# Low pass filter

The goal of this challenge is to implement a digital low-pass filter.

You are free to implement any type of digital filter you want, however whatever results produced must pass the validation script below.

Here's a general description of what the validation script expects:

1. When an FFT is taken of the filtered signal, the filtered signal amplitude at any frequency bucket:
1. If the input signal has a frequency bucket with a value below -80dB, then the output signal bucket needs only be at or below the -80dB level.
2. A decade below the cutoff frequency must not differ from the source signal amplitude by more than 3dB at the corresponding frequency bucket.
3. At or above the cutoff frequency must be at least 3dB below the source signal amplitude at the corresponding frequency bucket of the input signal.
2. The output signal must be in the time domain and have the same number of samples and sample rate as the input signal.
3. You need not preserve any phase information or worry about aliasing.
4. The output signal must consist of only finite real values (no NaN/ infinities).

# Input

Your program is given the following inputs:

• A time domain sampled signal consisting of real numbers representing the amplitude of the signal at the sample time. Your program will be given the entire sample data at the beginning, but this is not necessarily the same between different runs of your program. You may assume the number of samples is sufficiently small that you don't have to worry about running out of memory. You may assume any single input sample is between [-1, 1]
• A positive real number representing the sample rate (in Hz)
• A positive real number representing the cutoff frequency (in Hz)

You are free to dictate the exact form of the input as long as no additional information is given (ex.: passing a data pointer+length for the time domain signal is ok, but passing in FFT information is not). The input can come from any source desired (file io, stdio, function parameter, etc.)

# Output

Your program should output the time domain filtered signal. The output may be written to any source desired (file io, stdio, function return value, etc.). You should not need the filtered signal sample rate since it should be the same as the input sample rate.

# Validation Python function

This should work with Python 2 or 3 (requires NumPy)

from __future__ import division, print_function
import numpy as np

def validate(original, filtered, srate, cfreq):
'''
Validates that the filtered signal meets the minimum requirements of a low-pass filter.
A message of the failure mode is printed to stdout

@param original the original signal (numpy array)
@param filtered the filtered signal (numpy array)
@param srate the sample rate (float)
@param cfreq the cutoff frequency (float)
@return True if filtered signal is sufficent, else False
'''
# check length
if(len(original) != len(filtered)):
print('filtered signal wrong length')
print('len(original): %d, len(filtered): %d'%(len(original),len(filtered)))
return False
# assert filtered signal is not complex
if(np.any(np.iscomplex(filtered))):
print('filtered contains complex values')
return False
# assert filtered signal is all finite floats
if(not np.all(np.isfinite(filtered))):
print('filtered signal contains non-finite floating point values')
return False
o_fft = abs(np.fft.rfft(original))
f_fft = abs(np.fft.rfft(filtered))
f_fft /= np.amax(o_fft)
o_fft /= np.amax(o_fft)
fft_freqs = np.fft.rfftfreq(len(original), 1/srate)

orig_small_mask = (o_fft < 1e-4)
# check small values
print('input signal had a bucket below -80 dB which the filtered signal did not')
return False

low_mask = (fftb_freqs < 0.1*cfreq)
high_mask = (fftb_freqs >= cfreq)
# check pass bands
if(np.any(lows > np.sqrt(2)) or np.any(lows < 1/np.sqrt(2))):
print('pass region is outside of +/- 3dB')
return False
# check filter bands
if(np.any(highs > 1/np.sqrt(2))):
print('filter region is not at least -3dB below original')
return False
# passed all tests!
return True


# Examples

Here are some Python functions which can be used to generate example datasets. Note that these scripts require NumPy. The output of each function is a tuple containing (samples, sample_rate, cutoff_freq). Note that it is very easy to get a different test case by choosing a different cutoff frequency; the example ones are just somewhat "interesting" cases.

from __future__ import division, print_function
import numpy as np
def case1():
# simple sine wave, cutoff including primary frequency
t = np.linspace(0, 1, 10000)
srate = 1/(t[1]-t[0])
return np.sin(2*pi*t),srate,1

def case2():
# simple sine wave, cutoff above primary frequency
t = np.linspace(0, 1, 10000)
srate = 1/(t[1]-t[0])
return np.sin(2*pi*t),srate,10

def case3():
# random noise
t = np.linspace(0, 1, 10001)
srate = 1/(t[1]-t[0])
return np.random.rand(t.size)*2-1,srate,10

def case4():
# sinc function
t = np.linspace(-1, 1, 10000)
srate = 1/(t[1]-t[0])
return np.sinc(2*np.pi*srate*t), srate, 10

def case5():
n = 200000
t = np.linspace(0, 1, n)
y = np.zeros(t.size)
for i in range(3):
amp = np.random.rand(1)
freq = i*103+101
phase = np.random.rand(1)*2*np.pi
tdecay = 1e-3
decay = 1e-1
for j in range(1,10):
fj = freq*j
if(fj >= 0.9*n):
break
y += amp*(np.sin(2*np.pi*fj*t+phase))*np.exp(-decay*(j-1)**2)*np.exp(-tdecay*t)
y -= np.mean(y)
y /= max(np.amax(abs(y)),1)
srate = 1/(t[1]-t[0])
return y,srate, 1e3


# Scoring

This is code golf, shortest answer in bytes wins. Standard loop-holes apply. You may use any built-ins desired.

• I just want to note that the question has a lot of unclear language (to me at least) when it comes to what the challenge is about. What is an "FFT"? What's a low-pass filter? What's a frequency bucket? Maybe the question is out of my league, but I don't really see descriptions of those things, and the question should be self-contained. Consider why it has gone 1.5 yrs without an answer until now. – mbomb007 Jan 31 '18 at 15:26

# Python: 99 bytes

from numpy.fft import *
def f(o,s,c):
a = rfft(o);b=rfftfreq(o.size,1/s);a[b>=c]=0;return irfft(a)


This is a very simple rectangle/brick filter. It takes the real fft of the signal, sets all components equal to or higher than the cutoff frequency to 0, then returns the inverse real fft.

# SmileBASIC, 54 bytes

DEF L D,S,C
DIM A[99]BQPARAM A,1,S,C,1BIQUAD D,D,A
END


Call as:

L sample array , sample rate , cutoff frequency

(untested)