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...ires 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] Sent: Monday, August 29, 2011 6:38 PM To: speex-dev at xiph.org Subject: Re: [Speex-dev] Speex VAD always returning 1 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%2F...

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

...utSize/2)](); short *out = new short[(outputSize/2)](); /*Decode the data*/ speex_decode(mState, &mBits, output); for (i=0;i<(inputSize/2);i++) { out[i]=output[i]; } return (Enigma::u8*)out; } On Wed, Nov 16, 2011 at 5:58 PM, Ken Smith <ken at alanta.com> wrote: > 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 >...

...myself, but it would be > great if someone else with the right background who wanted to make a > name for themselves in this field was able to pick up where Valin left > off. (To be clear: we're all very much in his debt.) > > Ken Smith > Cell: 425-443-2359 > Email: ken at alanta.com <mailto:ken at alanta.com> > Blog: http://blog.wouldbetheologian.com/ > > > On Thu, Nov 11, 2010 at 11:02 PM, James Mbuthia <jmmbuthia at gmail.com > <mailto:jmmbuthia at gmail.com>> wrote: > > Hi, > > Just wanted to know, is speex still bein...

...i<(inputSize/2);i++) >> ? ? ? ? ? ? ? ? ? { >> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? out[i]=output[i]; >> ? ? ? ? ? ? ? ? ? } >> >> ? ? ? ? ? ? ? ? ? return (Enigma::u8*)out; >> ? ? ? ? ? ? ? ?} >> >> On Wed, Nov 16, 2011 at 5:58 PM, Ken Smith <ken at alanta.com> wrote: >>> >>> 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. >>&gt...

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

...> > /*Decode the data*/ > speex_decode(mState, &mBits, output); > > for (i=0;i<(inputSize/2);i++) > { > out[i]=output[i]; > } > > return (Enigma::u8*)out; > } > > On Wed, Nov 16, 2011 at 5:58 PM, Ken Smith <ken at alanta.com> wrote: >> 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. >> >&...

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

...y qualified to take on these jobs myself, but it would be great if someone else with the right background who wanted to make a name for themselves in this field was able to pick up where Valin left off. (To be clear: we're all very much in his debt.) Ken Smith Cell: 425-443-2359 Email: ken at alanta.com Blog: http://blog.wouldbetheologian.com/ On Thu, Nov 11, 2010 at 11:02 PM, James Mbuthia <jmmbuthia at gmail.com> wrote: > 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....

...ode.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: 425-443-2359 Email: ken at alanta.com Blog: http://blog.wouldbetheologian.com/ On Mon, Aug 29, 2011 at 1:36 PM, Yanick Bourbeau <ybourbeau at mrgtech.ca>wrote: > From 1.2rc1 source code: > > preprocess.c: > > /* FIXME: This VAD is a kludge */ > st->speech_prob = Pframe; > if (st->vad_enabled) &g...

...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 gmail.com>wrote: > 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 > transmis...

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

...cles Valin has wrote or referenced (I'm a bit out of my depth in these), but so far I haven't managed to wrap my head around the Speex AEC internals. So any pointers or shortcuts, or troubleshooting hints, would be appreciated. Thanks in advance. Ken Smith Cell: 425-443-2359 Email: ken at alanta.com Blog: http://blog.wouldbetheologian.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20101008/d565018d/attachment.htm

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