Linux-audio-dev July 2005

linux-audio-dev@lists.linuxaudio.org

72 participants
64 discussions

[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, 4 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, 4 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, 4 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, 4 months

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Linux-audio-dev July 2005