similar to: speex preprocess redux

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

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

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 the way. I'm trying to make the denoiser work with my application and has got reasonable noise reduction after applying the denoiser. I, however, haven't been able to find any information of what the purpose of

Hi Jean Marc & List, So I have been fiddling with the denoiser (again). While poking around I noticed that nb_preprocess is basically a counter that is mod-ded with 100 (the default), which causes Smin[] to be re- seeded with the value in Stmp[] (min of the previous adaptation period). Smin[] is then used to update the noise probability, which is (probably) less likely when adaptation

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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: > 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 don't seem to get that; I suppose

Jean-Marc, So I saw in the latest code that the vad in the preprocessor is gone/going to be re-written. Is there a plan as far as this goes? Just wondering as the current one seems to work pretty well. Thanks! Tom ______________________________________________ Tom Harper Lead Software Engineer SightSpeed - <http://www.sightspeed.com/>http://www.sightspeed.com/ 918 Parker St, Suite A14

Hello, I'm using Speex1.1.11.1 source code. I enabled dereverb by speex_preprocess_ctl(). I try different dereverb_level and dereverb_decay values and it seems to not work. I generate some test files in CoolEdit using reverb or echo and then preprocess it by speex_preprocess(preprocess, input, NULL); and output is the same as input:(. Denoising and VAD works very good. I tried dereverb with

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

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

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,

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

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

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

Jean Marc, I have been fiddling around with preprocess to try to get it sounding less metallic/underwater- so based on some things said recently, I have been first fiddling with mean_post and mean_prior to try to get them to do change the update rate, so that update_noise gets called at the right times. Haven't made it too deep into the code, but I have a few questions in speex_preprocess:

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

> 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

> 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