similar to: Fw: Voice Activation Level (speex 1.1.11.1)

Lis, I suggest you try tweaking Speex's VAD probabilities as Steve suggested. But consider a simple threshold-based approach as a backup option. Personally, I struggled with Speex's VAD algorithms (both encoder and preprocessor) for a long time, tweaked the probabilities, wrote special case code to work around the mistakes, and was still never satisfied with the results. In times

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:

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

Hi all and Jean-Marc, Firstly thanks a lot to you for helps, at last I accomplished to use speex-1.2beta1 in my speech application. I try to understand speex source code and improve speech quality. The problem in my application (like many people in the list says) is noise. So I've added VAD in preprocessor as following. VAD in preprocessor is very good, there is not noise outside speech. I

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

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"

Jean-Marc Valin wrote: >>Reverberation suppression? >> >> > >Basically, it means that if you are in a room with lots of echo (long >decay), I can reduce it a bit. > > > >>I guess this would help reduce local source echoes? I've never >>_noticed_ that to be a problem in my use, but I would imagine that >>using a notebook's

> how about tcp? > in tcp you write a packet that got a possible length. > you send one packet after another, whitch stamp is incrementet by one > and if your incoming packet is gone in other steps than 1, the client has > to resend it. > Let me think some days about it and i will get another system. > Time is relative. > > Hm, you send a packet that needs to be in a

Hi all and Jean-Marc, Firstly thanks a lot to you for helps, at last I accomplished to use speex-1.2beta1 in my speech application. I try to understand speex source code and improve speech quality. The problem in my application (like many people in the list says) is noise. So I've added VAD in preprocessor as following: int param_enabled = 1; int param_disabled = 0; int sampling_rate

Hello I apologise for the length of this entry but please bear with me. In short: I need a way of subsampling communities from all possible communities of n taxa taken 1:n at a time without having to calculate all possible combinations (because this gives me a memory error - using combn() or expand.grid() at least). Does anyone know of a function? Or can you help me edit the combn or

David Brazier wrote: > Hi > > The latest SVN trunk for speex has changed the SpeexPreprocessState to > an opaque structure, for jolly good software engineering reasons. > However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) > relies on some members of this. It uses speech_prob to detect when > there is enough speech to consider AAGC and then loudness2 to

Hello, I'm having a problem using Speex - I'm getting white noise/static from the following process: (Load File) -> (Encode) -> (Decode) -> (Play) I know there's no issue with the loading or playing because (Load File)-> (Play) plays the file properly. I believe the problem lies with the speex_bits_write and speex_bits_read_from functions, because if I change my encode

When using the abx utility that is in LAME's misc dir, trying it on an ogg (with ogg123) gives the following output: Standard Open Source Bug detected, try murksaround ... (ogg123 is decoding to stdout) IIRC this also happened with older mpg123's but no longer with the latest release. Latest ogg123 from CVS still gives it though. Someone stole code but forgot to check for fixes? :)

I checked it against the latest code in the git repository and it wasn't there. Mihai PS: if JM wants a @hates.ms address, I could prolly hook him up. Especially if he throws in some VAD code that's not "a hack" :) On Nov 5, 2007, at 2:08 PM, Tom Grandgent wrote: > Did you check it against the trunk in SVN? > > If it's not applied, and you can hook Jean-Marc up