search for: acoustically

All, I'm trying to set up a function which calls the partialPlot function but am getting an error that I can't seem to solve. Here's a simplified version of the function and error... > pplot <- function(rf,pred.var){partialPlot(x=rf,pred.data=acoust,x.var=pred.var)} > > attach(acoust) > acoust.rf <-

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,

Hi, member-list i'm searching for packages or routines to analise fisheries acoustics data. Something make import data of scientific ecosound (EK500) of paralel port, filter signal, analyse pattern of pixel fish shoals... Is there someone work with this ? If no, why dont make colective effort to do this?? (package fish_acoustic). Thanks Marcelo

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

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

*August 12, 2012* ** *Postdoctoral Researcher working on spatial models of marine wildlife, using R. * Integrated Statistics is looking for a post doctoral scientist to work with the Passive Acoustics Research Group and Large Whale Research Group, part of the Protected Species Branch at the Northeast Fisheries Science Center in Woods Hole, Massachusetts

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being

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,

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

Hi everyone, For those who like to play with experimental stuff, I've got something new to test. If you compile with --enable-vorbis-psy , Speex uses a variant of the Vorbis psycho-acoustic model to shape the coding noise. So far, I've obtained a slight increase in quality, but I'm interested in feedback from others. This still needs a lot of tuning and has received only a minimal

Hello? My name is Hyeonwoo Oh. I?m developing video conference software using speex library. Can you give me some sample codes for acoustic echo canceller and noise reduction? I tried noise reduction API, but failed. Normal encode/decode API is working well. Thank you very much for your good job. I?m very pleased with speex. Best regards, Hyeonwoo Oh

>> 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 >>

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).

I wonder what psycho-acoustic model is used in ogg vorbis. Did they create their own model (this requires big listening tests) ? Or is it something that was already there, like gpsycho ? <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org'

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

After reading http://mp3.radified.com/mp3.htm I sent Rad an explanation of some things as I understand them. He liked it and posted it on his site (still unlinked, use the URL below). Can somebody with better understanding of psycho-acoustic terms and the vorbis model check it and comment on it? In particular I didn't know how vorbis handles quantization noise. If you reply with

Hi, Does anyone have an idea about the possibility to apply psychoacoustic models as the ones in mp3 or AAC to a CELP coder? Thanks! /Pontus <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the

> Hi everyone, > > For those who like to play with experimental stuff, I've got something > new to test. If you compile with --enable-vorbis-psy , Speex uses a > variant of the Vorbis psycho-acoustic model to shape the coding noise. > So far, I've obtained a slight increase in quality, but I'm interested > in feedback from others. This still needs a lot of tuning

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),