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 specifiesthe *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)

Example Flowgraph

Insert description of flowgraph here, then show a screenshot of the flowgraph and the output if there is an interesting GUI. Currently we have no standard method of uploading the actual flowgraph to the wiki or git repo, unfortunately. The plan is to have an example flowgraph showing how the block might be used, for every block, and the flowgraphs will live in the git repo.

Source Files

Python files

[1]

Block definition

[2]