similar to: Voice Activation Level (speex 1.1.11.1)

I implemented the calcPower(). It works perfectly. The example is given you in just about 6 hours. Cant paste the whole source here and need to meet someone now. Thanks all (particulary tom). I try to figure out whitch problem exists with the #define SPEEX_PREPROCESS_SET_PROB_START 14 theese days ----- Original Message ----- From: "¼Õ½Â¿ø" <ssw0725@ncsoft.net> To: "Tom

Hi...Tom, How to use the code you written? Can you show me some example? Thanks, -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On = Behalf Of Tom Grandgent Sent: Friday, March 03, 2006 12:57 AM To: Steve Kann; Lis Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Voice Activation Level (speex 1.1.11.1) Lis, I suggest you try tweaking Speex's

I done it speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_START, &g.s.VADstart); speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_CONTINUE, &g.s.VADcontin); /*char *lisDebugCh = (char*) malloc(20); int decimalFcvt, signFcvt; static int firstDebug = 1; lisDebugCh = gcvt(g.s.VADstart, 20, lisDebugCh); if(firstDebug == 1) { ::MessageBoxA(NULL,

Manisha, I'm still here. :-) Here's the function: // Returns the average power level in the given signal float getPower(signed short int *signal, int numSamples) { int i; float amp; float powerSum = 0.0f; for (i = 0; i < numSamples; i++) { amp = (float) abs(signal[i]); powerSum += amp * amp; } return powerSum / (32768.0f * 32768.0f *

Sorry. I forgotten the words volume or loudness. But it is know as microphone stroke too, i think. If something can tell me something about that procedure it would complete my pleasure. To bring back memories, i only wanted to know wheather i can change a variable that holds the sound intensity (loudness) needet to start "encoding >> sending" if the speex codec is in voice

What you want is simply a loudness threshold-based detector. It's not very complicated to do, but there's nothing in Speex that currently does that (not that I think it's really useful in practice). Jean-Marc Le jeudi 02 mars 2006 ? 04:26 +0100, Lis a ?crit : > Sorry. > > I forgotten the words volume or loudness. > But it is know as microphone stroke too, i think. >

I havent had found anything in the documentation about voice activation levels. Does i can change a variable to change the accuracy for activations? If not does the speex lib already implement a function for read out the sound level of a frame? Thanks for the advance. Lis (Louis Hoefler) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Jean-Marc Valin wrote: >> 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"

Steve, You're right. The AGC gain does not max out when using VAD (via the preprocessor). So instead of not transmitting when the AGC max gain is reached, I now do this instead: Start the call with VAD enabled and AGC disabled. When speech is detected, disable VAD (if 100% continuous transmission is desired for the call) and enable AGC. This seems to be working reliably so far. However,

OK, so the problem doesn't seem to be the VAD specifically. Can you tell me how much audio you had in the test? It may be that nothing's wrong and the code just isn't so fast that you can do 100 channels. Or maybe it just needs a bit of optimization... Jean-Marc Le mer 31/03/2004 à 10:03, Steve Kann a écrit : > Jean-Marc Valin wrote: > > >If you set the denoiser

> 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

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

Jean-Marc Valin wrote: >> 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

On 06/22/2011 09:30 AM, Steve Kann wrote: > Speaking of AEC (thought not quite on topic for this thread), > > Has anyone on this list played with the GIPS code that google just > open-sourced? It looks like their AEC also has code to handle > differential sample rates, though I haven't really evaluated it > thoroughly. > > There is really a lot of code in the drop ?

Hi people I'm new in Wine, so pleas, don't kill me... :) I tried to install MS Office. I found some stupid tutorial on net, whitch said me to remove rpcrt4.dll, how you can see here http://www.quicktweaks.com/2008/04/09/install-ms-office-2007-in-linux/ In step five I found, that link for new rpcrt4.dll is invalid ... so i didn't finish the tutorial. My Wine started to give me

On 08/06/07, Steve Kann <stevek@stevek.com> wrote: > > I'd look at the speech-to-text implementations for this -- I think CMU > Sphinx has done something like this. > Thanks. I had a look at their web pages, and the Sphinx software looks interesting, but I was unable to determine if there is a "hook" in their system to allow simple speech _detection_ rather than

Jean-Marc Valin wrote: >>In particular, (I'm not really sure, because I don't thorougly >>understand it yet) I don't think your jitterbuffer handles: >> >>DTX: discontinuous transmission. >> >> > >That is dealt with by the codec, at least for Speex. When it stops >receiving packets, it already knows whether it's in DTX/CNG mode.

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hello Jean-Marc: On 08/06/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > Either one. The question is: If we treat the software like a black > > box, and we feed in PCM audio, we get Speex encoded data out. Where is > > the information that indicates whether the encoded data contains > > speech or not? The API has a "get VAD status", but it