similar to: AEC in live performance

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

If the left and right channels are processed separately as you do, the parameter Mic-Number in speex_echo_state_init_mc should be 1, not 2 On Wed, Feb 10, 2010 at 4:00 AM, <speex-dev-request at xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

Fixes issue #95 on GitHub. --- src/opus_multistream_encoder.c | 57 ++++------------------------------ 1 file changed, 6 insertions(+), 51 deletions(-) diff --git a/src/opus_multistream_encoder.c b/src/opus_multistream_encoder.c index 6cc1f432..9cb9bf34 100644 --- a/src/opus_multistream_encoder.c +++ b/src/opus_multistream_encoder.c @@ -742,20 +742,9 @@ static void ambisonics_rate_allocation( {

Hi, I wanted to know how the speex echo canceller can be used for multiple channels audio. I am aware that support had been added for multiple speakers and multiple microphones. Is there any documentation on how speex_echo_state_init_mc can be used in conjunction with speex_channel_decorrelate? Thanks, Mustafa -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Can anybody show me how to enable stereo AEC, I tried with the following code and the result is bad, degraded output. Init: ec_state = speex_echo_state_init_mc(frame_size, aec_tail, 2, 2); speex_echo_ctl(ec_state, SPEEX_ECHO_SET_SAMPLING_RATE, &sample_rate); preprocess_state_left = speex_preprocess_state_init(frame_size, sample_rate); preprocess_state_right =

I'm looking for suggestions as to the best multimedia headsets for a call center environment. A few considerations: 1) USB headsets are preferable, because they don't require a soundcard. 2) Omnidirectional microphones are problematic, because they pick up too much background noise. Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

Hi, Two related news here. First, I've merged in support for stereo (or more channels) acoustic echo cancellation in mainline. The standard AEC remains the same, but I added a speex_echo_state_init_mc() call that takes the number of speakers and microphones. Now, one of the consequence of this new work is that by merging this work into the main tree, I had to do things in git that svn just

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

first: hi i'm wouter new to the forum, lon time linux user(still a noob), been using wine (for school issued software) for 2 years now in Ubuntu 64 bit(current Ubuntu 10.04 64bit wine-1.3.0(from the Ubuntu repo)) need help installing "windows 95 Audio Compression Codecs" for a school issued program named "the ultimate human body 2.0" for windows 95. (my school does not

ComRades: I am geeting the error message Error in ldots[[which(MET)]] : attempt to select less than one element when I try to fit the geostatistical model with the likfit() function of geoR. I have tried with old data for which likfit() successfully maximised the likelihood in previous versions of geoR, and yet the current version fails. I have tried in Windows Vista and Windows XP (I haven't

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

Hi everyone, I've just released Speex 1.2rc1. This adds support for acoustic echo cancellation with multiple microphones and multiple loudspeakers. It also adds an API to decorrelate loudspeaker signals to improve multi-channel performance. In the bugfix department, there are fixes for a few bugs in the echo canceller, jitter buffer and preprocessor. At this point, the API for 1.2 should be

Please see the code below. When I try to run the variogram - vg.deft<-variog(rd,uvec=10) I keep getting this error- variog: computing omnidirectional variogram Error in vector("double", length) : vector size specified is too large Also, when I try to define distance-based neighborhood - nb.tr=dist.neighbors(tr.locs,2) I get this error - Error in vector("double", length) :

Hello All! I calculate a variogram using the function variog (package geoR) afterwards I use variofit to fit a spherical model (see code below). Now I just changed the units of the variable (in this case MPa to kPa just a factor of 1000). If I do so, I get a different fit and therefore different ranges etc. Why? The semi-variance is of course 6 orders of magnitude higher but the values

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, Indeed I too have troubles implementing echo removal, I like ask kindly to Jean-Marc (or any other) if him can put a source code demo to show us how to use effectively echo removal API and parameters in real case scenario. A big thank you! Roberto -----Messaggio originale----- Da: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] Per conto di hs Inviato: marted? 11 ottobre

Hello, I am attempting to get the standard deviation in multiple distance bins in my spatial data. It appears as though the 'variog' command in the geoR package will do the trick, as one of the outputs from 'variog' is 'variog$sd', which, according to the manual, is the "standard deviation of the values in each bin". However, when I run this command, the

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

Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the