Linux-audio-dev

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4 participants
7441 discussions

Re: [linux-audio-dev] desktop audio resumed

by Jan Holst Jensen

--- Lee Revell wrote: > So all the advice needed is "make sure you use the > latest ALSA". Just got dmix configured, inspired by this thread. It is really simple using the steps in http://alsa.opensrc.org/index.php?page=DmixPlugin. Even Flash-stuff in Firefox behaves nicely now (just start Firefox with "aoss firefox")! Have I understood it right that ALSA 1.0.9 and later sets up dmix per default, also for aoss, so that you can play aoss stuff and multiple alsa streams simultaneously ? That sounds exactly like what we need to get stuff to "just work". Cheers -- Jan __________________________________ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail

19 years, 2 months

[linux-audio-dev] Re: [linux-audio-user] [ANN] Snd-ls V0.9.4.3, Mammut V0.20, Ceres V0.43 and Snd RT-extension documentation updates.

by Kjetil Svalastog Matheussen <k.s.matheussen＠notam02.no>

Dave Phillips > > Please tell me more about snd-rt and CM, I'm very interested. > Hmm, well, theres no special support in CM for snd-rt,but common music (at least partly) can be loaded into SND, so you can use some of common musics functionality to schedule realtime events and control various things. I don't know CM very well though.

19 years, 2 months

[linux-audio-dev] New ladspa plugin - "BuzzKill" - Request For Comments

by Daniel Lyons

Hello, I made a new plugin based on the cmt lib and swh libs called buzzkill. It's a simple "reverse hard limiter" - it kills amplitudes below a certain level. It works good for two purposes: Eliminating "noise" or "hiss" at about -50 or -47 DB, which was my intention. But it also works good put before guitar distortion (I was using the CAP libs tube amp sims) - eliminating the noise you get (I've got an accoustic-electric plugged in through a m-audio delta card hooked through jack-rack and the CAP's tube amp sim... then I had BuzzKill set at about -25 to -20). Compilation: I just copied and modified the stereo amp from cmt to make this, I put the "identifier numbers" to 10,000 and 10,001 (i dunno whatchya call'em - each plugin's got a unique one). So, if you throw this file into the cmt libary, add it to cmt.h or cmt.cpp right, and add an "-lm" to the makefile in the right place it compiles (call it "alpha version" if you have no idea what I'm talking about). So - the guitar usage sounds good, right? But, if you make a sound right about at the threshold (which you get a bit of, being sloppy on the fretboard and maybe a beer here or there), it cracks up a bit, sounds weird. So - rather than use a "look ahead" type algorithm like I saw i nthe limiters, I thought about something simpler to understand (at least for me).. which was having the parameters be two dB values (let's call'em i1 and i2 between friends)... then, having a linear scale from the input to the output as follows: if input is of i1 amplitude, 0 dB is output. If input is between i1 and i2, output is linearly scaled from 0 to i2 (let's see, i'd say output=input*(i2/i1)+i1 .. i think that's right) If input is greater than i2, output equals input It gives you a rough transition at input=i2 but I think it'll be "good nuf, ya, let's jam, ya know" I dunno, that's what I was thinkin.. hey, it's simple, right? :) Oughtta be fast too. Cheers, Dan -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

19 years, 2 months

[linux-audio-dev] Why alsamixer starts with all the volumes settings in 0?

by Lisandro Damián Nicanor Pérez Meyer

Hi everyone! I was just wondering why alsamixer starts with all volume settings in cero. I guess there should be an aswer to this, so I decided to ask you. What I didn't know was if to ask here or to lau's list, so I decided to go directly to the designers :-) Also, it is possible to, at least, lstart with the main volume and pcm/wav just a little bit raised? I think this would help when configuring alsa, specially for newbies. Thanks in advanced! Cheers, Damian.- Outside of a dog, a book is man's best friend. Inside of a dog, it's too dark to read. -- Groucho Marx

19 years, 2 months

Re: [linux-audio-dev] jack_convolve-0.0.10, libconvolve-0.0.3 released

by Chris Cannam

Florian Schmidt: > P.S.: I am also currently working on a qt app which > uses libconvolve Any interest in making a DSSI plugin? Chris

19 years, 2 months

[linux-audio-dev] [ANN] Snd-ls V0.9.4.3, Mammut V0.20, Ceres V0.43 and Snd RT-extension documentation updates.

