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 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:

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. >

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 *

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,

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

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

Ok, I figured it out. :) This seems to work: 1) Call speex_bits_read_from() once, specifying the location in memory of the compressed data, and the total length of that data. 2) Keep calling speex_decode() until speex_bits_remaining() returns 0. Then you don't have to keep track of the # of frames per packet, or the size of each compressed frame. It's done magically by the codec.

This is a really good point, and definitely a recurring theme on this mailing list. :) I wonder, what are some better options for handling this issue, other than to keep saying "just use 8/16/32kHz"? - Extend Speex to support other sample rates (seems unlikely..?) - Integrate a resampling algorithm into libspeex - Maintain a list of recommended resampling libraries that work well

Ohh... Nice! This is new in 1.0.1, isn't it? It doesn't seem to be included in the reference manual yet, though. Thanks! Tom <p>Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > OK, this is how it works: > > The encoder calls speex_encode any number of times and then calls > speex_bits_insert_terminator before sending the bits. > > The

Yes, you can use it independently by using speex_preprocess(). This function can do noise removal, AGC, and VAD. I've been using it to do all three and it usually works very well. There can be a train going by outside, producing lots of sound that makes it through the noise filter, and yet the VAD knows it's not speech. However, sometimes the VAD seems to get into a bad state and

Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase in audio quality and the bitrate is still very low, especially if you take advantage of Speex features such as VBR. 8kHz seems totally inappropriate to me for desktop streaming audio, let alone 8-bit samples. Or perhaps your recording equipment is an original Sound Blaster from 1989? (Even that could record at

Ok here is something I couldn't see on the description page. We need to convert to 8 bit audio right? I'm assuming narrow band is 8 KHz 8 bit. Wide band is 16 KHz 8 bit. UW band is 32 KHz 8 bit audio? George -----Original Message----- From: Tom Grandgent [mailto:tgrand@canvaslink.com] Sent: Tuesday, October 03, 2006 11:42 AM To: Bertie Coopersmith; George Ou Cc: speex-dev@xiph.org

Thanks all for your prompt reply. I agree with Alex. I think jitter buffer will have to drop packets if its input bitrate is higher than it is configured for. I never tried speex jitter buffer. What does it do when input bitrate is higher or lower ? As an alternate solution, I was thinking to play with directsound playrate [using SetFrequency()]. Has anyone used it to solve such problem? Any

I am a Java developer. I represent a small company writing a audio archiving application in Java using SPEEX files on Windows 2000/XP. We have been using the software found at the main SPEEX site and looking for API's we can use there. We are looking for a Java based SPEEX API that allows us the following playback functionality: - Open the SPEEX file from local file system or via a http

Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote: > > See the SPEEX_SET_VBR and SPEEX_SET_VBR_QUALITY options. But be aware > that unless you're aggregating many VoIP calls on the same link, there's > little use for VBR in VoIP. I don't agree with this. I've used Speex's VBR mode exclusively in my P2P VoIP software for the past 4 years and never

But that's what a jitter buffer will do, eventually - drop a packet. And it's better to drop a silence packet. Cheers, .a -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Wednesday, December 21, 2005 3:39 PM To: Alex Bakaev Cc: Shantanu Gramsci; speex-dev@xiph.org Subject: RE: [Speex-dev] Bitrate problem On Wed, 2005-12-21 at 15:29 -0800,

hi Tom, i downloaded your speex wrappers and they're really what i'm looking for! i really cannot worry about container support at the moment because i'm working with real-time audio, and hopefully toward VoIP when everything else is in order. im using waveform to capture data from the microphone. the raw PCM sample is stored in the WAVEHDR structure's member lpData, and this

I am a Java developer. I represent a small company writing a audio archiving application in Java using SPEEX files on Windows 2000/XP. We have been using the software found at the main SPEEX site and looking for API's we can use there. We are looking for a Java based SPEEX API that allows us the following playback functionality: - Open the SPEEX file from local file system or via a http