similar to: Acoustic echo cancellation

>>>> 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

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

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

>> 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

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

Hi Jean Marc, We are running the speex echo canceller code on a Windows Mobile smartphone using a TI OMAP 201MHz 850. We are making voip calls and audio is working without echo canceller code. Simply adding the following 2 functions whether the echo cancellation is wired in (ie we use the output frame ) or not performance becomes terrible. Note 2 separate threads are used to handle audio.

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

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

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, 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 Is it possible to use only the speex echo cancellation module w/o using the speex codec? Here's the scenario: 1. I have my voice recorded in PCM audio file format 2. I want to cleanup the recorded voice by removing any echo included in the audio file/buffer 3. can I just use the ff APIs? - SpeexEchoState* speex_echo_state_init() - int speex_echo_ctl() - void speex_echo_capture()

May I still try to sync them using some kind of DPLL (digital phase locked loop) mechanism? Something like matching clocks and interpolating samples to try to sync them together? Do the interpolated samples get to be eligible for an aec in your opinion? Thanks, Dario > To everyone on the list: do *NOT* attempt to do echo cancellation with > signals sampled using different clocks. This will

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

Thanks for alerting me to the new changes. I just tried the latest code from SVN, but unfortunately I still have just about the same results. The estimated echo that gets subtracted from the actual echo is such a small signal that it doesn't really result in any noticeable echo attenuation. I currently have my filter size set to 2 seconds even though the echo in my microphone file is only

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

Then why ONE sound card have different capture and playback rate? It must be ONE single physical clock generator which is used by both ADC and DAC in the sound card, isn't it? If you are a hardware engineer. Will you design two different physical clock for ADC and DAC seperately? What on earth causes this problem? Who knows its intrinsic real reason? Isn't there any other solutions? For

From: Steve Underwood <steveu at coppice.org> > It seems some cards use a PLL for their ADC, so they can lock to an > incoming SPDIF signal, but always use a local crystal clock source for > their DAC. These cards do not have their ADC and DAC synchronised. Do common on-board or PCI sound card lock to some incoming signal? Yes, there is a crystal oscillator and a PLL or divider to

Hi Folks, I am having trouble using speex_echo_cancel. As a starting point, I am using the testecho.c source code. I compiled the 1.2beta1 version. I have not tried any other versions of speex. The document says that the order of arguments to Speex_echo_cancel is (echo_state, input_frame, echo_frame, output_frame, residue) where "input_frame" is as captured from mic and

Had a try. The reason why a simple delay is not that good is mainly due to the initialization of the filter parameter that still takes a few seconds (if they are perfectly in sync, you sort of get lucky). Otherwise, you real recording seems to have something odd in it. Are you sampling from a different card then the one that's playing the sound? or maybe the mic (or something else) in the room

Hi all, The conclusion of the discussion is that most sound cards indeed have different capture and playing frequencies for the unknown reasons. But we all know the adaptive filter of the AEC relies on the synchronization of the far-end and near-end sampling rates. Then Has anybody tried to use speex AEC in Windows system? How do you solve this problem? (I have tested speex AEC. In most