similar to: AEC in wideband

Hi! In a message posted on May 31 Jean-Marc wrote in a topic about AEC: > - set sampling rate to 8 kHz (at least for now) Does it mean that echo cancellation algorithm won't work in wideband mode? I can't get aec to work well in my application and I am wondering whether it's not a matter of using wideband instead of (preferred?) narrowband mode? Thanks in advance for any

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? 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

> 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:

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

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

Hi Jean-Marc, You make that sound like its just a matter of meeting some arbitrary spec. Let's be more specific..... If you use narrow band voice down to deep bass frequencies: - 16 bit linear audio sounds good - alaw or ulaw sounds muddy - low bit rate codecs, like speex or G.729, sound awful. I assume QinBin only listened to some uncompressed audio in his evaluation.

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 codec), you're not really supposed to have anything below ~200 Hz, but a lot of people forget that. Jean-Marc On 2010-03-03 02:40, brant wrote: > Hi, > > The notch filter in

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

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

Hi, The notch filter in AEC is only used to remove DC signal, and the time of convergence is not important, right? If so, I think preset value of notch_radius is too small, and it causes noticeable distortion(freq < 200hz cut). There is a picture in attachment to show signals under different radius in time-domain. By setting notch_radius to 0.999 for all sampling rates, I found better

Hi! I have a few questions considering echo cancellation algorithm in Speex: 1. In the manual it's stated that the delay between the input signal and the echo signal must be small. How small should it be? Is for example 100 ms acceptable? 2. Does echo cancellation algorithm deal well with situation, when one of the users of Speex-based VOIP application has "record-what-you-here"

> 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

BTW, when you have something working and stable, I could include it in the main Speex distribution. Jean-Marc Le mar 18/02/2003 à 20:57, Bernard Blackham a écrit : > On Tue, Feb 18, 2003 at 06:09:43PM -0500, Jean-Marc Valin wrote: > > Le mar 18/02/2003 ? 17:38, John Hayes a ?crit : > > > If I encode something in ultra-wideband, can I decode it in wideband by > >

I have read the speex manual, looked at the example code, but for some reason the first encoded byte does not have the wideband bit set and the mode bits are also wrong even though I am encoding in wideband mode. I am sure that I am doing something wrong, but I can't figure out what it is. I have included my code below.... I would expect that the MSB of the first encoded byte is 1

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

I downloaded speex-1.2beta3-win32 binaries, and used the speexenc.exe and speexdec to process the file male.wav download from http://speex.org/samples I used both the narrowband and ultra-wideband mode: >speexenc -n --quality 3 male.wav n3.spx Encoding 8000 Hz audio using narrowband mode (mono) >speexdec n3.spx n3.wav Decoding 8000 Hz audio using narrowband mode (mono) Encoded with Speex

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