similar to: ERLE Measurement

Hi We are working on a speaker phone system using PJSIP and Speex Speech processing API on an ARM platform. Currently we have spent about a month on getting the AEC to work properly and we have worked through the most common causes of problems (such as clock drift, synchronization problems and non-linearity's in echo path). Now we achieve ERLE of about 17 dB which tells me that the AEC is

I wasn't implying that anyone do anything about it, just that's it a real problem. Unfortunately, most of the crappy sound cards are the ones that ship with your typical PC, so it's just something that people should be aware of. The solution is pretty straightforward -- just resample the audio data in real time using a reference clock. -----Original Message----- From: Jean-Marc

Thanks Jean, I missed that detail. I have a few more compile errors if you could help me with it => In function speex_echo_state_init in mdf.c I am getting error: "undefined reference to exp". I don't see a fixed point implementation of "exp". Can you point me to the correct files. I am using the following files from speex 1.2 beta1 => mdf.c,

> 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:

When I ran test 4 as originally described there is substantial echo cancellation (but not as good as when the files are perfectly aligned). When I invert the inputs, there is no noticeable cancellation. I'm using testecho with the preprocess line commented out. Preprocess seems to work very well at cleaning up the residual echo when mdf does its job, so I'm just focusing on testing mdf.

This kind of behaviour is odd. One of the reason could be the fact that you're using a really long impulse response. Try syncing your signals and making the tail length more in the order of 100 ms to 300 ms. Jean-Marc Le dimanche 06 novembre 2005 ? 21:25 -0800, Jason Harper a ?crit : > Thanks for alerting me to the new changes. I just > tried the latest code from SVN, but

Are you sure you're not just inverting the two inputs? Jean-Marc On Wed, 2005-11-09 at 22:16 -0800, Jason Harper wrote: > I ran some further tests on mdf and here are the > results: > 1. reduced tail length to 100ms, aligned mic and > speaker signals to within 10ms - almost no echo > attenuation > 2. aligned mic and speaker signals to within 5 samples > - still almost

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 somewhere in some of the OS resampling algorithms, but I've seen that on many machines.

Hi Jean-Marc, Thanks for the suggesting. I tried the test wtih my recording file offline through the testecho. I got the same result(no prepocessor echo suppression) as my recording file for that live call. Basically I don't see the difference between input_frame and output_frame. As I mentioned in my original post, the echo return loss is about 20db. I guess the echo cancelleris not really

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 >

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

OK, here's what happens. There is indeed a small difference between beta3 and rc1, but the fundamental problem isn't there. I've attached plots of the speaker signal (blue) alongside the mic signal (green). You can see the delay is in the order of 1000 samples. That's way too much to do anything useful because the tail doesn't even "see" the echo. You need to reduce

When you say inverted, do you mean a) input and output exchanged with each other, or b) input or output having changed sign numerically (e.g. multiplied by -1)? Andras Jean-Marc Valin wrote: > Make sure that: > > 1) You're using svn head, which has many improvements > 2) The AEC inputs aren't inverted > 3) The delay between the near end and far end signals isn't too

All, Wonder why all my posts has no line breaker when it displayed the forum... I used Hotmail(Windows Live Mail) with FireFox 2.0. I am posting this msg with IE 7.0 and see if it make any difference. Anybody experienced the same issue and how to workaround it? Thanks a lot! Regards, William From: espzzh@hotmail.comTo: jean-marc.valin@usherbrooke.caCC: speex-dev@xiph.orgSubject: RE:

On 04/14/2011 07:26 PM, LiMaoquan2000 wrote: > 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,

It seems like the firefox 2.0 not quite works with hotmail... I am reposting my message again with IE for easy readability. Sorry about mess. Hi Jean-Marc, Thanks for the suggesting. I tried the test wtih my recording file offline through the testecho. I got the same result(no prepocessor echo suppression) as my recording file for that live call. Basically I don't see the difference

I'm pretty much sure of it. When I test inverting the inputs, my output is pretty much the same as my speaker signal. Whereas the way that I normally test the output is my mic signal with very little attenuation. If you are interested I can send my test files; they are about 94KB each. -Jason --- Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Are you sure you're

Hi All We are working on an ARM-based speaker phone application where the speaker and microphone is placed roughly 8 cm from each other (similar to this one: http://www.voipsupply.com/images/CHAT50SPLASH.jpg). We are using Speex AEC and preprocessor to handle the acoustic echo. The AEC typically contribute with ERLE of 15-20 dB which should be as expected from the algorithm. Additionally we need

Johan Nilsson a ?crit : > We are using Speex AEC and preprocessor to handle the acoustic echo. The > AEC typically contribute with ERLE of 15-20 dB which should be as > expected from the algorithm. Additionally we need about 30 dB echo > suppression which we hope to get from the preprocessor. So far it seems like you're doing things right. > However it has shown to be very

Make sure that: 1) You're using svn head, which has many improvements 2) The AEC inputs aren't inverted 3) The delay between the near end and far end signals isn't too big (if you don't know what I'm talking about, RTFM) 4) The capture and playback are done on the same device (synced clock) Jean-Marc gorun.butail@wipro.com a ?crit : > Hi all, > > > > I