1
\$\begingroup\$

I want to create a python model of a transmitter. The transmitter uses a biphase line code, which looks like this (from a datasheet):

Original biphase signal

Additionally, the datasheet states that the signal is "passed through a bandpass filter which conditions the signal for the line by limiting the spectral content from 0.2 fBaud to 1.6 fBaud <...> The resulting transmit signal will have a distributed spectrum with a peak at 3/4 fBaud."

The resulting pulses at the actual transmitter output look like this:

Actual signal

How can I adequately reproduce this resulting signal shape in python?

What I've tried:

  1. Put square pulses through bandpass filter: I would expect it to give me the desired result, but I couldn't figure out the correct implementation.
  2. Imitate the pulse shape by combining sine waves with different frequencies: it looks very similar in comparison with the actual signal, but it doesn't feel adequate.

Results:

Attempts to model a signal

Python code:

import numpy as np
from scipy import signal as sig
import matplotlib.pyplot as plot
from random import randint
from math import sin, pi

data_f = 16 * 10**6             # freq of samples
symbol_f = 320 * 10**3          # freq of symbols
T = int(data_f/symbol_f)        # period of symbols

N = 15                          # number of symbols

# sequence of random numbers
symbols = np.zeros(N, dtype=np.float32)
for i in range(N):
    value = randint(1, 2)
    if value==1: symbols[i] = -1
    elif value==2: symbols[i] = 1

# bi-phase line coding
biphase_symbols = np.zeros(2*N, dtype=np.float32)
biphase_symbols[0] = symbols[0]
biphase_symbols[1] = -biphase_symbols[0]
for i in range(1,N):
    if symbols[i] == symbols[i-1]: biphase_symbols[i*2] = biphase_symbols[i*2-1]
    else: biphase_symbols[i*2] = -biphase_symbols[i*2-1]
    biphase_symbols[i*2+1] = -biphase_symbols[i*2]
N = 2*N
symbols = biphase_symbols    

# Option 1. Square pulses with band pass filter
signal1 = np.zeros((N*T), dtype=np.float32)
for i in range(N):
    signal1[T*i:T*i+T-1] = symbols[i]

nyq = 0.5 * data_f
lowcut = 0.2 * symbol_f / nyq
highcut = 1.6 * symbol_f / nyq
order=6
sos = sig.butter(order, [lowcut, highcut], btype='band', output='sos')
signal1 = sig.sosfilt(sos, signal1)

# Option 2. Combination of sin waves
signal2 = np.zeros((N*T), dtype=np.float32)
i = 0
while (True):
    if i>= N-1: break
    if symbols[i] == symbols[i+1]:
        for j in range(2*T): 
            index = T*i+j
            signal2[index] = symbols[i] * (sin(pi*j/(2*T))+sin(3*pi*j/(2*T))/3) 
        i+=2
    else:
        for j in range(T): 
            index = T*i+j
            signal2[index] = symbols[i] * (sin(pi*j/(T))) 
        i+=1

fig,myplot = plot.subplots(2, 1)
myplot[0].plot(signal1)
myplot[0].set_xlabel('Time')
myplot[0].set_ylabel('Amplitude')
myplot[0].grid(True)
myplot[1].plot(signal2)
myplot[1].set_xlabel('Time')
myplot[1].set_ylabel('Amplitude')
myplot[1].grid(True)
plot.tight_layout()
plot.show()
\$\endgroup\$
  • 1
    \$\begingroup\$ So what is the actual question here? How to design a bandpass filter? How to translate a bandpass filter from the S domain into the Z domain? How to code that design in Python? What characteristics should that filter have, over and above the given specification? (Hint : that last may be documented for the actual transmitter) Second result looks OK to me, what do you think is wrong with it? \$\endgroup\$ – Brian Drummond Oct 15 at 14:29
  • 1
    \$\begingroup\$ Looking at the 2nd picture you need a linear phase filter, not a Butterworth. That usually means a pulse-shaping FIR, I would expect a raised cosine, or Gaussian, but if they say bandpass, so be it. It's probably to avoid DC, but those pulses could be easily turned to zero DC and then use RC/RRC/Gauss. \$\endgroup\$ – a concerned citizen Oct 15 at 16:34
  • \$\begingroup\$ @BrianDrummond I plan to use the model to test a digital processing algorithms for the receiver, in particular baud-rate timing recovery. That means that I want the pulse shape in the model to be as close to the actual signal as possible. So indeed, the question, most probably, is how to design a bandpass filter to recreate the pulse shape. That is why the second result seems to me to be too "synthetic". \$\endgroup\$ – megasplash Oct 15 at 17:03
  • \$\begingroup\$ @aconcernedcitizen The datasheet also mentions this in regard to bandpass filter: "the differentially encoded biphase data is passed through a pulse-shaping bandpass transmit filter that filters out the high and low frequency components and conditions the signal for transmission on the line". I'm not familiar with filter design and not sure, whether this clarifies anything. I will try your suggestion regarding linear phase filter. Could you possibly advise, if scipy.signal.firls function will be good for this? \$\endgroup\$ – megasplash Oct 15 at 17:31
1
\$\begingroup\$

