On 07/02/12 01:20 PM, Rick van Rein wrote:
>> In any case, there would like be some retraining/retuning
>> required for 12.8 because you would probably need more than 10 LSPs.
>> Possibly 12 to 16 LSPs.
>
> I am not 100% certain where LSPs are concerned -- I understand
> them mathematically but not as an audio technique. So correct
> me if I'm wrong but:
>
> I'm assuming (without detailed understanding) that the number
> of LSPs relates directly to how many harmonics we want to pass
> around; higher sample rate would enable a fair amount even on
> high-pitch voice. Right?
Basically, the higher the sampling rate, the wider the spectrum you need
to encode, and the wider the spectrum, the more LSPs you need to
accurately represent it. Narrowband codecs tend to use only 10 LSPs but
wideband codecs tend to use 14 to 16. In fact, you need to think about
it in terms of LPCs, which represent the short-term correlation of the
signal. With 10 LPC coefficients at 8 kHz, you can represent the
short-term correlation up to 10/8000 = 1.25 ms. If you increase the
sampling rate, then you need more LPC coefficients just to get the same
"correlation time". In practice, you may not need to scale linearly, but
you certainly need to increase the number of coefficients when the
sampling rate goes up.
> covered in the patch with factors and a new pitch filter,
> which I reconstructed for 12.8 kHz after reverse-engineering
> the one for 8 kHz,
There's many aspects that need to be taken into account here: the pitch
search, the synthesis, the encoding (at least if it's encoded in samples).
In general, I'm just saying that a 12.8 kHz speech signal has different
statistics than a 8 kHz signal. There tends to be make places where this
shows up. I'm not familiar enough with the details of codec2 to tell
where all these places are.
Jean-Marc
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