by Kjetil Svalastog Matheussen

***************************************************************************** 1. Snd-ls v0.9.4.3 --------------- Released 30.6.2005 About ----- Snd-ls is a distribution of the sound editor Snd. Its target is people that don't know scheme very well, and don't want to spend too much time configuring Snd. It can also serve as a quick introduction to Snd and how it can be set up. Changes from 0.9.3.0 to 0.9.4.3 -------------------------------- -Upgraded various rt-stuff. -Upgraded various rt-stuff++ -Removed jack_set_server_dir guile-binding from rt-engine.scm, because its removed from the newer versions of jack. -Updated SND to v7.14 from 19.6.2005. Many important changes. http://www.notam02.no/arkiv/src/snd/ ***************************************************************************** 2. Mammut V0.20 ------------ Mammut will FFT your sound in one single gigantic analysis (no windows). These spectral data, where the development in time is incorporated in mysterious ways, may then be transformed by different algorithms prior to resynthesis. An interesting aspect of Mammut is its completely non-intuitive sound transformation approach. Changes 0.18 -> 0.20 -------------------- -Fixed a bug in the Makefile caused by a fix in the previous release. -Fixed some bugs in the Makefile. http://www.notam02.no/arkiv/src/ ***************************************************************************** 3. Ceres V0.43 ----------- Ceres is an advanced program for displaying sonograms and for sound effects in the frequency domain. And more. Changes 0.42 -> 0.43 -------------------- -Fixed some bugs in the Makefile http://www.notam02.no/arkiv/src/ ***************************************************************************** 4. Snd RT-extension documentation ------------------------------- The realtime extension for SND makes it possible to do signal processing in realtime using guile and a special scheme-like compiled language. It has many features, as jack-support, CLM, a realtime scheduler, a compiler, busses, dynamic patching, shared variables, ladspa and more. It can also be made to work with Common Music. http://www.notam02.no/arkiv/doc/snd-rt/ --

19 years, 2 months

[linux-audio-dev] Midi/OSC help

by Olivier Guilyardi

Hi, I am thinking about adding midi or/and osc support to Jackbeat (http://xung.org/jackbeat). My idea is to use a hardware midi controller such as the evolution uc33 to control pitch, volume and muting on every tracks. This would help using Jackbeat for live performances. I have never coded with midi, neither with osc. What's the fastest and smartest way to achieve this idea ? Regards -- og

19 years, 2 months

[linux-audio-dev] help with writing a plugin (part 2)

by Artemio

Dear friends, thanks again for your help with my ports definition yesterday - with it, I created a clipping booster plugin with three control ports and it works perfectly. However, today I added the fourth port, following all rules I learnt, but I get the same segfaults as yesterday. If you could have a look at the sources (attached), it would be of fantastic help for me. Once again, please forgive me, I am a real newbie at this and things that are very obvious for you are not such for me... But as I understand all these bells and whistles, I will be more silent ;-) Thanks a lot again. Regards, Artemiy.

19 years, 2 months

[linux-audio-dev] help with multiple control ports

by Artemio

Hello all! I have a problem with creating a plugin with more than one control port. In the attachment you'll find "booster-simple.c" which has one "Gain" port and my attempt to add a second port in "booster.c". For some reason the latter cannot be used - the sources seem to be correct, it compiles fine, but then if I test with "applyplugin" it segfaults and other hosts complain about a malloc() error and hang. Debugging shows that ports are not properly connected: [connectPortToBooster] setting BOOSTER_INPUT1 [connectPortToBooster] setting BOOSTER_OUTPUT1 [runMonoBooster] applyplugin: booster.c:112: runMonoBooster: Assertion `psBooster->m_pfGainValue != 0' failed. ...but I cannot investigate further as my C is very basic. I would really appreciate any help. I based my plugin on the amp.c from the SDK, and I see that in example plugins like simple delay (where there is more than one control port) memory allocation is done differently, but I just don't know how to properly apply these techniques to my plugin... Thanks very much! Artemiy. P.S. I've sent a message with a 50Kb attachment and it's in moderation queue, but these sources are newer.

19 years, 3 months

[linux-audio-dev] All things LADSPA, your input please

by Tim Goetze

Greetings, you may have seen the recent mail on this list looking for an author doing a LADSPA article in a german Linux magazine; I've decided to step in. Now, while I do have some knowledge of the technical nature of LADSPA I can't truthfully say I know all the plugins and hosts out there as well as they deserve. So, I'd love to hear from you: what do you write or use (hosts and plugins, realtime or offline, a shameless plug for your efforts is welcome too!), any issues you take with it, LADSPA vs. VST, the DSSI expansion etc etc; in short, anything you find worth noting is appreciated. Thanks, Tim

19 years, 3 months

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Linux-audio-dev