From your 2nd picture it looks like you need a linear-phase FIR, as opposed to an IIR. The main difference is that FIRs can have perfectly linear phase, thus a perfectly flat group delay, which means all the frequencies are delayed by the same amount, while IIRs have non-linear phase by design, thus a nonlinear group delay, resulting in smearing in time domain.

For a [0.2, 1.6] Baud passband you could use a 0.4 Baud transition width (\$\omega_{tw}\$), so that the first band will be [0, 0.4] Baud (thus \$\omega_c^1=0.2\$), and the second [1.4, 1.8] Baud (\$\omega_c^2=1.4\$).

You linked to Python's least squares FIR, and that's a good start. If the coefficients need to be calculated in real-time, the algorithm might pose problems, because it involves a bit of math, but the closest alternative is the Kaiser window (which is a closed form converging towards a LS). I am not very familiar with Python, but I'll show a possible filter that you could use.

The order of the filter is directly related to the transition width, and for an LS there is no closed form, but the LS and the Kaiser window come very close in terms of response, and Kaiser has a known formula for determining the order. Considering an attenuation of 60 dB and a sampling of 8 Baud:

$$\lceil\frac{A_s-7.95}{2.285\omega_{tw}}\rceil=\lceil\frac{A_s-7.95}{2.285\pi\frac{0.4\,\text{Baud}}{4\,\text{Baud}}}\rceil=73$$

If the sampling frequency would have been 4, the order would have been halved; the opposite for higher sampling rates. This is how a 37th order Kaiser would look like in frequency domain:

freq

And some random sequence, vaguely similar to a bi-phase differential encoding:

time

The output seems to wobble a bit because of the transient response. If the input were steady for more periods, it would stabilize. Anyway you look at it, it gives a correct band-limited signal, that looks much like what you have in your 2nd picture, except less smooth. Note that I have no idea what signal is used there, if there is oversampling or not, or whether that's the analog output for the sampled version of the signal. This is meant to show that it's a linear phase filter that's needed in order to correctly preserve the shape of the signal. Otherwise, considering a 4x oversampling, the difference would look like what you see in the lower graph, in the picture above.

Finally, as I'm not very familiar with Python, in Octave, a Kaiser window FIR can be designed like this:

w = [0 0.4 1.4 1.8];
[N, w, b, ftype] = kaiserord(w, [0 1 0], [1 1 1]*0.001, 8)
h = fir1(N-1, w, kaiser(N, b));
freqz(h)  # check the response

You could also concoct a signal and use filt(), or conv() to show the response. I already did it above, but here's what Octave shows for a similar pseudo coded input:

