similar to: AEC: Tips on signal synchronization.

The problem is that now I have two files very similar (one for mic, the other for speaker). If I play them together I can hardly say that they are two different file for most of the file length. When I try to cancel the echo, I obtain as output almost the same input. The update_foreground variable is always 0. I don't get the conditions which must be fulfilled in order to have

I used to run in the same problem. A difference higher than 3Hz in sampling rate between mic and speaker data can cause a totally failure of AEC. However, I have not seen a proper solution to this problem in real time. If you roughly know the difference of sampling rate between the speaker and the mic file, you may try to resample each frame of mic file according to it. Otherwise, or if the

2010/5/11 Elston Sa <jose at rebaca.com>: > Is there an API in speex or have you implemented your own? I don't know if there is an API in speex for this, probably not. There is a test for AEC which does something very similar. I implemented it starting from a FFT library. It is quite easy to do. > > I am currently trying to find one. But if you could point me to one it will

I usually use the cross-correlation between the speaker signal and the mic signal. It works pretty well. The problem is the you need enough data in order to get a good S/N. I compute the cross-correlation between 1.5 second data of each signals. In this way the application detection of any anomalies is quite slow. I am looking for something faster, but I fear it will be something very application

Maybe you shifted the signal too much. If the mic signal is just one single sample *ahead* of the playback signal, then there's no cancellation possible (because the system is not causal). Be careful with that. Otherwise, if could simply be dropped frames. Jean-Marc On 2009-12-30 07:06, Marco Pierleoni wrote: > The problem is that now I have two files very similar (one for mic, the

I tried cross-correlation. It does locate the sync point and it's almost perfect. I guess now I have to make it more perfect. Thanks for the pointers. On the side note: I have a test environment. Where I created the stream such a way that I have one "captured stream" purely with echo (i.e. no one else is speaking) and one stream with echo + someone speaking. I confirmed that I

Is there an API in speex or have you implemented your own? I am currently trying to find one. But if you could point me to one it will reduce my search time. :) Many Thanks, Elston -----Original Message----- From: Marco Pierleoni [mailto:pierleoni.m at gmail.com] Sent: Tuesday, May 11, 2010 5:55 PM To: Elston Sa Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] AEC - Echo is cancelled

1) Everytime a participant speaks there is a echo for a short duration (maybe a word or two) but as the participant continues to speak without a any break the echo is 95% cancelled (i.e there is a feeble echo still present if observed very carefully). 2) The moment the participant stops / pauses speaking and start talking again, scenario 1 is repeated as if the echo state has been re-initialized

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

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 would like to have some more information about the parameter of speex_echo_state_init. I have a clear idea of the importance of the filter_length in the echo cancellation, what it is more obscure for me is the role of frame_size and how its value affect the algorithm. Let use, for example, frame_size = 400, would I obtain the same result of using frame_size = 200 but with half method call?

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

Marco, Thank you for your response. Is there a way I can determine if the synchronization is perfect or not? If not by how much it is off. - Elston -----Original Message----- From: Marco Pierleoni [mailto:pierleoni.m at gmail.com] Sent: Monday, May 10, 2010 3:46 PM To: Elston Sa Subject: Re: [Speex-dev] AEC - Echo is cancelled however..... Hi Helton, it is now several months that I am using

On 2010-03-16 14:22, Josh Gargus wrote: >> If more than one speaker receives the *same* signal, it doesn't >> matter the number of speakers. It only gets tricky when the >> speakers are playing slightly different signals (e.g. from a stereo >> song). > > Does "tricky" mean that the Speex AEC won't handle such situations > well? Or just that you

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it. It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is

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

Le vendredi 11 novembre 2005 ? 01:21 -0800, Duane Storey a ?crit : > This is a very real problem though.. I've encountered many sound cards that > use different clocks for input and output (even on the same card!) Also, if > you open up a sound device on windows at 8kHz, the microphone is often > around 8100Hz, while the output is 8000Hz.. I'm not sure if there's a bug >

Greetings everyone, I was about to compare AEC performance between 1.1.12 and 1.2beta1 when I noticed something. If I configure (and compile) speex-1.1.12 with ./configure --enable-shared=no --enable-static=yes it compiles and works as expected: I can run a mic and speaker signal through testecho, it runs in a reasonable amount of time (about 23 secs for 3 minutes of audio) and I get back