search for: limaoquan2000

...Sound card problem in acoustic echo >To: speex-dev at xiph.org > >Direct Sound AEC never worked for me, even when it did with Speex.?I was told that after disabling effects in DS the play and record streams could be synchronized,and then Speex AEC did the job. > >--- El vie, 23/7/10, limaoquan2000 <limaoquan2000 at 126.com> escribi?: > > >De: limaoquan2000 <limaoquan2000 at 126.com> >Asunto: Re: [Speex-dev] Sound card problem in acoustic echo >Para: speex-dev at xiph.org, p_j_r_m at yahoo.com >Fecha: viernes, 23 de julio, 2010 22:42 > > >>I remember?...

...is well within the capability of the adaption to keep up. > >Note that a one off measurement of the frequency difference is not >ideal. The DAC and ADC sample rates will change with temperature, and >not necessarily in the same direction. > >Steve > >On 09/30/2010 04:45 PM, LiMaoquan2000 wrote: >> >> 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 sp...

...gt;and now,i would dive first in DirectSound to try to obtain synchronized streams before passing them to speex. Waiting for your good news. >Regarding ASIO, you have to install a special "driver"? for it to work,and this seems unaceptable for general users. >--- El jue, 22/7/10, limaoquan2000 <limaoquan2000 at 126.com> escribi: >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 res...

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

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

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

...put 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 cards. Vas ________________________________________ From: Li Maoquan [limaoquan2000 at 126.com] Sent: Tuesday, April 12, 2011 2:48 PM To: Shridhar, Vasant Cc: speex-dev Subject: Re:RE: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams? Hi Shridhar, Sample rate conversion is not enough to solve this problem. I hav...

...make the echo perceptually tolerable - an approach which has historically worked pretty well (e.g. the DSP Group solution from the 90s). At least one person reported, on this list, that their solution is the best around. > Vas > ________________________________________ > From: Li Maoquan [limaoquan2000 at 126.com] > Sent: Tuesday, April 12, 2011 2:48 PM > To: Shridhar, Vasant > Cc: speex-dev > Subject: Re:RE: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams? > > Hi Shridhar, > > Sample rate conversion is n...

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,

...lerable - an approach which has historically worked > pretty well (e.g. the DSP Group solution from the 90s). At least one > person reported, on this list, that their solution is the best around. > > Vas > > ________________________________________ > > From: Li Maoquan [limaoquan2000 at 126.com] > > Sent: Tuesday, April 12, 2011 2:48 PM > > To: Shridhar, Vasant > > Cc: speex-dev > > Subject: Re:RE: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams? > > > > Hi Shridhar,...

...version. There is a lot of literature out there on the different techniques to do this. A common method is sinc interpolation. This is how I have handle these types of things in the past. Vasant Shridhar From:speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org]On Behalf OfLiMaoquan2000 Sent: Tuesday, April 12, 2011 12:36 AM To: speex-dev Subject: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams? 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-...

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 trou...

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

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-c...

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 dea...

...timing data in order to synchronize m[i] and s[i]. The information is always noisy, due to limited numerical precision, data transfer delay, multithreading, etc." m[i] is microphone signal after ADC, s[i] is speaker signal before DAC. Maoquan > > Steve > > 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 exi...

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 f...

...timing data in order to synchronize m[i] and s[i]. The information is always noisy, due to limited numerical precision, data transfer delay, multithreading, etc." m[i] is microphone signal after ADC, s[i] is speaker signal before DAC. Maoquan > > Steve > > 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 exi...

>>>> 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