similar to: AEC in speex

Hi, Im trying to complete a list of jobs using SQL Querries and some "if else" commands but im stucked in some steps. Could any of you give me some help? -in COLUMN1 change the format 20JAN2000:00:00:00 to 20JAN2000 and exclude every row that date is different from 20. -extract the first character in COLUMN2 and creat COLUMN3 with that value ex: COL2,COL3 135,1 461,4 247,2 -in every

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 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

On 04/04/2011 06:58 PM, Simon Morlat wrote: > 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

Dear all: I need the help desparately. The code is attached below. If you guys don't mind take a look at the code below and see how to fit speex's AEC into it. Help me look at the #defines, and give me some suggestions on the AEC parameters, I totally have no idea about them. Feel free to do anything with the code, if it is by any chance valuable. Any ideas or suggestions or sharing

> On Jul 9, 2015, at 3:33 PM, Swaroop Sridhar <Swaroop.Sridhar at microsoft.com> wrote: > > Regarding Call-site size specification: > > CoreCLR (https://github.com/dotnet/coreclr <https://github.com/dotnet/coreclr>) requires the size of the Call-instruction to be reported in the GCInfo encoding. > > The runtime performs querries for StackMap records using

> 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, 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

The AEC is not designed to cancel sinusoids. Try with speech. Jean-Marc Johnson Peng a ?crit : > 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. >