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This works well. Thanks for the link. However I am curious about something. Speex requires a VAD for the VBR mode and the comfort noise preservation. Is the Speex portion for this different than the VAD result returned from the preprocessor call? Vas ________________________________ From: speex-dev-bounces at xiph.org [speex-dev-bounces at xiph.org] On Behalf Of Ken Smith [ken at alanta.com]

Hi, Just wanted to know, is speex still being developed? I was going through the manual and its dated December 2007 for speex version 1.2 beta 3. As far as I can see there doesnt seem to be any new versions since then, is there any development work on speex? regards, James -------------- next part -------------- An HTML attachment was scrubbed... URL:

From 1.2rc1 source code: preprocess.c: /* FIXME: This VAD is a kludge */ st->speech_prob = Pframe; if (st->vad_enabled) { if (st->speech_prob > st->speech_prob_start || (st->was_speech && st->speech_prob > st->speech_prob_continue)) { st->was_speech=1; return 1; } else { st->was_speech=0; return 0; } } else { return 1; } AND case

Alright noted, I changed me code so that the state is created in the constructor and destroyed in the destructor of the object. However I'm still getting the same issue although I'm sure that would have bit me sooner or later. The new code is as follows. virtual Enigma::u8* Encode(Enigma::u8* inputBuffer,size_t inputSize, size_t& outputSize) { short *in=(short*)inputBuffer;

Essentially, the thing that's been holding up 1.2 was the AEC and preprocessor. The codec itself is pretty stable and I do not really see more work happening on it. As for the AEC and preprocessor, these are really open-ended project and I lack sufficient time to take care of them. Jean-Marc On 10-11-12 03:27 AM, Ken Smith wrote: > Basically, Speex is the brain child of Jean-Marc

I'm only doing one frame using speex_encode_int greatly simplifies my code I'm not sure why the sample I was working off of was converting the shorts to floats then calling the other encode/decode methods. Based off of your suggestions I tried the following but I get the same result. virtual Enigma::u8* Encode(Enigma::u8* inputBuffer,size_t inputSize, size_t& outputSize) {

I'm completely new to speex and I'm having issues adding it to my application. I can pass raw PCM and that works fine so I know my transmission code is fine however when I try to encode/decode using speex I get noise almost like a whining/buzzing sound. I'm sure it's and issue in my code but I'm not sure where to start looking other then my gut tells me I'm not sizing or

The way I do this is to compute the number of frames (by dividing the size of my input by the number of bytes per frame and then calling speex_encode_int() that many times. Something like speex_bits_reset( &bits ); for( n = 0; n < num_frames; ++n ) { speex_encode_int( enc_state, (int16_t*)input_tail,

Just an update, and a follow-up question: I'm making progress on this issue, and will likely have something working very soon, now that I understand how the jspeex transcoding classes work. Unfortunately, I will need to make a couple small changes to the jspeex source code. My question is - is anybody using jspeex for streaming speex-encoded audio? It works great for static audio, but seems

Basically, Speex is the brain child of Jean-Marc Valin, and from what I can tell, he has since moved on to other projects, though he continues to monitor this list. So no, it's not really being developed anymore. And what's out there really is good enough for a great many applications. That said, there remains room for improvement, mostly in the pre-processor (the codec itself is very

For what it's worth, I've had significantly better luck with the WebRtc VAD. It's pretty good. http://code.google.com/p/webrtc/source/browse/#svn%2Ftrunk%2Fsrc%2Fcommon_audio%2Fvad%2Fmain%2Fsource In general, the WebRTC voice engine seems to be more sophisticated and mature than the Speex preprocessor (as opposed to the Speex codec, which is pretty good). Ken Smith Cell:

...in this case I have to leave in the model x2 and x3, which causes the increase in AIC. What's the solution? Can anybody help me? Cheers ________________________________________________________ &#8222;Olyan cikkeket akarunk, amelyek k?zelebb viszik az orsz?got ahhoz, hogy n?pbut?t?s ?s alantas ?szt?n?k helyett v?giggondolt gondolatok ir?ny?ts?k.&#8221; &#8211; komment.hu http://ad.adverticum.net/b/cl,1,6022,318025,391319/click.prm

Hello. I'm from Federal University of Rio Grande do Sul - Brazil and I'm trying to adapt speex_echo.h (speex-1.2beta3-win32) to our video-conference software code. We are working on amplitude values, but I realised that in order to AEC works, one has to convert amplitude to frequency and filter the echo frequencies. So, my question is: Do I have to manually convert amplitude samples to

I'm developing an application that have a video conference component. For that I need echo cancellation, and is looking around for algorithms/implementations of that, and the one in speex is an alternative. In the documentation for speex I find the following sentence however. "Using a different soundcard to do the capture and plaback will *not* work, regardless of what you may

At just a first quick glance, I'm pretty sure that you don't want to be creating and destroying the speex encoder/decoder with each frame. That's supposed to stay the same for more-or-less the entire session. Ken Smith Cell: 425-443-2359 Email: ken at alanta.com Blog: http://blog.wouldbetheologian.com/ On Wed, Nov 16, 2011 at 2:52 PM, Christopher Schaefer <disks86 at

Oh I overlooked that one.. Thanks Darren! Btw, I was wondering if anyone has tried using the echo canceller function of speex with other codec like g.711? On Tue, Jan 17, 2012 at 8:32 PM, Darren Longhorn < darren.longhorn at redembedded.com> wrote: > On 14/01/12 12:26, Christopher Adoremos wrote: > > What is the highest quality audio data rate and sample rate Speex can >

Hi,

I have a couple apps running on my machine; one takes a PCM audio stream, converts it to Speex, and sends it over the network using RTP. The other receives the RTP packets, and then converts the Speex data back to PCM. The PCM is then played out the audio system. I'm currently ending up w/ static. Anybody have any pointers as to how to debug the situation? Thanks. -- Jeff Ramin Software

Background: I'm working on a web conferencing application hosted in Silverlight, and we desperately need an echo canceller. A while back, someone made a Java port of Speex (JSpeex), which was later ported to C# (CSpeex). However, those are based on an older version of the Speex code, and while they seem to interoperate just fine with the latest official release, they don't have certain

Hello, In a way similar to speexclient voip example, I want to have a voip application where 3 or more users can interact. This creates the following problems "1. How to playback 2 or more audio streams such that they appear overlayed (parallel)". The audio streams are speex encoded. I have few approaches in mind. 1. Create multiple decoder states and multiple ALSA audio playback