similar to: Sound card problem in acoustic echo cancellation

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

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 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

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,

>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

>>>> 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

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

I read the question: http://stackoverflow.com/questions/3403152/acoustic-echo-cancellation-in-java/3403198 But my problem is that my project is using Encoding and decoding mechanism using JSpeex and I am not sure whether I can integrate the above java code with my project. So can someone tell me how can I implement Acoustic Echo Cancellation (AEC) using JSpeex(Java Port of Speex).

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,

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

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 ? basically all of the GIPS DSP stuff (AGC, VAD, Denoise, echo canceller, etc),

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 ?

2011.06.21. 19:22 keltez?ssel, Arun Raghavan ?rta: > 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) >>

Hi Steve, Have you read this paper? (Heping Ding, David I. Havelock, Drift-Compensated Adaptive Filtering for Improving Speech Intelligibility in Cases with Asynchronous Inputs. EURASIP J. Adv. Sig. Proc. 2010:) Let me call is paper-Drift. It provided a method to evaluate Relative Sample Offset (RSO, d[i]) which is omitted in the microsoft paper (Challenges and Solutions for Designing Software

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

On Tue, 2011-06-21 at 20:18 +0200, Kadinger Andr?s wrote: > 2011.06.21. 19:22 keltez?ssel, Arun Raghavan ?rta: > > 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. >

Hi Steve, > I don't know if this has only recently been put on line, but I never > noticed it until today - > www.iwaenc.org/proceedings/*2008*/contents/papers/9044.pdf > > That paper is from people at MS describing, in some detail, what the > Windows kernel echo canceller does to handle synchronisation issues. It > tracks both time varying sample clock drift and hiccups

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