's another offer: As per my discussion with Joern, on behalf of linuxaudio.org I also offer free unlimited space for porting over LAU and LAD lists to lists.linuxaudio.org in hopes that we continue consolidating these invaluable resources. Maintainers will be given appropriate access privileges etc. Any takers? Best wishes, Ico

18 years, 10 months

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