similar to: More Speex (pre-processing) Options

I don't think the echo canceller works yet. The best I could get it to do is trash my audio. If it does work, some sample code demonstrating its use would be really appreciated... Here is what Jean-Marc has said about it in the past: > What should I use for a filter length for speex_echo_state_init()? "Well, it depends on your problem. The filter length represents the maximum

Tom, You can look in my archives for my posts on this topic, but what I've found is that the echo canceller works to some extent, but if you're also using the other preprocessing (which you'd definitely need to do _after_ ec, otherwise ec probably won't work at all), the AGC function ends up reversing the effect of ec. That is, while ec reduces the echo, it doesn't

This is only remotely related: just out of curiosity, does speex_preprocess work very well for you ? It only works to certain extend for me, and does not seem to completely remove noise from sound card. Even if you unplug the microphone, it still could not get rid of the noise. Any idea how it could be totally eliminated ? <p>Thanks <p><p>-----Original Message----- From: Tom

> So I have had a chance to delve deeper into the new pre-processing > code- I was also able to get VAD, AGC, Denoising, etc working inside of the > current release candidate (1.0.3). For some reason, the 1.1.2 > release didn't properly encode/decode in our system (on windows) > and I haven't had time to determine why. In any case, all of the features > jointly work

Hi, Thanks, that gave me some ideas. I have tried calling speex_echo_cancel and speex_preprocess separately and jointly, with a number of different filter lengths, and using the cancelled and non-cancelled data as input to the pre-filter, with and without the residual echo, etc. but there doesn't appear to be valid data in the echo or the cancelled frame any way I try it. Back to the

Hi, I have speex running as a part of a voice conferencing app. Well, one under development anyway. I'm running VBR at quality 3 and get a "hissy-squelchy" background noise. This is fine, kinda, because the internal microphone in the laptop picks up hiss, the sound of the (actually very quiet) hard drive and generally speaking is of less than exemplary quality. To help

These are my results with the Speex AGC and denoiser: Over the past couple of months, I've spent probably 10-20 hours of conversation with Speex (wideband) doing AGC + denoiser on both ends of the call. I think it does an excellent job of removing most noise while keeping speech intact. Yes, it's possible to make some constant noise (like humming) and have that get removed, but

hi! I'm using the speex library in my RTP project (i'm using GNU ccRTP library). My program creates and destroys speex objects everytime a new RTP connection is made. But, when i test it with a client which create a connection to it one after another, the free memory decrease constantly to the point where the programs execution crash because not more free memory. It seems like the

Tom Harper wrote: > Hi All & Jean Marc, > > Once again I find myself delving into the pre-processing code to fiddle > with the VAD, AGC and denoising code. > > Where i am at is that I have implemented all of Steve Kann's mods, and > they are 90% of the way there in terms of working, except that I am still > having issues denoising open air mics. But that is

First, I'd just like to thank the Speex community, and Jean-Marc especially, for their great work. I'm developing a VoIP library (which uses IAX, the asterisk protocol) as the network protocol. I've been putting off integrating Speex for a while, as things have been working pretty well so far with GSM. (for those interested, the code is at iaxclient.sourceforge.net). However,

> 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

Jean-Marc Valin wrote: > The noise suppressor will only attempt to remove stationary noise, > such as thermal noise, fans, ... The AGC can indeed do strange > things in these cases, but it's been improved in svn (compared to > 1.2beta1). OK, then the problem is that I misunderstood the feature. I assumed that dynamic squelch was part of it, but it's really something more

Andras Kadinger wrote: > I have now found a testcase where SVN AGC starts out good, but then it > keeps increasing the gain but never decreasing it, and eventually goes > out through the roof, during the same single recording. I've been doing massive changes in the svn preprocessor, so I'm not too surprised I broke the AGC, because I haven't updated it. Did it also do that in

I'm having trouble with the preprocessor's noise reduction feature. The basic issue is that it simply doesn't work very well. With my laptop (whose microphone is otherwise quite capable) I routinely hear transient background noise, typing, and other "quiet" sounds leaking through to the speex stream. Even worse, the AGC feature is blowing these things up into just awful

I'm not sure if this is due to the same problem, but I get the same clipping behavior when trying to use VAD+denoiser on a particular mic+speaker setup that has significant background noise (enough to really foul up the codec should the denoiser be turned off.) Leaving VAD off and transmitting continuously seems to be the only way to avoid this clipping. Another person I regularly

> 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

I am trying to use the speex echo canceler and preprocessor with sipXtapi to develop a sip user agent. the sipXmedialib call flow graph uses 10ms frames and I am not sure what the implications are if I try to change this. The documentation seemed to indicate that a 20ms frame was recomended, but it didn't go into the consquences of using other frame sizes. We are going to use headsets

Jean-Marc Valin wrote: > Andy Ross wrote: > > Uh, production applications almost always require squelch, no? > > Some do, some don't. In general, distinguishing between a keyboard > and a speech transient is next to impossible based only on a few ms > of speech. That is true for distinguishing it by waveform, but not by amplitude. As I mentioned, these transients are

Jean-Marc Valin wrote: >If you set the denoiser to "on" and the VAD to "off", what difference >does it make in CPU time? > <p>Same program, running on Athlon XP 1700+: Test 1, using VAD, but AGC, denoise off: tevek@canarsie:~/work/hms/app_conference $ time ./vad_test /tmp/demo-instruct.sw 5 reading from /tmp/demo-instruct.sw, repeating 5 times read 537760

Hi, Arun and me are working on a echo-canceler module for pulseaudio build around speex. When using the preprocessor's echo filter, specifically in combination with denoise and AGC, it's unclear whether these have to run before or after the AEC. Right now we've got one instance of speex_preprocessor doing AGC and denoise running before the echo canceler; and a second for echo