similar to: How to use aec correctly?

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

? 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 flexibility in chosing the frame lenght? on what parameters does the frame length choise depends? thank you all, Shri. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I'm just new to this list. I'm user of a voIP application which uses Speex Acoustic Echo Canceller (so only user of speex). The only parameter it has to tune is the length of the ec tail. I'm testing the echo canceller on an arm9 platform. Tests made only allow me to raise the length of the tail until 250 msec (upper the cpu power consumption is 100% and there are audio

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it. It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using

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

Hi, my VoIP system uses speex with framesize = 160 samples(20 ms) at 8khz, sending and receiving paquets of 1600 samples(200 ms). When I receive a packet, I buffered it (I have also tried with speex_echo_playback) before sending to the speaker. When I capture from microphone, I fist remove DC offset (I saw in OPAL sources) and then I call speex_echo_cancellation for every one of the 10

Hi, Arun and me are working on a echo-canceler module for pulseaudio build around speex. When using the preprocessor's echo filter, specifically in combination with denoise and AGC, it's unclear whether these have to run before or after the AEC. Right now we've got one instance of speex_preprocessor doing AGC and denoise running before the echo canceler; and a second for echo

Hi, Can anybody show me how to enable stereo AEC, I tried with the following code and the result is bad, degraded output. Init: ec_state = speex_echo_state_init_mc(frame_size, aec_tail, 2, 2); speex_echo_ctl(ec_state, SPEEX_ECHO_SET_SAMPLING_RATE, &sample_rate); preprocess_state_left = speex_preprocess_state_init(frame_size, sample_rate); preprocess_state_right =

Le vendredi 11 novembre 2005 ? 01:21 -0800, Duane Storey a ?crit : > This is a very real problem though.. I've encountered many sound cards that > use different clocks for input and output (even on the same card!) Also, if > you open up a sound device on windows at 8kHz, the microphone is often > around 8100Hz, while the output is 8000Hz.. I'm not sure if there's a bug >

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 Mon, Aug 11, 2008 at 12:34 PM, Jean-Marc Valin < jean-marc.valin at usherbrooke.ca> wrote: > 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.

To everyone on the list: do *NOT* attempt to do echo cancellation with signals sampled using different clocks. This will *NOT* work. Just a 0.1% difference between the two sampling rates (it's sometimes worse than that) means that the impulse response drifts by 8 samples every second. There's just no way to efficiently track this. Or at least no way that doesn't involve something 100x

I ran some further tests on mdf and here are the results: 1. reduced tail length to 100ms, aligned mic and speaker signals to within 10ms - almost no echo attenuation 2. aligned mic and speaker signals to within 5 samples - still almost no echo attenuation 3. ran testecho using the same file for mic and speaker - very good echo cancellation (of course this is expected, but I needed to do a sanity

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

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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

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) =

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

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

> Yes, you are not forced to reset the AEC between calls at all, if you > can assume the actual echo path (physical environment) will not change > much, you can keep the instance and continue using it for the next call. > > Also what we did was keep the AEC running between calls - but we had a > rather high-duty system with sounds and signals played between calls, so > the AEC