similar to: More Speex (pre-processing) Options

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

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

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

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

Steve, The main problem I am having with the system is clipping off the start of someone's speech when they first start talking- the ends of the sentences seem to be handled properly. I am wondering whether this is the fault of the audio playback system or whether this is a speex issue- I also get the musical artifacts problem with the denoiser. This seems to be more of a problem on open

Jean Marc, Thanks for the advice. The estimate update may come in handy. I have had some time to play with the adaptation time. I tried making the adaptation time 80, 160 and 320. It seemed like the smaller value was actually the best in my initial testing, but I need to test this against a more noisy setup than I have here. I am guessing that for vad a shorter adaptation time is probably

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

> 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

This works really well for white noise reduction. However what I've noticed was the amplitudes of normal speech samples also get reduced. Is this something by design, or is there a way to automatically recover the original speech sample volumes ? <p>Thanks. <p>Tongbiao <p>-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@hermes.usherb.ca] Sent:

Hi, 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 very well- great job! I

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

Bjoern, As far as I understand it, AGC keeps the volume level of the speech elements (i.e. gain) stable no matter how quiet or loud the input becomes- Tom At 10:53 AM 3/28/2004, Bjoern Rasmussen wrote: >Hello > >The st->zeta pointer isn't freed in the speex_preprocess_state_destroy() >function of the preprocess.c file (alloced in line 167). It's in Speex >1.1.4 by

I am using the denoiser and AGC, but the problem persists if I turn them off (stop calling speex_preprocess() altogether). Also, I have reproduced it on both a noisy system and a quiet system that hardly needs the denoiser. I know what you mean though, I have heard similar sounds when pushing the denoiser too hard. For example, doing 100x AGC, or trying to use an extension cord with my

> 1) AGC: This seems to work pretty well in all cases. I had previously > hacked-in the "compander" filter from sox for a similar effect. What > I've noticed is that speex_preprocess's AGC has no "knobs", and it > seems to use an attack/decay that is a lot faster than what I had > chosen from the sox compander, but it works pretty well nonetheless.

That's pretty bad. Both DirectSoundCapture and WinMM are capable of recording at 16kHz. I don't know why OpenAL would be incapable of handling it. It's not like it's at all rare or new. I would try 16000 and see if it works. Maybe the docs are wrong? Note that one option to retain high quality is to capture at a higher rate and then downsample using a resampling

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,

Hi All I am using speex in one of my WinCE project. I have been trying to use speex to perform denoise on the captured audio packet and echo cancellation. Following behavior I have observed while using various options. I would really appreciate if you could help me with the issues I am facing. 1. Denoise: I have written the following to code to perform denoise. //To initialize speex

hi there, I am new to mailing list so excuse me if I don't obey to the 'netiquette'. i am writing voice chat and speex is in the root of it. i write it in Java and use JNI to link with 'C'-based Speex 1.2beta. [I know of JSpeex but there are not implemented some features] recently i decided to use DTX feature of speex as well. the code follows. The problem is that no matter

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

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