similar to: AEC in wideband mode

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

Hello! I am using speex in wideband mode and I'm trying to get AEC to work. It seems the algorithm is working, but the result is not satisfactory. Output signal of the algorithm still contains echo signal, although it's level is diminished and sounds like processed with low-pass filter. So the echo is audible, although it's quite quiet and the speech it contains incomprehensible. I

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: comunip.gif Type: image/gif Size: 1663 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.gif

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

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

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

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.

Hello? I am a new user of speex.I am currently working on speex frames and I have some questions. I am using narrowband and long tail length, and it works very well with speex test DEMO. But it is very difficult to have speaker input in perfect sync with mic input. Speex does not work at all. Any suggestion? Regards? -------------- next part -------------- An HTML attachment was scrubbed...

It looks like I'll need to go further into the guts of speex to do this. I do, however, see some lines in nb_celp.c/nb_decode() that look interesting. I guess I'll play with it. I doubt that it will be terribly clean, though. Jean-Marc: Take a look at line 1195 in nb_celp.c (CVS). It reads "speex_warning ("More than to wideband layers found: corrupted

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"

Is there any way to access only the narrowband portion of a wideband stream? I'd like to be able to encode the audio only once, but allow members in a conference to have some rough selection of bandwidth, and allow them to move to a lower-bitrate stream if there is a need to do so. Thanks, Matthias -- Matthias Granberry matthias@utdallas.edu (469) 371-0596 --- >8 ---- List archives:

Dear Speex developers, I am currently experimenting with Speex on Symbian smartphones. I have compiled the Speex library, and I am now using it in the following way: 1. Record 320-byte buffers of data in PCM16 format, 8000 Hz sampling rate. 2. Feed the resulting buffer to an instance of a narrowband Speex encoder. 3. Send the encoded data over RTP. 4. Upon receiving on the other side, feed the

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 > > chopping off bytes in every frame? > > All you have to do is use the --force-wb switch with speexdec. It will > decode as if the file were wideband, ignoring the ultra-wideband

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

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

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 couldn't find a discussion that specifically addresses this, so here it is. I'm using Speex AEC in my mobile VoIP application to cancel speaker echo. The used version is 1.2rc1 from the website, and I'm compiling with fixed-point. On most occasions, the AEC works very well and cancels most of the echo (combined with the preprocessor). On some devices, where the microphone signal

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

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 Jean-Marc, Regarding you points: 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 ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is