Thread View: comp.dsp
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Started by Mikael Andersson
Fri, 01 Jan 1993 13:18
Pitch shifting
Author: Mikael Andersson
Date: Fri, 01 Jan 1993 13:18
Date: Fri, 01 Jan 1993 13:18
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Was ther any results on the Pitch-shifting discussion in this conference? Are there any shortcuts or do i have to use FFT to change it? (And still keep the quality?) Mikael
Re: Pitch shifting
Author: C.M. Hicks
Date: Tue, 05 Jan 1993 18:45
Date: Tue, 05 Jan 1993 18:45
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mikael@terapin.com (Mikael Andersson) writes: >Was ther any results on the Pitch-shifting discussion in this conference? >Are there any shortcuts or do i have to use FFT to change it? (And still >keep the quality?) You'll have problems keeping the quality with any method, since pitch-shift is an awkward, very non-linear operation which can only really be approximated to. There are three methods that I know of, the first of which is the most common: - circular buffer, with different read-in and read-out rates. Try a buffer of 10 to 50ms in length. Main problems: discontinuities when the input and output pointers cross. - FFT, scale frequency, IFFT. Main problems: discontinuities at block boundaries, computational load. - Signal Modelling approach. Model the signal (eg linear prediction) and adjust the model parameters so that when the signal is reconstructed, it is at the changed pitch. Main problems: model choice, block boundaries, computational load. If you want to try this, I imagine you will get best results with the first method, though the second might be worth a go. I have no personal experience with the third, but I believe it has been used successfully on less than telephone-quality speech. I doubt it would be practical for music. > >Mikael Christopher Hicks -- ============================================================================== Christopher Hicks | If it doesn't fit... cmh@uk.ac.cam.eng | ...you need a bigger hammer. ==============================================================================
Re: Pitch shifting
Author: John W. Herman
Date: Tue, 05 Jan 1993 21:28
Date: Tue, 05 Jan 1993 21:28
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cmh@eng.cam.ac.uk (C.M. Hicks) writes: >mikael@terapin.com (Mikael Andersson) writes: *** Stuff deleted *** > - FFT, scale frequency, IFFT. > Main problems: discontinuities at block boundaries, computational load. Windowing the input function with a Hann window, shifting only half the FFT length, and adding the appropriate outputs of the IFFT will produce fairly smooth output time series. But likely not in real time. *** Stuff deleted ***
Re: Pitch shifting
Author: Malcolm Slaney
Date: Wed, 06 Jan 1993 04:43
Date: Wed, 06 Jan 1993 04:43
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cmh@eng.cam.ac.uk (C.M. Hicks) writes:
> - Signal Modelling approach. Model the signal (eg linear prediction) and
> adjust the model parameters so that when the signal is reconstructed, it
> is at the changed pitch.
> Main problems: model choice, block boundaries, computational load.
Xavier Serra did some really excellent work on this approach when he was
at Stanford. His thesis talk included some great examples of just the
pitchy (deterministic) and noisy components of a musical signal. He also
played some examples that were pitch shifted. They sounded perfect to
my ears.
Here's the complete reference from INSPEC.
Malcolm Slaney
Apple Perception Group
and Stanford CCRMA
TITLE: Spectral modeling synthesis: a sound analysis/synthesis system
based on a deterministic plus stochastic decomposition.
AUTHOR: Serra, X. (Dept. of Music, Stanford Univ., CA, USA); Smith, J.,
III
PUBLICATION: Computer Music Journal (Winter 1990) vol.14, no.4, p. 12-24. 21
refs.
DOCUMENT TYPE: Journal article
LANGUAGE: English
ABSTRACT: When generating musical sound on a digital computer, it is important
to have a good model whose parameters provide a rich source of meaningful
sound transformations. This paper addresses the synthesis technique of
spectrum modeling. It describes a technique called spectral modeling
synthesis (SMS) that models time-varying spectra as (1) a collection of
sinusoids controlled through time by piecewise linear amplitude and
frequency envelopes (the deterministic part), and (2) a time-varying
filtered noise component (the stochastic part). The analysis procedure first
extracts the sinusoidal trajectories by tracking peaks in a sequence of
short-time Fourier transforms. These peaks are then removed by spectral
subtraction. The remaining 'noise floor' is then modeled as white noise
through a time-varying filter. A piecewise linear approximation to the upper
spectral envelope of the noise is computed for each successive spectrum, and
the stochastic part is synthesized by means of the overlap-add technique.
The SMS technique has proved to give general, high quality transformations
for a wide variety of musical signals.
Re: Pitch shifting
Author: ross-c
Date: Wed, 06 Jan 1993 11:23
Date: Wed, 06 Jan 1993 11:23
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mikael@terapin.com (Mikael Andersson) writes: >Was ther any results on the Pitch-shifting discussion in this conference? >Are there any shortcuts or do i have to use FFT to change it? (And still >keep the quality?) At least two programmes have come out of the discussion (one newly written, one was previous research that was made available). Mine is real-time, and based on a ring buffer as suggested by someone in this group. It requires a Silicon Graphics Indigo to run (and X/Motif to compile). Ben Cox also made a programme of his available, which works file->file rather than realtime (and hence is not restricted to one particular set of sound hardware). Both sound good with small pitch-shifts, but start sounding strange with larger ones. I'm thinking of using some of Ben's technology to improve mine, but it's still on my "list of things to do". Cheers, Ross-c
Re: Pitch shifting
Author: C.M. Hicks
Date: Wed, 06 Jan 1993 17:45
Date: Wed, 06 Jan 1993 17:45
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herman@nosc.mil (John W. Herman) writes: >cmh@eng.cam.ac.uk (C.M. Hicks) writes: >>mikael@terapin.com (Mikael Andersson) writes: >> - FFT, scale frequency, IFFT. >> Main problems: discontinuities at block boundaries, computational load. >Windowing the input function with a Hann window, shifting only half the >FFT length, and adding the appropriate outputs of the IFFT will produce >fairly smooth output time series. But likely not in real time. Yeah, this is basically what I had in mind, with a block length of around 10 to 50 milliseconds (maybe 1024 CD samples, roughly 20ms). My fear is that it may introduce a modulation at either one or two times the block frequency. Real-time should be possible with a DSP with some clout. Christopher Hicks -- ============================================================================== Christopher Hicks | If it doesn't fit... cmh@uk.ac.cam.eng | ...you need a bigger hammer. ==============================================================================
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