similar to: Reducing volume problem after aec and preprocessing

Hi: By 'no success', I mean result gets worse in terms that, there comes in a shrill noise with high gain and also *no* attenuation of the echo. I am trying to validate the DC offset part. But can you please feed some light on how does the DC offset matter for AEC. As I have already mentioned, I tried adding silence (from 2 ms to 10 ms) in the start of the reference signal. This is for

Hi: I have been trying to use speex for AEC. I have recorded two voice samples separately using audacity using the same soundcard. I reduced the gain of one of the samples and mix this signal with the other voice sample (saved as a raw sample, with frame size 64 and sampling rate of 8 KHz. I use the mixed one as a ref signal and the other unaltered signal as the echo. Am trying to run the

Thanks all for your help. I fear text mining is an abstract little corner of "R". I have imported 3228 text (.txt) files, each a news story, into R using [tm]: textd <- Corpus(DirSource("other/docs"), readerControl = list(reader =readPlain)) I can pre-process each individual document using tolower(textd[[1]]) however, when I try to run tmTolower() I get a no such command

I wrote a 5 line hack that you can drop into your initializers directory to enable preprocessing in Erubis. This will run code in < %=== %> brackets at compile time making views considerably faster for rails helpers that only need to be evaluated once instead of on each render (the Erubis docs mention 20-40% speed increase but obviously it depends on the amount you precompile). For example:

(Rails3 latest git master) Subject says it. I''m trying to figure out how to use erubis features described here: http://www.kuwata-lab.com/erubis/users-guide.05.html#rails, especially preprocessing. My question is, where do I place the Erubis::Helpers::RailsHelper.preprocessing=true Not in application.rb, at that point the object does not exist yet it seems. I wonder why I

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

tree: https://git.kernel.org/pub/scm/linux/kernel/git/mst/vhost.git config-endian head: d8ede9b5e734748613d3e3108a4026244aa8e41f commit: 392755e77e260e8a173b228e8ecc17917f97d5eb [38/39] fixup! virtio_net: correct tags for config space fields config: arm64-allyesconfig (attached as .config) compiler: aarch64-linux-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget

> 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:

> 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

Well, I am not the person to be asked about how Speex AEC works :) And I was referring speaker input as having direct access into what has been played thru speaker. (ASIO Soundcard provides interface for that) So if I am remebering correctly, with data from Mic and Speaker in sync, speex AEC should do the job. Tim _____ From: Guilherme Balena Versiani [mailto:guibv at

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 output have very fine tolerances on most standard audio

Hi ? I used this "sample counting " method to?resample and put my audio signals in synch. It worked perfectly?in XP machines using a SoundMax?audio card, but it failed in other XPs using Realtek cards. As seen on http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html?my application?continously checked my AEC level to slighly modify resample frequency, but convergence was

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. >

On Mar 15, 2010, at 8:46 AM, Jean-Marc Valin wrote: > If more than one speaker receives the *same* signal, it doesn't matter the > number of speakers. It only gets tricky when the speakers are playing slightly > different signals (e.g. from a stereo song). > Does "tricky" mean that the Speex AEC won't handle such situations well? Or just that you had to be

Hi, The Speex AEC is simply not designed to deal with non-linear echo, as is the case when clipping of AGCs are involved. Make sure all your path is linear or forget about the Speex AEC. Jean-Marc On 11-01-03 11:31 AM, Omer Gilad wrote: > Hi, > I couldn't find a discussion that specifically addresses this, so here > it is. > > I'm using Speex AEC in my mobile VoIP

Just started using 'R'. Nice system. Once quick question: Is it possible to put plot more than one series against a common 'x variable on the same plot? In otherwords, I may want a stacked bar chart of line chart containing more than one series. Could not figure out how to do this. Thanks for your help. Marc Diamond Aviation-Software, Inc. [[alternative HTML version deleted]]

> 1) AGC: This seems to work pretty well in all cases. I had previously > hacked-in the "compander" filter from sox for a similar effect. What > I've noticed is that speex_preprocess's AGC has no "knobs", and it > seems to use an attack/decay that is a lot faster than what I had > chosen from the sox compander, but it works pretty well nonetheless.

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) =

Hi Dmitry, I met the same problem a few months ago and at that time I used 1.2beta2 code/library downloaded from Speex homepage. That bug has been fixed in the most updated source code in svn. Regards, Eddy ----- Original Message ---- From: Dmitry Nikitin <dmitry.nikitin@advatel.com.au> To: speex-dev@xiph.org Sent: Wednesday, December 5, 2007 12:31:57 PM Subject: [Speex-dev] AEC

Hi, Speex devs, I apologize in advance if this is not the proper venue for this question, but I had seen in the archives that other threads of this nature had been discussed. In brief, I'm trying to get AEC working on a simple Android NDK app. It's a basic "play from a file, record from the mic to file" loopback test. I'm using OpenSL ES. I establish a player and a