On Thu, Jun 3, 2010 at 9:52 AM, arthur de conihout < [email protected]> wrote:
> Hi, > thanks for your answer > > *why don't you compute all possible versions beforehand* > > > thats exactly what i m doing presently cause i m using a 187 filters > database(azimuth and elevation).I would love to be able to reduce the angle > thread under 5° which triggers around 1000 files to produce.For a 3Mb > original sound file, it becomes huge. > > Indeed ! I'll be curious to see what solutions ends up working best. Keep us posted. David > Thanks > > Arthur > > > 2010/6/3 David Huard <[email protected]> > > Hi Arthur, >> >> I've no experience whatsoever with what you are doing, but my first >> thought was why don't you compute all possible versions beforehand and then >> progressively switch from one version to another by interpolation between >> the different versions. If the resolution is 15 degrees, there aren't that >> many versions to compute beforehand. >> >> David >> >> On Thu, Jun 3, 2010 at 6:49 AM, arthur de conihout < >> [email protected]> wrote: >> >>> Hello everybody >>> >>> i m fighting with a dynamic binaural synthesis(can give more hints on it >>> if necessary). >>> >>> i would like to modify the sound playing according to listener's head >>> position. I got special filters(binaural ones) per head position that i >>> convolve in real time with a monophonic sound.When the head moves i want to >>> be able to play the next position version of the stereo generated sound but >>> from the playing position(bit number in the unpacked datas) of the previous >>> one.My problem is to hamper audible artefacts due to transition. >>> At moment i m only using *short audio wav* that i play and repeat if >>> necessary entirely because my positionning resolution is 15° >>> degrees.Evolution of head angle position let time for the whole process to >>> operate (getting position->choosing corresponding filter->convolution->play >>> sound) >>> >>> *For long **audio wav* I could make a fade-in fade-out from the >>> transition point but i have no idea how to implement it(i m using audiolab >>> and numpy for convolution) >>> >>> An other solution could be dynamic filtering means when i change position >>> i convolve the next position filter from the place the playing must stop for >>> the previous one(but won't pratically stop to let the next convolution >>> operates on enough frames) in accordance with the filter frame length(all >>> the filters are impulse response of the same lenght 128). >>> >>> The "drawing" i introduce just below is my mental representation of what >>> i m looking to implement, i already apologize for its crapitude (and one of >>> my brain too): >>> >>> >>> t0_________t1__t2__t3___________________________________________________________t=len(stimulus) >>> monophonic sound(time and bit position in the unpacked datas) >>> >>> C1C1C1C1C1C1C1C1C1C1C1... >>> running convolution with filter 1 corresponding to position 1 (ex: angle >>> from reference=15°) >>> >>> P1_______ >>> sound playing 1 >>> >>> ^ >>> position 2 detection(angle=30°) >>> >>> C2C2C2C2C2C2C2C2C2C2C2... >>> running convolution with filter 2 >>> >>> P1_____x >>> keep playing 1 for convolution 2 to operate on enough >>> frames (latency) >>> >>> FIFO >>> fade in fade out >>> >>> P2_________ >>> sound playing 2 >>> >>> >>> I don't know if i made myself very clear. >>> >>> if anyone has suggestions or has already operated a dynamic filtering i >>> would be well interested. >>> >>> Cheers >>> >>> Arthur >>> >>> >>> _______________________________________________ >>> NumPy-Discussion mailing list >>> [email protected] >>> http://mail.scipy.org/mailman/listinfo/numpy-discussion >>> >>> >> >> _______________________________________________ >> NumPy-Discussion mailing list >> [email protected] >> http://mail.scipy.org/mailman/listinfo/numpy-discussion >> >> > > _______________________________________________ > NumPy-Discussion mailing list > [email protected] > http://mail.scipy.org/mailman/listinfo/numpy-discussion > >
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