similar to: Extracting Coefficients from the AEC

Steve U., Thanks for the quick response. I will look into OSLEC for the Hybrid case. With regards to using the last known good set of coefficients for re-use, I am assuming that any changes in the echo path would be minimal to none (in the hybrid case of course) and therefore if my system was reset, only a small amount of convergence would be needed if at all. Of course I am basing my

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

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

Hi Jean-Marc, I tried with both testecho and my test program, and for some reason it just doesn't cancel any echoes with the 1.2-rc1. The testecho from beta3 binaries works fine, and also if I replaced mdf.c in 1.2-rc1 with mdf.c from beta3 and use my test program, it will work again. This happens for both 8KHz and 16KHz. Any ideas? I could upload the test samples and results if needed.

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

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

On Sat, Aug 9, 2008 at 12:59 PM, Jean-Marc Valin < jean-marc.valin at usherbrooke.ca> wrote: > Hi Benny, > > Can you send me your pair of testecho input files that work well with > beta3 and not with rc1? I'll have a look. > > Thanks for the help. The files are on their way now, the upload will take few more minutes to complete. In the mean time let me explain more

Hi, I've set up speex AEC in our application. The echo's seem to be canceling, but the captured voice is distorted somehow. It sounds to me like low and/or high frequencies are removed. This happens even if I set my playback(echo) data to only zero's. Is this "normal" for the AEC? As far as I understand, if I send 0's as playback/echo data, the resulting data should be

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, the differences range from 0.5Hz to more than 50Hz when sample rate is set

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]]

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

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

Hi all, LLVM needs a solution to the long-standing problem that the IR is unable to express certain semantics expected by high-level programming languages that target GPUs. Solving this issue is necessary both for upstream use of LLVM as a compiler backend for GPUs and for correctly supporting LLVM IR <-> SPIR-V roundtrip translation. It may also be useful for compilers targeting

Hello. I've just reinstalled a server that was with RH9 and now it is with Fedora Core 4. I'm using the nut package that comes with the distro, version 2.0.2 to check through serial port a MGE Galaxy PW (comm card 66060). Everything goes fine except for a little thing, when checking the UPS with '#upsc ups@localhost' I receive a line like this: 'ups.status: OL OFF' It

> 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