similar to: Sound card problem in acoustic echo

Displaying 20 results from an estimated 5000 matches similar to: "Sound card problem in acoustic echo"

2010 Jun 10
1
Sound card problem in acoustic echo cancellation
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
2010 Jun 09
3
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 Apr 19
1
Acoustic echo cancellation
>>>> 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
2010 Sep 30
1
Sound card problem in acoustic echo
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,
2011 Apr 21
3
Acoustic echo cancellation
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
2010 Oct 01
0
Sound card problem in acoustic echo
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
2010 Jul 24
1
Sound card problem in acoustic echo
>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
2010 Jul 22
1
Sound card problem in acoustic echo
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
2011 Jan 19
3
About Sampling Rate Correction in acoustic echo cancellation
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,
2011 Apr 12
4
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-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
2011 Apr 21
0
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
2009 Aug 11
2
AEC troubleshooting
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2011 Apr 13
1
Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams?
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
2007 Jul 25
2
Speex optimization and 12 bits conversion for 12 bits ADC
Hi?Jean-Marc? Thanks for your suggestions very much! > > I am porting speex on ARM7TDMI, I have done some optimization, the > > result is that the encoder and decoder need about 60 MCPS for 5.96kbps > > bitrate and complexity 0. Can someone give me informtion about Speex > > optimization on ARM7? > > That's quite good. A few suggestions here: > 1) Don't
2011 Apr 12
1
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 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
2007 Jul 24
2
Speex optimization and 12 bits conversion for 12 bits ADC
Hi, all, I am porting speex on ARM7TDMI, I have done some optimization, the result is that the encoder and decoder need about 60 MCPS for 5.96kbps bitrate and complexity 0. Can someone give me informtion about Speex optimization on ARM7? Another question, my ADC and DAC are 12 bits, but Speex codec is 16bits, Did someone try to modify speex to 12 bits? I think if I modify speex to 12 bits, the
2011 Apr 14
2
Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams?
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
2006 Aug 17
2
AEC on a TI C6x - has no effect
Itay, >I am trying these things, but the main problem that has been bothering >me recently is that the fixed-point algorithm works "sometimes". Meaning >that sometimes it will work well, and other times it will not work at >all. > >I think I've found the source of the problem. The speex_alloc() >function, called by speex_echo_state_init(), calls calloc() and
2011 Apr 16
0
Speex-dev Digest, Vol 83, Issue 10
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
2010 Jan 15
3
Inexpensive Flac player for separates system?
I'm after the cheapest way to decode a flac stream with following criteria; - Transport from UPnP (DNLA) NAS using either WiFi or Cat5 networking - Toslink digital out for using existing HiFi DAC. - Low power consuption (eg just few Watts - it's just a decoder anyway). There seems a real lack of Flac players that are cheap and HiFi separates integratable like current-day CD players. All