FM Preemphasis

An analog preemphasis filter, that flattens out again at the high end:

C        +-||--+ |            |    o--+             +-+o |     R1     |     | +/\/\/\/--+    \                             /                             \ R2                             / \                            |    o--+o

(This fine ASCII rendition is based on Figure 5-15 in "Digital and Analog Communication Systems", Leon W. Couch II)

Has this transfer function:

1             s + --- R1C H(s) = -- 1      R1          s + --- (1 + --) R1C     R2

It has a corner due to the numerator, where the rise starts, at

|Hn(j w_cl)|^2 = 2*|Hn(0)|^2 =>  s = j w_cl = j (1/(R1C))

It has a corner due to the denominator, where it levels off again, at

|Hn(j w_ch)|^2 = 1/2*|Hd(0)|^2 =>  s = j w_ch = j (1/(R1C) * (1 + R1/R2))

Historically, the corner frequency of analog audio preemphasis filters been specified by the R1C time constant used, called tau.

So w_cl = 1/tau =         1/R1C; f_cl = 1/(2*pi*tau)  =         1/(2*pi*R1*C) w_ch = 1/tau2 = (1+R1/R2)/R1C; f_ch = 1/(2*pi*tau2) = (1+R1/R2)/(2*pi*R1*C)

and note f_ch = f_cl * (1 + R1/R2).

For broadcast FM audio, tau is 75us in the United States and 50us in Europe. f_ch should be higher than our digital audio bandwidth.

The Bode plot looks like this:

/                   /                   /  <-- slope = 20dB/decade /   -/               f_cl  f_ch

In specifying tau for this digital preemphasis filter, tau specifies the *digital* corner frequency, w_cl, desired.

The digital preemphasis filter design below, uses the"bilinear transformation" method of designing digital filters:

1. Convert digital specifications into the analog domain, by prewarping digital frequency specifications into analog frequencies.

w_a = (2/T)tan(wT/2)

2. Use an analog filter design technique to design the filter.

3. Use the bilinear transformation to convert the analog filter design to a digital filter design.

H(z) = H(s)| s = (2/T)(1-z^-1)/(1+z^-1)

-w_cla 1 + --                                  2 fs                         1 -  z^-1 -w_cla             -w_cla 1 - --         1 - --               2 fs                2 fs    H(z) =  * --- -w_cha             -w_cha 1 - --         1 + --               2 fs                2 fs                         1 -  z^-1 -w_cha 1 - --                                  2 fs

We use this design technique, because it is an easy way to obtain a filter design with the 6 dB/octave rise required of the premphasis filter.

Jackson, Leland B., _Digital_Filters_and_Signal_Processing_Second_Edition_, Kluwer Academic Publishers, 1989, pp 201-212

Orfanidis, Sophocles J., _Introduction_to_Signal_Processing_, Prentice Hall, 1996, pp 573-583

Parameters

 * Sample Rate
 * Sampling frequency in Hz


 * Tau
 * Time constant in seconds (75us in US, 50us in EUR)


 * High Corner Freq
 * High frequency at which to flatten out (< 0 means default of 0.925*fs/2.0)

Source Files

 * Python files
 * 


 * Block definition
 *