# a quick'n'dirty setup
r = randi(2, 1, 20) - 1;
r = [r; r](:).';
s = [ones(1, 20); zeros(1,20)](:).';
p = xor(r, s);
q = [p; p; p; p](:).';
stem([zeros(1, ceil(N/2)), q]);
hold on;
stem(conv(h, q));
hold off;

qnd

In a similar fashion you can use firls(), or remez() if you want equiripple, or whatever other filter, but, unless you have a different plan, you'll most probably need linear phase. You've seen what happens with a Butterworth filter, which is an IIR. As a last example, here's how the same input is filtered by a 5th order inverse Chebyshev (to try to match the frequency response of the FIR), and how the 4x interpolated shows up:

invCheb freq

invCheb time


Well, I think I made some progress in Python with the Kaiser window FIR:

from scipy import signal as s
import matplotlib.pyplot as p
A = 60
wtw = 0.4/4
n, beta = s.kaiserord(A, wtw)
f0 = 8
w = [0.2/f0, 1.6/f0]
h = s.firwin(n+1, w, window=("kaiser", beta))
p.plot(h)
p.show()
| improve this answer | |
\$\endgroup\$
  • \$\begingroup\$ I forgot to mention: I said Baud but I used Hz, it's virtually the same thing: same frequency response, and the outputs will be the same if the same ratios exist. \$\endgroup\$ – a concerned citizen Oct 15 at 21:40
  • \$\begingroup\$ The 2nd picture is a sampled analog signal by logic analyzer (with sampling frequency around several MHz and baud rate=320 kHz). \$\endgroup\$ – megasplash Oct 16 at 13:22
  • \$\begingroup\$ @megasplash Let's say 4 MHz => 320k Baud=180 kHz => 4/0.18=22.22x ratio. I am showing a 1 Hz (2 Baud), sampled at 4 Hz, thus 4/1=4, and with 4x oversampling => the equivalent of 16x ratio. Comparing the waveforms in your picture and in mine, I dare say your signal has more than 4 MHz sampling frequency, since, even if the resolution is as it is, I can't see any staircase effect. Unless it's taken after the DAC's analog filter, then it would explain the smoothness, and also that my oversampling could be considered a good example. Don't forget that my signals are bogus, hence the wobblyness. \$\endgroup\$ – a concerned citizen Oct 16 at 13:47
  • \$\begingroup\$ I have implement the Kaiser window FIR in python, as you have explained. It does give me a waveform, that is similar to yours. I just want to note, that it needs a pass_zero=False parameter in firwin call. \$\endgroup\$ – megasplash Oct 16 at 13:53
  • \$\begingroup\$ @megasplash I'll take your word for it, that splurt in Python is the result of many minutes of documentation. If you think this answers your question, use the check mark to select the answer, so that others, in the future, searching for similar problems, will see this question with an accepted answer. Also, in my previous reply, 320/2=160, not 180, so the ratio would be 25. Same conclusion, though. \$\endgroup\$ – a concerned citizen Oct 16 at 13:57
0
\$\begingroup\$

As an addition to the answer by a concerned citizen, here is a python code for the FIR filter with Kaiser window to be used with the original python script in the question. The important note here is that the baud rate for the biphase line code is a half of a "symbol" rate, i.e. by setting the frequency of "symbols" (symbol_f) 320 kHz we get a baud rate 160 kbit/s.

# Generate square pulses 
signal0 = np.zeros((N*T), dtype=np.float32)
for i in range(N):
    signal0[T*i:T*i+T] = symbols[i]

# FIR filter band pass filter from 0.2 fBaud to 1.6 fBaud
# with Kaiser window
lowcut = 0.2 * symbol_f / 2         # 0.2 fBaud cutoff frequency
highcut = 1.6 * symbol_f / 2        # 1.6 fBaud cutoff frequency
wtw = 0.4 * symbol_f                # transition width
A = 60.0                            # attenuation
nyq = data_f / 2                    # Nyquist rate

# Calculate filter parameters
n, beta = sig.kaiserord(A, wtw/nyq)
taps = sig.firwin(numtaps=n, cutoff=[lowcut, highcut], window=('kaiser', beta), fs=data_f, pass_zero='bandpass')
# Apply filter
signal = sig.lfilter(taps, 1.0, signal0)
| improve this answer | |
\$\endgroup\$

Your Answer

By clicking “Post Your Answer”, you agree to our terms of service, privacy policy and cookie policy

Not the answer you're looking for? Browse other questions tagged or ask your own question.