similar to: speex preprocess redux

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

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

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

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

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

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:

Jean Marc, So assuming the quality setting was set as high as possible, and the audio stream was 8000 hz, 16 bit mono, I have two follow up questions: 1. Does the complexity setting override the quality setting, or does increasing the complexity only lower the rate at a given quality? Or would setting both higher increase the quality more? 2. What sort of impact would setting quality as high as

? is there any speex code in place that may help enable variable speed playback ? or perhaps another open source project has some code for this ? variable speed playback for voice messaging is an attractive feature, that would be well recieved for sure ! thx for any reply, Mark Tom Harper <tharper@sightspeed.com> wrote: At 12:35 PM 10/6/2004, Matthias Granberry wrote: >There is

At 12:35 PM 10/6/2004, Matthias Granberry wrote: >There is some SSE assembly >language, but it's in GCC/AT&T syntax rather than the windows-standard >Intel syntax, so you might have to do some of your own translation to >something your compiler understands. We submitted an intel patch a ways back- maybe it is in the archives somewhere... Tom -- Tom Harper -

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 & list, I was going to start fiddling with using the ability to pass NULL to speex decode and using that to return an interpolated frame. I am curious what the intended use of this feature was- I.e. if there is a packet loss situation is it preferable to keep interpolating until new data shows up, or to cut your losses after a certain number of frames? I suppose some of this

Hi Jean Marc, I think there is just a confusion over terminology going on here- I agree that support for 3dnow base version may not necessarily be relevant; However, even though 3dNow extended is a bastardized version of SSE, it still supports the same instructions, and that is what is important- I don't think we intend to add any AMD specfic code. The real issue is cross CPU SSE support,

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! 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

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

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Jean Marc, Ok so I tested with the new code, same result- frame size= 160, filter length=160 ms/1280 samples, 8000 Hz diverged after 9564 calls/packets or 191 seconds. My system stays relatively in synch but is not perfect- difficult to measure if there is any clock drift, but it probably is. In order to make it break faster I use an open air usb microphone that is part of a logitech notebook

Jean Marc, I ran across the silence issue that results from the echo canceller producing "bad" floating point data. Unfortunately this hasn't been done in a way i can reproduce, but I found that calling reset didn't fix the silence, unless I also reset: st->PHI[i] st->Eh[i] st->Yh[i] So it may be good to add these to the reset function. I was using these in

Not a big deal, but the "<http://xiph.org/svn.html>Subversion Access" link on http://www.speex.org/download.html page should probably point to: http://www.xiph.org/svn/ rather than: http://xiph.org/svn.html Tom ______________________________________________ Tom Harper Lead Software Engineer SightSpeed - <http://www.sightspeed.com/>http://www.sightspeed.com/ 918 Parker

Congrats Jean Marc, Just heard you got a new google assistant for the Ghost project! Tom ______________________________________________ Tom Harper Lead Software Engineer SightSpeed - <http://www.sightspeed.com/>http://www.sightspeed.com/ 918 Parker St, Suite A14 Berkeley, CA 94710 Email: tharper@sightspeed.com Phone: 510-665-2920 Fax: 510-649-9569 My SightSpeed Video Link: