similar to: How do I implement Acoustic Echo Cancellation (AEC) using JSpeex(Java Port of Speex)

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, 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,

Hi All, I am a research associate in the Hong Kong Polytechnic University. One of my research interests is acoustic echo cancellation. Now I meet a big problem. When I was testing my own AEC module, I found that it was almost perfect in few computers but much worse in other computers. Then I tried AEC module in speex, it was almost the same result. Then I found the reason is that sound cards of

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),

>>>> 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

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

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 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 ?

On 06/22/2011 04:57 AM, Arun Raghavan wrote: > On Tue, 2011-06-21 at 11:39 -0700, Arun Raghavan wrote: > [...] >> I'm also running this on x86 (x86_64, technically), and it's all >> floating-point, so I guess this is a regression somewhere. Will try to >> see if I can run it without any optimisations if possible, which I >> assume should serve as an adequate

>> 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 >>

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 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

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

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 04/14/2011 07:26 PM, LiMaoquan2000 wrote: > 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,

Background: I'm working on a web conferencing application hosted in Silverlight, and we desperately need an echo canceller. A while back, someone made a Java port of Speex (JSpeex), which was later ported to C# (CSpeex). However, those are based on an older version of the Speex code, and while they seem to interoperate just fine with the latest official release, they don't have certain

Well, AFAIK AEC engine needs time to adapt. So probably that's why at the beginning of the conversation you hear the consequences of this adaptation. Btw, what sound system do you have? I mean laptop or arbitrary acoustic system? I'd like to know, cause I couldn't get enough echo cancellation with speex in my environment with standalone speakers AT ALL. -----Original Message-----