similar to: AEC: Tips on signal synchronization.

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

Hello, I am using the speex AEC in a real time application. I have experienced that when the mic and the speakers tracks are on sync or with a small delay the AEC works very well. I understood that when they are out of sync the AEC cannot works, so what the "user" should do is to focus in order to have the tracks on sync. Since I am working in an environment where it is not rare to

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

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?

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

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

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 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

On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: > The fact that the AEC takes a few seconds to converge is normal. The Do you think there might be a way to reduce this? > fact that it needs to completely re-converge in the middle of a call > probably indicates that something went "wrong" in the audio > capture/playback. For example, that could be an

Perhaps you could add a warm-start to the AEC, such that the parameters start near the correct values on all but the first use? Stuart On 05/25/2011 08:22 AM, Andras Kadinger wrote: > 2011.05.25. 4:52 keltez?ssel, Arun Raghavan ?rta: >> On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: >>> The fact that the AEC takes a few seconds to converge is normal. The >> Do

Hi, I am using the sim1000G R package to simulate data for case/control study. I can not figure out how to manipulate this code to be able to generate 10% or 50% causal SNPs in R. This is whole code provided as example on GitHub: library(sim1000G) vcf_file = "region-chr4-357-ANK2.vcf.gz" #nvariants = 442, ss=1000 vcf = readVCF( vcf_file, maxNumberOfVariants = 442 ,min_maf =

Check out the proposed prices when this is approved. BELL CANADA REPORT ON THE ECONOMIC EVALUATION FOR THE TARIFF REVISION OF Bell Canada's Access Services Tariff Item 315 ? Zero-Dialed Emergency Call Routing Service (0-ECRS) *2 March 2006 TABLE OF CONTENTS Page 1.0 GENERAL 3 1.1 Purpose of the Study 3 2.0 SERVICE DESCRIPTION 3 2.1 Service Characteristics 3 2.2

2011.05.25. 4:52 keltez?ssel, Arun Raghavan ?rta: > On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: >> The fact that the AEC takes a few seconds to converge is normal. The > Do you think there might be a way to reduce this? The shorter you make the tail length the faster it will adapt. But this will make it more important to reduce HW/SW latency so that you don't waste

Fully up-to-date CentOS 5.2 and Firefox browser. A previous thread had a problem with plugins disappearing. I've discovered, by accident, a causal relationship. Running a Thunderbird mail client (could happen with others? Haven't tested with Evolution yet.) I click a link that opens the browser. BANG! The plugins.dat is "emptied". Restored from my rsync backup, repeat, BANG!

> 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:

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

speex-devDear Jean-Marc Valin I got some problems with evaluating the AEC module of speex. I wrote a test main function and compiled it with the speex lib in VC6.0, it initialized the AEC state and called the AEC main function in the same way as what was done in testecho.c. The near-end input wave file was a simple delaying and adding version of the far-end input wave, eg. y(n) =

Ok, let me be more clear on this. AFAIK, Windows OS doesn't expose speaker input as other OS ( Linux, Mac OS...). That puts you in bad spot in using Speex AEC with windows. Only way to work is to use Soundcard with ASIO functionality which does give you speaker input. But then you would have to impose that requirement on all your users. I heard Speex AEC works great from developer who