similar to: Extracting Coefficients from the AEC

Hello, I am new to this forum and have done my best to search for my answer before emailing, but have been unsuccessful. I have an application where I would like to utilize the speex AEC library for canceling hybrid echo. I thought it would be useful to save the coefficients of the filter for reuse, especially since the characteristics of the echo path will not vary. The specific application for

Hi, But in fact, it really affects the voice quality. One of my tester says, "Is your mouth far way from the mic?" Could you explain why we should cut 200hz below? >The notch filter is specifically designed to cut below 200 Hz when >working in narrowband. In wideband, the cutoff is more around 50 Hz. The >reason is that in narrowband operation (irrespective of the

On 03/03/2010 10:22 PM, QianBin wrote: > Hi, > > But in fact, it really affects the voice quality. One of my tester says, "Is your mouth far way from the mic?" > Could you explain why we should cut 200hz below? > We already said it affects the quality when the voice is compressed. Are you asking why that should be? Even with a simple form of lightly lossy

In CentOS5 you were able to create a server section in /etc/gdm/custom.conf such as [server-Standard] name=Standard server command=/usr/bin/Xorg -br -audit 4 -s 15 chooser=false handled=true flexible=true priority=0 After this change, Xorg would run with the -br -audit 4 -s 15 options. Unfortunately in CentOS6 this is not the case. It completely ignores anything put into custom.conf as far

Dear R community, xyplot (lattice) has been great in displaying tons of data for my research. I have used the following two xyplot commands (with example dataframe) to create two separate postscript/pdf files with respect to the variable "acft" and subset "status": test.df [[alternative HTML version deleted]]

Hi Shridhar, Sample rate conversion is not enough to solve this problem. I have tried this method several months ago. The first step is to measure the difference between sample rate of capturing and rendering. Then resampling (by what you said "sinc interpolation") one signal to eliminate the difference. The frequency step in my experiment is less than 0.1Hz. I have tried speex AEC

On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and

> 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

> 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 ? No. The entire path from AEC to loudspeaker and from mic back to AEC must be free of any non-linearity, codec, drift, ... > 2) The audio is moved in 32ms (512 byte) chunks and the reading and > writing to the

Jean-Marc Valin <Jean-Marc.Valin <at> USherbrooke.ca> writes: > > Le mercredi 07 juin 2006 ? 10:03 +0000, shridhar desai a ?crit : > > > > hi all, > > i am using the Acoustic Echo Cancellation from "Speex 1.1.12 version" > > in my VOIP application. Is it that the frame length to be chosen > > should always be 20ms or can i have

Hello Jean-Marc Valin and all the Speex-dev team, Today I have a little question about patents and the AEC. I' am not a lawyer, but after a deep research in esp@cenet.com, I found that in August 2003, an AEC called "Double talk insensitive NLMS algorithm" (DNLMS) have been patented by Gary Q Jin, Gordon J Reesor, and others (US 6.608.897 B1). It uses a continuous adaptation step

Analog Devices sponsored a fixed point version for AEC which was posted early this year, but it seems from the source code that the fft routines are still in floating point. Is the port still not complete or am I missing something? Has anyone out there ported speex AEC on RISC architecture? Please let me know. Thanks in advance, -Deepa -------------- next part -------------- An HTML

I've looked at some partial examples and read in the pdf Manual that the Acoustic Echo Canceller can be used outside of the speex codec as a preprocessor. I currently have a java app that uses the ffmpeg api for live streaming. I'm exploring integrating echo cancellation & I'd like to use the speexdsp AEC by mapping a javacpp library to communicate between java & speex. Audio

AFAIK, that's a common point for all AECs. But some of them solve the problem by resampling on of the end to keep it in sync with the other. On Tue, Jul 7, 2009 at 5:14 PM, ggb<ggb at tid.es> wrote: > Thank you John. > > On 07/06/2009 11:03 PM, John Ridges wrote: > > ly synchronized, and therefore the clock drift adds a non-linear > factor to the audio path. The AEC

Hi, I've got an AEC Star T4110 connected with an RS232 port by a megatec cable. On the list of supported devices (http://www.networkupstools.org/stable-hcl.html) there is only one AEC device, so I tried genericups with upstype=21. The problem is, that the UPS (while being on-line) reports constantly being on battery. # upsc aec ups.status OB My question is: What is the recommended procedure

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

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