similar to: About Sampling Rate Correction in acoustic echo cancellation

Hi all, We all know that mismatch between clocks of ADCs of far-end voice and near-end voice is not allowed in a time-domain or frequency-domain LMS based AEC system. It means that capture and render audio streams must be synchronized to a same sample rate. However, I found that this restriction is removed in microsoft AEC from Windows XP SP1. Anyone knows how microsoft AEC do it? This technology

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,

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

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, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set

Hi Shridhar, Sample rate conversion is not enough to solve this problem. I have tried this method several months ago. The first step is to measure the difference between sample rate of capturing and rendering. Then resampling (by what you said "sinc interpolation") one signal to eliminate the difference. The frequency step in my experiment is less than 0.1Hz. I have tried speex AEC

On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and

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 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

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

>I remember?I had to expose the echo cancelation level implementing a get_echo_level( ) function based on this: >http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html This is really a good idea to determine the frequency difference between capture and play of the sound card. But it need constant far-end voice and a long time because it must repeat the process of

Thank you. But it will cost you a long time to get the accurate play and capture frequencies. Does your program test two frequencies of the sound card each time? Because different sound cards have different frequency errors. And the resampling program is also time consuming because the target frequency is so close to the sampling frequency of the input signal, isn't it? I have tested program

>> There is also a IEEE paper, Adaptive Sampling Rate Correction for >> Acoustic Echo Control in Voice-Over-IP, which introduced a complex >> method to estimate the frequency offset and resynchronize the signals >> using arbitrary sampling rate conversion. I wonder if it can provide >> enough performance. Because I have also designed a sampling rate >>

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, >>> >>> 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 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

I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and output have very fine tolerances on most standard audio

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

On 06/22/2011 09:30 AM, Steve Kann wrote: > Speaking of AEC (thought not quite on topic for this thread), > > Has anyone on this list played with the GIPS code that google just > open-sourced? It looks like their AEC also has code to handle > differential sample rates, though I haven't really evaluated it > thoroughly. > > There is really a lot of code in the drop ?

Hi Andras, others, Andras Kadinger <bandit <at> surfnonstop.com> writes: > > > Daniel, > I recommend you to start from a simple case and gradually progress > towards your goal. > Can you make things work with the "Speex in a Disco" (Example 6) > testcase at http://ns.surfnonstop.com/~bandit/speex/echocard1/ ? > These files