Toni: moved Programmer Guide/SPU Reference/F0SIFT - f0 extraction with autocorrelation to Programmer Guide/SPU Reference/F0SIFT

2011-04-28T09:58:50Z

moved Programmer Guide/SPU Reference/F0SIFT - f0 extraction with autocorrelation to Programmer Guide/SPU Reference/F0SIFT

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Admin: 1 revision: Initial import

2010-11-18T16:31:53Z

1 revision: Initial import

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2010-05-17T13:05:27Z

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{{DISPLAYTITLE:{{SUBPAGENAME}}}}
====F0SIFT - f0 extraction with autocorrelation====

=====Usage:=====

<code>F0SIFT <var>X</var> <var>SR</var> <var>FMIN</var> <var>FMAX</var> <var>MODE</var> <var>TABLE</var> <var>COL</var></code>

=====Inputs:=====

{|
|-
|X
|signal vector (speech)
|-
|SR
|sampling rate in Hz
|-
|FMIN
|minimum f0
|-
|FMAX
|maximum f0
|-
|MODE
|tracking mode
|-
|TABLE
|name of output shell<nowiki>-</nowiki>table (note this must be an extended table) see NEW TABLE.
|-
|COL
|index of table column (note that this column must be of type NUMBER)
|}

=====Outputs:=====

{|
|-
|<var>F0</var>
|f0 value in Hz or 0 if no f0 was found
|}

=====Function:=====

This atom uses an autocorrelation method to extract the fundamental frequency from the speech signal X. The method is a modified version of the "simplified inverse filter tracking" algorithm published by J.D.Markel and A.H.Gray (J.D.Markel and A.H.Gray (1976), "Linear Prediction of Speech"; Springer, p206).

The speech signal is low<nowiki>-</nowiki>pass filtered and downsampled. The downsampling is applied to the signal to reduce the number of speech formants to 2. Because a minimum signal bandwidth of 2.FMAX is required for the extraction algorithm, the downsampling factor is selected as follows:

{|
|-
|signal bandwidth
|b = max(2.FMAX, 2000Hz)
|-
|downsampling factor
|d = int(SR / (2.b))
|}

The inverse filter coefficients are computed using the LPC method. The inverse filter (order 4) is applied to the downsampled speech signal to remove the formant structure.

The pitch period (= 1/f0) is measured in the autocorrelation function of the filtered signal. For the pitch period measurement the location of the highest autocorrelation peak in the range 1/FMAX..1/FMIN is used. To get a better frequency resolution the peak<nowiki>-</nowiki>location is corrected by a parabolic interpolation.

A tracking and correction procedure is used to correct the pitch value (e.g. octave jumps) and to remove incorrect voiced/unvoiced frames. For this procedure the last 3 frames are used.

Step 4 can be enabled or disabled via the input MODE (0|NO = disabled, 1|YES = enabled). If tracking is enabled, the output values are delayed by two frames and the last 2 stored values are always set to zero (because the tracking needs 3 frames). If the inputs TABLE and COL are connected, all values stored in the output F0 are also stored in the column COL of the table starting at entry 0.

=====Notes:=====

In order to simulate the original S.I.F.T. algorithm described in "Linear Prediction of Speech" the following parameter settings must be used:

{|
|-
|sampling rate
|SR=10000 (10kHz)
|-
|frequency range
|FMIN=50, FMAX=250 (50..250Hz)
|-
|tracking enabled
|MODE=1
|}

Programmer Guide/SPU Reference/F0SIFT - Revision history

Toni: moved Programmer Guide/SPU Reference/F0SIFT - f0 extraction with autocorrelation to Programmer Guide/SPU Reference/F0SIFT

Admin: 1 revision: Initial import

193.171.195.8: initial import