similar to: Acoustic echo cancellation

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, *>>* *>>* 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

>>>> 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

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 requires aggregation for both playback and capture. So for playback, I coalesce 4 packets in a buffer and queue them as a

>> 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

_____ 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 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

On 01/19/2011 06:44 PM, LiMaoquan2000 wrote: > > Hi all, > > We have discussed so many about sampling rate asynchronous (or offset) > between rendering (D/A converter) and capturing (A/D converter) of > most PC soundcards. It seems all acoustic echo cancellers, include AEC > in speex, can not deal with this trouble, because it causes a drift of > echo path and also

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

Hi Underwood, Thank you for your help. I agree with your opinion. But it is almost impossible to further reduce the frequent difference between play and capture. 1. I used a 2^18 step FFT to analyse the echo frequency. So the freq resolution is 8000HZ/(2^17)=0.0625Hz. The analyser need at least 2^18/8000=32 seconds acoustic echo record signal from the microphone. Better freq resolution relies

On 01/20/2011 04:26 AM, Steve Underwood wrote: > On 01/19/2011 06:44 PM, LiMaoquan2000 wrote: >> Hi all, >> >> We have discussed so many about sampling rate asynchronous (or offset) >> between rendering (D/A converter) and capturing (A/D converter) of >> most PC soundcards. It seems all acoustic echo cancellers, include AEC >> in speex, can not deal with this

Hi all, We have discussed so many about sampling rate asynchronous (or offset) between rendering (D/A converter) and capturing (A/D converter) of most PC soundcards. It seems all acoustic echo cancellers, include AEC in speex, can not deal with this trouble, because it causes a drift of echo path and also buffer overflow and underflow which jumps the delay of echo path seriously. Unfortunately,

2006/11/2, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>: > > Today I tried speex echo cancellation on my VoIP application running > > on a windows mobile based PDA (OMAP1510 168MHz). The result is > > disappointing: it takes 150-250 ms to do echo cancellation (by calling > > speex_echo_capture() ) for a 20 ms frame! > > Are you by any chance using an insane

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,

Hello, i'm adding echo cancellation to my voip application, but when i call speex_echo_capture() or speex_echo_playback() causes an assertion failed, it seens that there is a delete in any where. i've encapsulated echo canceller in a class that storage the state and simply call the functions: void EchoCanceller::Playback(short *echo_frame) { speex_echo_playback(echo_state,

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.

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 All, In order to deal with acoustic echo cancellation problems of most PCs which sound cards have different capture and play frequencies. I made a trial. At first, a 1000Hz sine wave is played for a long time via a speaker and its acoustic echo is recoreded. Seconds, get the frequency of the echo by a FFT analyser. So the difference between capture and play frequencies is obtained. Thirdly,

Hi Jean-Marc and all others! I am having some troubles with the echo cancellation feature of speex in a voip project. I have two threads, one for sound input and one for sound output. In the output thread, I call speex_echo_playback(). or every frame sent to the sound card. In the input thread, I call speex_echo_capture() for every frame captured from the sound card. Also, I do