similar to: Speex-dev Digest, Vol 83, Issue 17

2011/4/20 Li Maoquan <limaoquan2000 at 126.com> > Simply to say, in a quiet room, you can play a impulse signal and then find > it's impulse response signal from the > microphone. For example, if the delay between the impulse signal and its > response signal range from 500 to > 3000 cycles, you can buffer the far-end signal to 0-300 cycles and set the > filter length

Simply to say, in a quiet room, you can play a impulse signal and then find it's impulse response signal from the microphone. For example, if the delay between the impulse signal and its response signal range from 500 to 3000 cycles, you can buffer the far-end signal to 0-300 cycles and set the filter length to 4000. It is also called to align far-end signal and near-end signal. BTW: Speex

>>>> Hi, >>> >>> I have a scenario in a mobile VoIP app that requires echo cancellation but >>> is somewhat different from what's described in the docs. >>> >>> Audio is received from and sent to the network at 8000Hz. Each packet >>> contains 160 samples worth a playback of 20ms. >>> >>> But the hardware

Hi So in your case if you don't want to use speex_echo_cancellation than you should do next way: Save the playback frames (clean echo) within separate playback thread with speex_echo_playback func. Then use speex_echo_capture - it will remove the clean echo frame (saved with speex_echo_playback func) from const spx_int16_t *rec buffer (saved with speex_echo_capture) and you will get

Well, I'm not a professional in AEC theory, but what I've mentioned is: speex_echo_state_init(20, 320*10) - frame size should correspond to 20ms. At your sampling rate (16000hz) is should be 16K*0.02 = 320. The same I can notice about echo tail. 100ms: 16000*0.3 = 4800, not 3200 as you has. But that's not crucial I think. Just wanna you get the point. "Internally,

_____ From: Mohammed Sulaiman [mailto:mhh.sulaiman@talk21.com] Sent: Monday, February 11, 2008 12:53 AM To: Dennis Liu Subject: RE: [Speex-dev] Echo Canceller on Windows Mobile Hi, thanks. One question did you use these 2 functions speex_echo_playback speex_echo_capture No, I didn't use them. I use my own code to buffer microphone and speaker data. or do you have your audio in

Hi Anton Thanks for the info... What if I use this function do I still need to sync the playback and capture? Assuming that the captured voice (*rec) already contains the echo which I need to cancel out... -------------------------------- void speex_echo_capture (SpeexEchoState *st, const spx_int16_t *rec, spx_int16_t *out) Parameters: st Echo canceller state rec signal

I wasn't implying that anyone do anything about it, just that's it a real problem. Unfortunately, most of the crappy sound cards are the ones that ship with your typical PC, so it's just something that people should be aware of. The solution is pretty straightforward -- just resample the audio data in real time using a reference clock. -----Original Message----- From: Jean-Marc

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 somewhere in some of the OS resampling algorithms, but I've seen that on many machines.

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

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 >

Hello all. A few questions on using speex's echo cancellation, for clarifying: 1) I assume that the echo_frame in this method: speex_echo_playback(echo_state, echo_frame) is just raw "pcm_s16le" type bits/data? I'm a bit new to all this... 2) I assume from this thread: http://lists.xiph.org/pipermail/speex-dev/2009-March/007151.html that speex_echo_playback and

hello jean-marc and everybody, I keep getting no results when trying to use speex_echo_capture, speex_echo_playback and speex_echo_cancel in a multi-threaded application, as suggested in the manual. Though, the cancellation works properly when i use a file with human voice for far-end input and the same file with echo added in SoundForge for mic input. When i try to insert and remove silence

Terrific! The patch fixes my problem quite decisively. I will want to use this with a 16K sample rate. Is it expected to work well with just another slight patch, or will we need to modify it more deeply? Also, are there any tunable parameters besides tail length? I am wondering about the aggressiveness of adaptation and I know you are an expert in that area. Thanks so much, Michael

> Hi, > > I have a scenario in a mobile VoIP app that requires echo cancellation but > is somewhat different from what's described in the docs. > > Audio is received from and sent to the network at 8000Hz. Each packet > contains 160 samples worth a playback of 20ms. > > But the hardware requires aggregation for both playback and capture. So for >

>>* Hi, *>>* *>>* I have a scenario in a mobile VoIP app that requires echo cancellation but *>>* is somewhat different from what's described in the docs. *>>* *>>* Audio is received from and sent to the network at 8000Hz. Each packet *>>* contains 160 samples worth a playback of 20ms. *>>* *>>* But the hardware requires aggregation for

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

Anton A. Shpakovsky <saa <at> tomsksoft.com> writes: > > 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

Hi, when I use preprocessor with AEC, VC++ alerts me about heap corruption. I have protected speex_echo_playback, speex_echo_capture and speex_preprocess_run with mutexes, to avoid echo_state being used at the same time, but it still happens. Any help about this ussue? Thank you.

> 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