Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?
I think I figured it out. I use 2^octave * SR/FFTsize -> toERBscale -> * log2(FFTsize)/42 as a scaling factor for the windows. Means the window of the top octave is about 367 samples at 44100 SR - does that seem right? Sounds better but not so different, still pretty blurry and somewhat reverberant. I used the lower frequency limit of the octaves for the window sizes and Hann windows cause I don't want the windows to be too small. Do you think using Gaussian windows and the center of the octave will make a big difference? Or do I just need more overlaps in resynthesis now? ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?
Further tests let me assume that you can do it on a log2 scale but that appropriate window sizes are crucial. But how to derive these optmal window sizes I am not sure. I could calculate the bandwitdh of the octave band (or an octave/N band) in ERB for instance but then what? How do I derive a window length from that for that band? I understand that bandwitdh is inversly proportional to window length. So it seems very easy actually but I am stuck here... Am 06.11.2018 um 16:13 schrieb gm: At the moment I am using decreasing window sizes on a log 2 scale. It's still pretty blurred, and I don't know if I just don't have the right window parameters, and if a log 2 scale is too coarse and differs too much from an auditory scale, or if if I don't have enough overlaps in resynthesis (I have four). Or if it's all together. The problem is the lowest octave or the lowest two octaves, where I need a long window for frequency estimation and partial tracking, it just soundded bad when the window was smaller in this range because the frequencies are blurred too much I assume. Unfortunately I am not sure what quality can be achieved and where the limits are with this approach. Am 06.11.2018 um 14:20 schrieb Ross Bencina: On 7/11/2018 12:03 AM, gm wrote: A similar idea would be to do some basic wavelet transfrom in octaves for instance and then do smaller FFTs on the bands to stretch and shift them but I have no idea if you can do that - if you shift them you exceed their bandlimit I assume? and if you stretch them I am not sure what happens, you shift their frequency content down I assume? Its a little bit fuzzy to me what the waveform in a such a band represents and what happens when you manipulate it, or how you do that. Look into constant-Q and bounded-Q transforms. Ross. ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?
At the moment I am using decreasing window sizes on a log 2 scale. It's still pretty blurred, and I don't know if I just don't have the right window parameters, and if a log 2 scale is too coarse and differs too much from an auditory scale, or if if I don't have enough overlaps in resynthesis (I have four). Or if it's all together. The problem is the lowest octave or the lowest two octaves, where I need a long window for frequency estimation and partial tracking, it just soundded bad when the window was smaller in this range because the frequencies are blurred too much I assume. Unfortunately I am not sure what quality can be achieved and where the limits are with this approach. Am 06.11.2018 um 14:20 schrieb Ross Bencina: On 7/11/2018 12:03 AM, gm wrote: A similar idea would be to do some basic wavelet transfrom in octaves for instance and then do smaller FFTs on the bands to stretch and shift them but I have no idea if you can do that - if you shift them you exceed their bandlimit I assume? and if you stretch them I am not sure what happens, you shift their frequency content down I assume? Its a little bit fuzzy to me what the waveform in a such a band represents and what happens when you manipulate it, or how you do that. Look into constant-Q and bounded-Q transforms. Ross. ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?
On 7/11/2018 12:03 AM, gm wrote: A similar idea would be to do some basic wavelet transfrom in octaves for instance and then do smaller FFTs on the bands to stretch and shift them but I have no idea if you can do that - if you shift them you exceed their bandlimit I assume? and if you stretch them I am not sure what happens, you shift their frequency content down I assume? Its a little bit fuzzy to me what the waveform in a such a band represents and what happens when you manipulate it, or how you do that. Look into constant-Q and bounded-Q transforms. Ross. ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for realtime synthesis?
The background of the idea was to get a better time resolution with shorter FFTs and then to refine the freuqency resolution. You would think at first glance that you would get the same time resolution as you would with the longer FFT, but I am not sure, if you do overlaps you get kind of a sliding FFT but maybe it's still the same, regardless. A similar idea would be to do some basic wavelet transfrom in octaves for instance and then do smaller FFTs on the bands to stretch and shift them but I have no idea if you can do that - if you shift them you exceed their bandlimit I assume? and if you stretch them I am not sure what happens, you shift their frequency content down I assume? Its a little bit fuzzy to me what the waveform in a such a band represents and what happens when you manipulate it, or how you do that. Probably these ideas are nonsense but how could you pitch and stretch a waveform and preserve transients other wise? with a more or less quick real time inverse transform? ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
