similar to: AEC gets worse as sample rate increases

Try using SPEEX_ECHO_SET_SAMPLING_RATE to specify your sampling rate. Also, don't forget that the tail need to be longer (proportional to the sampling rate). Last thing, if you use resampling, make sure you use a decent resampler (the Speex one is fine) because otherwise, any aliasing left will not be cancelled. Jean-Marc Mihai Balea a ?crit : > Hi all, > > I am attempting to test

> Today's Topics: > > 1. AEC with WB mode (Jean-Christophe.Berge@etu.enseeiht.fr) > 2. Multiple frame encode and decode (Reza Fatahillah) > 3. cant link speex_echo.h (jesus) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 20 Sep 2006 08:46:03 +0200 (CEST) > From:

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 =

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, We are experiencing a few problems with Speex AEC. We are using it to process audio data on a real time stream over IP with Speex codec (frame size = 320). We initialize the echo state like this : SpeexEchoState * echo_state; echo_state = speex_echo_state_init(320,512); int sr = 16000; speex_echo_ctl(echo_state, SPEEX_ECHO_SET_SAMPLING_RATE,&sr); We are not really sure about

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,

On 06/22/2011 09:30 AM, Steve Kann wrote: > Speaking of AEC (thought not quite on topic for this thread), > > Has anyone on this list played with the GIPS code that google just > open-sourced? It looks like their AEC also has code to handle > differential sample rates, though I haven't really evaluated it > thoroughly. > > There is really a lot of code in the drop ?

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

Hi Aymeric, > 2 questions: I count 4 :-) > * Do you "feel" that the current svn is enough stable Yes. I'm planning on releasing it as 1.2beta3 this week. Even in normal time, I try to keep svn in good shape so it is almost always better than the last release. And even when it breaks, Speex bugs tend to be very obvious and easy to reproduce (no segfaults that happen once a

Hi Guys, I am trying to incorporate the speex AEC into my RTP stack. Trying to diagnose why the echo is not being cancelled I enabled DUMP_ECHO_CANCEL_DATA as per the docs. The problem is that the dump files the AEC generates are of different size. As I understand this and looking at echo_diagnostic.m, the files are supposed to be of same size. Can someone give me a hint as to why this could

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

I understand that the capture/playback signals need to be sync'd for an AEC to adapt. I'm a little bit confused on the requirements of synchronous sampling between the near end (mic/speaker) and the far end (phone line). I have an embedded DSP system with mic and speaker getting 1msec packets containing 8 samples. We can watch the DSP and ISDN clock frames drift and every few minutes we

2006/11/2, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>: > > Today I tried speex echo cancellation on my VoIP application running > > on a windows mobile based PDA (OMAP1510 168MHz). The result is > > disappointing: it takes 150-250 ms to do echo cancellation (by calling > > speex_echo_capture() ) for a 20 ms frame! > > Are you by any chance using an insane

Hi, I'm trying to use Speex Echo Canceller for mobile device. The AEC works well so far. However, I want to know how to remove reverberation significantly. I noticed it's not implemented yet in Speex AEC. Are you going to implement it in next version? Could you provide some ideas how to handle reverberation on current version? Thanks in advance. Sincerely, Dennis

Hi All, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set

The problem with different sound cards is that their clocks are not usually synchronized, and therefore the clock drift adds a non-linear factor to the audio path. The AEC can only cancel linear changes to the audio path, and so the AEC never converges.One solution is to measure the clock drift and resample either the input or output signal so that they *are* synchronized, and then the AEC

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Hi,all I have tested AEC on files, it works well.I have some files,one is echo file, others are echo-added files(an origin file adding echo at different delay,such as 20ms,40ms...120ms,140ms).AEC do wonderfully on those files except echo added at 140ms-delay. But ,when i use AEC in my voip project, it does feebly. Who can give me some hints why caused this.How long can sound be picked up by

Hi, I couldn't find a discussion that specifically addresses this, so here it is. I'm using Speex AEC in my mobile VoIP application to cancel speaker echo. The used version is 1.2rc1 from the website, and I'm compiling with fixed-point. On most occasions, the AEC works very well and cancels most of the echo (combined with the preprocessor). On some devices, where the microphone signal

Hi, I am looking for AEC software which can be run on the server side. This means there will be a fairly constant 600ms or so gap between sending out an audio frame and getting it back with echo. Could Speex AEC be configured to handle these conditions? If so, how good can I expect it to be? Thanks --------------------------------- Yahoo! Mail is the world's