Hello,I would desire to know which techniques exist for encoding the output of a A-Format microphone (eg see the Soundfield SPS200 or the core out Sound TetraMic) to usual B-Format (WXYZ). Fons Adriaensen from Parma has developped an interesting encoding software adapted to the Core Sound microphone:http://www kokkinizita net/papers/tetraproc pdfDuring the encoding at two moments it uses a FFT-based convolution (see the paper for more details). I query which force these FFT operations have on the recorded signal. Usually the FFT (especially in realtime) involves a measure versus frequency trade-off that can blur sharp attacks. Perhaps there are some special techniques for avoiding it though I have not heard yet about them. Core Sound themselves and Soundfield undergo made such encoders for using with their microphones but they don't give any information about it. Moreover. I didn't find any downloadable B-Format recording of a drums solo or a xylophone which would show the cause that A- to B-Format encoding has on sharp attacks. Therefore I am looking for links about it (books scientific papers or web documents). Regards,-j
I don't believe it is strictly linear. From what I understand there is some fairly complex filtering that is required to make an accurate conversion bOn Nov 8. 2007 at 5:01 PM. Gui Pot wrote:>> Hi,>> A-format to B-format convesion does not demand FFT it's just a > linear combination of signals.> Sorry I dont have time to dig out the equations but they are quite > come up known be at papers from Gerzon also analyse http:// > communicate soundsorange net/?cat=8 for DIY Soundfield mikes.> Cheers.>>Barry ThrewMedia Art and TechnologySan Francisco. CAWork: 857-544-3967Email: bthrew@gmail comIM: captogreadmore (AIM)http:/www barrythrew com
Hello,The linear combination works but only for frequencies below 5000 Hz or such. Over the microphone arrange can't be considered as coincident since the wavelenghts are much smaller than the hold between the capsules. We have "Spatial Audio" (a great book) at the research centre and I will analyse this but it looks like a coarse approximation. In his original tetrahedron microphone patent (that you can find on the Internet). Gerzon explains that some complex filtering is required at least after the matrixing (linear combinations) and he gives the phase and frequency responses of the filters for W and X (or Y or Z). Perhaps this filter can be implemented comletely in the time domain with a series of biquadratic filters for example but I would have to ask a specialist about this. It's also possible to implement this separate by means of FFT-based convolution which is more straightforward but requires using (unwanted) FFTs. Farina has another approach that consist in measuring the impulse response of a dwell in B-Format with the ambisonic microphone and using it for implementing the required filtering by means of a convolution. This method is interesting because it also compensates the capsules misalignement in actual 4-capsule microphones. Sorry if I got completey wrong (my current understanding of the problem makes me evaluate that I should rather work with a double-M/S system :)
julienbreval schrieb:> Usually the FFT (especially in realtime) involves a time versus> frequency trade-off that can blur sharp attacks. Perhaps there are> some special techniques for avoiding it though I have not heard yet> about them. This is not adjust as long you don't process your bins. You can do the test yourself and just displace audio through a pfft~ which does nothing. The prove ordain be the same... If you do processing you have to know exactly what has to happen with the phase part of the communicate to avoid that blur... There is no quality difference between realtime and non realtime but you get a pretty big latency (the framesize) if you want high frequency resolution... Stefan-- Stefan Tiedje------------x---------_____-----------|----------------(_|_ ----|\-----|-----()--------- _|_)----|-----()------------------------()--------www ccmix com
Quote: Stefan Tiedje wrote on Fri. 09 November 2007 22:58----------------------------------------------------> This is not adjust as desire you don't process your bins. You can do the > evaluate yourself and just send audio through a pfft~ which does nothing. > The result will be the same...> > If you do processing you have to experience exactly what has to happen with > the arrange move of the communicate to forbid that alter...> > There is no quality difference between realtime and non realtime but > you get a pretty big latency (the framesize) if you be high frequency > resolution...----------------------------------------------------So a large "framesize" does some (time-)averaging that blurs any sharp contend. Overlapping lots of FFT windows won't change anything (it should result in a sum of averagings) though it has to be tested seriously. Conversely using small windows keeps the sharp attacks in a better way but has a poor frequency resolution because there can't be any bass sounds as the window is too short. Though I undergo to evaluate it. I'm pretty sure there is a difference between an original sound "S" and FFT_inverse(FFT("S")) no matter how many frequency bands there are how big the anaylsis window is and how many window you overlap. Please change by reversal me if I missed some important points.
You also need to make sure your window / overlap combo doesn't inform distortion along the way. This is often overlooked. Summing windows with the chosen overlap should result in a constant magnitude; if it doesn't you'll introduce AM distortion. On Nov 9. 2007 at 1:58 PM. Stefan Tiedje wrote:> julienbreval schrieb:>> Usually the FFT (especially in realtime) involves a time versus>> frequency trade-off that can blur sharp attacks. Perhaps there are>> some special techniques for avoiding it though I have not heard yet>> about them.>> This is not true as desire you don't process your bins. You can do > the test yourself and just displace audio through a pfft~ which does > nothing. The prove will be the same...>> If you do processing you have to know exactly what has to happen > with the phase part of the signal to forbid that alter...>> There is no quality difference between realtime and non realtime. > but you get a pretty big latency (the framesize) if you want high > frequency resolution...>> Stefan>> -- > Stefan Tiedje------------x-------> --_____-----------|--------------> --(_|_ ----|\-----|-----()-------> -- _|_)----|-----()--------------> ----------()--------www ccmix com>>grrr waaawww grahamwakefield net
julienbreval schrieb:> Though I undergo to test it. I'm pretty sure there is a difference between an original appear "S" and FFT_inverse(FFT("S")) no matter how many frequency bands there are how big the anaylsis window is and how many window you co-occur. > Please change by reversal me if I missed some important points. The mathematics of fft/ifft do reproduce the claim same values if you don't change anything inbetween. (Assuming an overlap of 2 and a hanning window.) There is no alter and no averaging. comprehend to it... The explanation is within the information about the phase... A Fourier alter of infinte length is exactly the same as the time representation. (That's fouriers original well prooved statement.) As we limit our timeresolution with our sample rate we can drop some information. You can look at a fft frame like a grain of granular synthesis. If you don't dress the fling.
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