similar to: Stereo AEC

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 >

> No, the 1.2beta1 preprocessor works rather well. I have a different opinion, but glad you like it anyway :-) > In fact, I very much welcomed the change that the preprocessor in > 1.2beta1 now starts out with the assumption of there being no speech, > and only activating and adapting AGC when where there is new activity > detected above that. As a result, with 1.2beta1, there is

I am trying to compile speexclient.c. I downloaded all the files to ~/speex- 1.2beta1/include When I call ./compile.sh I get the following errors, speexclient.c: In function 'main': speexclient.c:164: error: 'SPEEX_PREPROCESS_SET_ECHO_STATE' undeclared (first use in this function) speexclient.c:164: error: (Each undeclared identifier is reported only once speexclient.c:164:

I am using 'portaudio' and 'speex' to implement a speech audio capture/playback. Preprocess works fine but not the echo canceller (associated with preprocessor). I have read 'testecho.c' but it didn't help me. What I am doing bad? This is my implementation: One thread to capture and playback, both are different functions called in different times directly from

Hi, I've set up speex AEC in our application. The echo's seem to be canceling, but the captured voice is distorted somehow. It sounds to me like low and/or high frequencies are removed. This happens even if I set my playback(echo) data to only zero's. Is this "normal" for the AEC? As far as I understand, if I send 0's as playback/echo data, the resulting data should be

Hello, all. 1) ACE does not work for me I am in a voip project using Speex, failed to have hte Speex ACE work. here is how I initialize it: /** * Configurations : * #define BITS_PER_SAMPLE (16) * #define SAMPLE_RATE (8000) * #define CHANNEL_NB (1) * #define DURATION (20) * SPEEX_MODEID_NB */ _eco_state = speex_echo_state_init(_encframe_size, 10*_encframe_size); speex_echo_ctl(_eco_state,

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

What is the current status of the residual echo suppressor in the preprocessor? I ask this because I have occasional problems with the AEC, in which it fails to cancel echo. I think the problems have to do with samples being dropped by the audio IO system, which is effectively impossible to detect and counteract reliably. A colleague recommended using an echo suppression technique (which he

Hi, We are using your speex codec in our VirtuvalClassRoom product. It's working good but we have problem using your echo canceller. Can you send me some code samples which use the echo cancelation? thanks suresh A Bring your gang together. Do your thing. Find your favourite Yahoo! group at http://in.promos.yahoo.com/groups/ -------------- next part -------------- An HTML

Hi All, I would like to communicate speech over a network compressed using speex. However, I do not want to communicate a whole Ogg-formatted file. I am interested in only the speech frames. I invoked: speexenc input-file-name - > raw-speech This, I am thinking gives the speech frames only. To play this back at the receiving end, do I need to format it into Ogg formatted file before I can

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, 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

HA! Just when I was getting ready to drop myself from this list, along comes a ham radio question! Shoving digital voice down a 100 Hz pipe would be mighty darned tough/impossible. N0YMV has a good suggestion. Try those WSJT modes. They use narrow bandwidth and are fun to play with. There are documented instances where moon-bounce QSOs have been completed with these modes. No, it's

Please, I noticed in the AEC module documentation that I can set the sampling rate and the frame size (in samples), but I found nothing regarding to the number of channels, no SET and no GET... So, I 'd like to know how I setup the channel's count on AEC.. if the buffer passed to the echo cancellation function should be mono or stereo... Another question: Is there a preferred

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0-rc1 DAHDI-Tools-v2.7.0-rc1 dahdi-linux-complete-2.7.0-rc1+2.7.0-rc1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In this release: *

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0-rc1 DAHDI-Tools-v2.7.0-rc1 dahdi-linux-complete-2.7.0-rc1+2.7.0-rc1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In this release: *

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0 DAHDI-Tools-v2.7.0 dahdi-linux-complete-2.7.0+2.7.0 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In this release: * Driving closer

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0 DAHDI-Tools-v2.7.0 dahdi-linux-complete-2.7.0+2.7.0 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In this release: * Driving closer

Hi, when I use preprocessor with AEC, VC++ alerts me about heap corruption. I have protected speex_echo_playback, speex_echo_capture and speex_preprocess_run with mutexes, to avoid echo_state being used at the same time, but it still happens. Any help about this ussue? Thank you.

>jesus: > Now I see that it happens when I specify a framesize in > speex_echo_state_init greater than the one used in > speex_preprocess_state_init. > > Next combination works: > codec framesize = 160 > preprocessor framesize = 160 > eco framesize = 160 > > > Next combination DOESNT work (produces heap corruption): > codec framesize = 160 >