similar to: Speex 1.2rc1 is out, status update

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

Hi, I plan to use AEC for a live performance, storytelling for very young children (and their parents!) in a mongolian yourte . Actually the storyteller can make vocal loops, there is an omnidirectional microphone in the center of the yourte, 5 loudspeakers in a circle along the yourte's wall and Pure Data in a linux box. And now she wants to make vocal loops over music and loops over

Hi Steve, > I don't know if this has only recently been put on line, but I never > noticed it until today - > www.iwaenc.org/proceedings/*2008*/contents/papers/9044.pdf > > That paper is from people at MS describing, in some detail, what the > Windows kernel echo canceller does to handle synchronisation issues. It > tracks both time varying sample clock drift and hiccups

Hi Steve, Have you read this paper? (Heping Ding, David I. Havelock, Drift-Compensated Adaptive Filtering for Improving Speech Intelligibility in Cases with Asynchronous Inputs. EURASIP J. Adv. Sig. Proc. 2010:) Let me call is paper-Drift. It provided a method to evaluate Relative Sample Offset (RSO, d[i]) which is omitted in the microsoft paper (Challenges and Solutions for Designing Software

Dear all, I have just implemented a gstreamer plugin for the Speex AEC, but now I would like to extend it to a multichannel echo cancellation I have been searching documentation about that but I could not find it Where it is? If it doesn't exist? Could you add a short example showing how do you think it should work? or It's "just" necessary to: init the multichannel echo

First, I'd just like to thank the Speex community, and Jean-Marc especially, for their great work. I'm developing a VoIP library (which uses IAX, the asterisk protocol) as the network protocol. I've been putting off integrating Speex for a while, as things have been working pretty well so far with GSM. (for those interested, the code is at iaxclient.sourceforge.net). However,

Hi! I want to use speex for echo cancellation in my program, but I have bad results. I will explain what my program does. it is a client-server application. I run a server in room A and a client in room B. the client sends some voice to the server and the server plays it on loudspeakers. I run another server in room B and connects to it from room A using the same application that runs

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

Hi, I implemented a small patch that allows the internal convergence state of the echo canceller to be saved in a file for later use, especially after a process restart or machine reboot. This enables immediate echo cancellation the second time the AEC is run. Of course this works only if the acoustic environment of the device doesn't change and if the soundcard latency is constant. To use

Hi, A couple things you may want to check: - set sampling rate to 8 kHz (at least for now) - make sure the far end signal in the playback signal is always a bit in advance (never late) compared to the mic signal. - Set the tail length to something around 100 ms. Also, if you're using two different soundcards (as I understand) for the playback and the capture, you're *never* going to get

All, Wonder why all my posts has no line breaker when it displayed the forum... I used Hotmail(Windows Live Mail) with FireFox 2.0. I am posting this msg with IE 7.0 and see if it make any difference. Anybody experienced the same issue and how to workaround it? Thanks a lot! Regards, William From: espzzh@hotmail.comTo: jean-marc.valin@usherbrooke.caCC: speex-dev@xiph.orgSubject: RE:

> I'm wondering if there's a way to speed up the initial convergence > time. Maybe I'm completely off base here, but specifically I was > wondering if the code starts the search assuming the echo is near to > zero, and works longer to find the adapted point. If so, is there a way > to have it reverse the search, that is, start with the assumption that >

Hi, I'd like to try using a good quality microphone and a set of PC speakers (in the first instance) to create a powerful speakerphone; if I get that working, I'll probably try more elaborate audio equipment. For this to work, I'll need software acoustic echo cancellation, or the caller at the other end will constantly hear his/her voice echoing back. I gather Asterisk can do

This is only remotely related: just out of curiosity, does speex_preprocess work very well for you ? It only works to certain extend for me, and does not seem to completely remove noise from sound card. Even if you unplug the microphone, it still could not get rid of the noise. Any idea how it could be totally eliminated ? <p>Thanks <p><p>-----Original Message----- From: Tom

(from thread Re: [Speex-dev] Speex echo canceller on TI C55 DSP, but this is a more general topic) >> With these changes, the link was successful, using testecho.c with some >> modifications for the C55 environment. The code and data memory >> requirements were a lot more than I had hoped (>20kbytes of dynamic data >> memory for block size=128, tail length = 1024),

On 04/04/2011 06:58 PM, Simon Morlat wrote: > Hi, > > I implemented a small patch that allows the internal convergence state > of the echo canceller to be saved in a file for later use, especially > after a process restart or machine reboot. This enables immediate echo > cancellation the second time the AEC is run. > > Of course this works only if the acoustic environment

Hi everyone, I've just released Speex 1.1.9. The main change in this release is the echo canceller work sponsored by Tipic Inc (http://www.tipic.com/). It is now possible to do acoustic echo cancellation and obtain good attenuation after a short adaptation time. This has been tested at 8 kHz, but it should also work at 16 khz and above, so give it a try. There were also some fixes to the

>> Simply to say, in a quiet room, you can play a impulse signal and then find >> it's impulse response signal from the >> microphone. For example, if the delay between the impulse signal and its >> response signal range from 500 to >> 3000 cycles, you can buffer the far-end signal to 0-300 cycles and set the >> filter length to 4000. It is also called

I'm trying to convert my current headphones and microphone chat application to support loudspeakers and microphone, and so I thought I'd give speex_echo_cancel() a try. Since my users quite frequently have other sound-producing applications running on their computer (such as winamp), I sample 'wave' recording device of the soundcard in addition to the microphone. I then call