similar to: Gain control

>Fromwhat you describe, your comp/limiter can't possibly be working correctly. It should be the last unit in line before the sound card, and needs to be adjusted properly. You also need to balance the levels on your mixing board (so that the correc t level comes at predictable place on the slider). It might be worthwhile to find someone with some sound-mixing or radio engineering experience

On Sun, Jun 24, 2001 at 01:51:05AM -0700, Dave Hayes wrote: > How are people doing gain control, out of curiosity? > > I know boxes exist (anyone have names and mfrs?) that ride the gain > for big radio stations, ensuring that there is no distortion and > raising the volume of songs recorded at a lower volume. I'd probably > buy one if I knew what to buy. This is a bit

Hi: I want to stream audio from my soundcard, but I'd like to apply some dynamic compression to it first (ala winamp's audiostocker plus shoutmuxer thingy). I heard rumours of compression in liveice, but Iv'e been told that this only works when it plays from prerecorded MP3, not the live input. I'd like to do this with darkice, but can't see how it could be done. I guess

> we have a web-based station running liveice and aumix and the levels are all > over the place. is there a way to do automatic gain control on the soundcard > input? > > -peter Run the signal through a Compressor/Limiter before sending it to your soundcard. I use Behringer Ultra-Dyne Pro DSP9024. Very nice. If you want to buy it look here:

On Wed, Nov 14, 2001 at 08:36:36AM -0800, William Goldsmith wrote: > >From what you describe, your comp/limiter can't possibly be working > correctly. It should be the last unit in line before the sound card, and > needs to be adjusted properly. You also need to balance the levels on your > mixing board (so that the correc t level comes at predictable place on the > slider).

Sounds like you need to fix the problem at the source first: balancing the levels going *into* your mixing board so that they're not all over the place coming out. You'll never fix that with any hardware or software device. I use a program called ecasound to do software dynamics processing, but you'd have to hack liveice pretty extensively to use it in that situation. Software

> As you can tell, the AAGC integration with speex was really a classic > hack. Instead of re-creating the hack, what's probably best here is to > integrate AAGC back into speex, and have a proper API. Agreed here. If you can come up with a clean patch to add that feature, it's something I'd like to see in Speex. > For those of you just tuning in, what I call

Hi, I've had a small case with noise suppression and AGC. I have a fairly noisy environment here, and with the default parameters, noise suppression works fairly well while I talk. However, when I shut up, AGC starts slowly increasing the gain until it has amplified whatever noise is left to levels about equal to having no filtering at all. As soon as I talk, AGC backs down fairly quick

>> Yes, after I stop speaking, the noise slowly starts climbing again, and >> if I peek at st->agc_gain, that's slowly climbing too. I think part of >> the trouble is that the noise in here isn't uniform white noise; there's >> traffic outside the window and people walking in the hallway outside my >> door. Each little event is enough to cause the AGC

I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from "certain" interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is

Jean-Marc Valin wrote: >> I've had a small case with noise suppression and AGC. I have a fairly >> noisy environment here, and with the default parameters, noise >> suppression works fairly well while I talk. However, when I shut up, AGC >> starts slowly increasing the gain until it has amplified whatever noise >> is left to levels about equal to having no

I have a recorded conversation from an analog trunk. As usual one side is stronger that the other one. In my case, the gap between signal levels are even bigger. How does speex AGC preprocessor will perform on this type of audio recording? Maybe I am wrong and AGC is not really what I need to equalize the two persons in my phone conversation? As I Understand, AGC will perform better if each

the ACG function that the compressor provides is limited at best. its adjustment is way too audible to make full use of it in balancing levels. yes, it is last in the chain. i'm just wondering if there's anything out there in the way of software. if a minidisc recorder can do it, why can't a $1000 pc? -p > From: "William Goldsmith" <wildbill@kpig.com> >

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Jean-Marc Valin escreveu: > > Well, I can change the defaults and then people with different setups > (e.g. no noise and low input volume) will ask me to change it the other > way... In the end, the only fix is to improve the AGC adaptation > decision, including (but not only) the VAD. Could that be a run time variable? So it could be adjusted somehow by a user interface, for

> I hate to be a talker and not a do-er, but I won't be able to write this > myself, probably someone on the iaxclient team could do it. Anyway, let me know if/when someone's working on that. >> Hmm, or does that mean the analogue AGC is actually completely >> independent from the "real" AGC. Any thoughts? >> > > It's actually a bit more

agc_gain seemed to fit with the idea of what I wanted to do, it was easy to understand its units and behavior, and freezing it produced the desired results. Also I wanted to cap it, so that's done at the same place, and that definitely works. All I want to do is be able to freeze AGC adaptation and put an upper bound on the AGC (for example, 2x amplification). Both of these things seem

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

On Thu, 4 Apr 2002, Akos Maroy wrote: > can you tell me more about these LADSPA plugins? LADSPA stands for Linux Audio Developer's Simple Plugin API (see http://www.ladspa.org/). Basically, it was pointed out on the linux audio dev (LAD) mailing list that numerous programs were using plugin architectures and all were different. So they fleshed out a plugin API and the rest, as they say,

I think you'll have to modify Speex to get the functionality you're looking for. I've made a few simple modifications to the AGC to prevent it from 1) exceeding a specified level of amplification and 2) enable and disable adaptation, so I can freeze it at a certain level while speech is not detected. It's mostly just a matter of doing this at the end of speex_compute_agc():