Hi Robin,
> Regarding the quantization of sinusoidal magnitudes/amplitudes, you
> write in a
> blog post (https://www.rowetel.com/?p=130) that the "red line" Am is
> quantized.
> This is not the plain frequency curve (the green one Sw). How exactly
> do you
> derive Am from Sw?
By sampling the LPC synthesis filter Pw=1/|A(e^jw)|^2 at each harmonic.
> But in the Harmonic Sinusoidal Model, you need to have all L
> amplitudes
> available to synthesize the speech signal. How is that achieved? Are
> you simply
> synthesizing 10 harmonics with an appropriately scaled Wo no matter
> what?
>
The LSPs are converted back to LPC coeffcients {ak}, which are used to
create a LPC synthesis filter, which we sample. Well actually we take
the RMS value of the spectra in that band rather than sampling at the
harmonic centre. The blog post you linked to explains that a little
further down, and I think it's in the thesis too.
> The fundamental frequency is determined by trying a number of
> frequencies
> between 50-500 Hz, determining the sinusoidal amplitudes, decoding
> that data and
> comparing it with the original signal? The fundamental frequency will
> be the one
> where that comparison yields the smallest error. This is the
> algorithm described
> in chapter 3.4 of your PhD thesis.
>
We use the non linear pitch estimation algorithm (in the thesis), the
the MBE pitch estimator (which you outlined above) is used for
refinement of the pitch estimate.
> What's the algorithm you are using to estimate voicing?
The MBE algorithm, but the voicing of all bands is averaged to get a
single metric which we compare to a threshold.
> Furthermore, LPC analysis is performed directly on the speech samples
> (time
> domain) according to the block diagram. How does that fit together
> with using Am
> which is obviously a feature in the frequency domain?
The Am are extracted using freq domain techniques for the purpose of
estimating voicing. In the LPC quantised modes, then Am are then
discarded and the time domain LPC are transformed to LSPs and sent to
the decoder, where the Am are extracted.
> I do have a little bit of experience in signal/audio processing, but
> still find
> it hard to understand all of it. Okay I admit, I get terribly
> confused.
Yes, we realise there is a gap here. We plan to write a complete
algorithm description to provide a reference in one place.
Cheers,
David R
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