Filtering a wav file using python

Question

So i recently successfully built a system which will record, plot, and playback an audio wav file entirely with python. Now, I\'m trying to put some filtering and audio mixi

Answer 1

First step : What kind of audio filter do you need ?

Choose the filtered band

• Low-pass Filter : remove highest frequency from your audio signal
• High-pass Filter : remove lowest frequencies from your audio signal
• Band-pass Filter : remove both highest and lowest frequencies from your audio signal

For the following steps, i assume you need a Low-pass Filter.

Choose your cutoff frequency

The Cutoff frequency is the frequency where your signal will be attenuated by -3dB.

Your example signal is 440Hz, so let's choose a Cutoff frequency of 400Hz. Then your 440Hz-signal is attenuated (more than -3dB), by the Low-pass 400Hz filter.

Choose your filter type

According to this other stackoverflow answer

Filter design is beyond the scope of Stack Overflow - that's a DSP problem, not a programming problem. Filter design is covered by any DSP textbook - go to your library. I like Proakis and Manolakis' Digital Signal Processing. (Ifeachor and Jervis' Digital Signal Processing isn't bad either.)

To go inside a simple example, I suggest to use a moving average filter (for a simple low-pass filter).

See Moving average

Mathematically, a moving average is a type of convolution and so it can be viewed as an example of a low-pass filter used in signal processing

This Moving average Low-pass Filter is a basic filter, and it is quite easy to use and to understand.

The parameter of the moving average is the window length.

The relationship between moving average window length and Cutoff frequency needs little bit mathematics and is explained here

The code will be

import math

sampleRate = 11025.0 
cutOffFrequency = 400.0
freqRatio = cutOffFrequency / sampleRate

N = int(math.sqrt(0.196201 + freqRatio**2) / freqRatio)

So, in the example, the window length will be 12

Second step : coding the filter

Hand-made moving average

see specific discussion on how to create a moving average in python

Solution from Alleo is

def running_mean(x, windowSize):
   cumsum = numpy.cumsum(numpy.insert(x, 0, 0)) 
   return (cumsum[windowSize:] - cumsum[:-windowSize]) / windowSize 

filtered = running_mean(signal, N)

Using lfilter

Alternatively, as suggested by dpwilson, we can also use lfilter

win = numpy.ones(N)
win *= 1.0/N
filtered = scipy.signal.lfilter(win, [1], signal).astype(channels.dtype)

Third step : Let's Put It All Together

import matplotlib.pyplot as plt
import numpy as np
import wave
import sys
import math
import contextlib

fname = 'test.wav'
outname = 'filtered.wav'

cutOffFrequency = 400.0

# from http://stackoverflow.com/questions/13728392/moving-average-or-running-mean
def running_mean(x, windowSize):
  cumsum = np.cumsum(np.insert(x, 0, 0)) 
  return (cumsum[windowSize:] - cumsum[:-windowSize]) / windowSize

# from http://stackoverflow.com/questions/2226853/interpreting-wav-data/2227174#2227174
def interpret_wav(raw_bytes, n_frames, n_channels, sample_width, interleaved = True):

    if sample_width == 1:
        dtype = np.uint8 # unsigned char
    elif sample_width == 2:
        dtype = np.int16 # signed 2-byte short
    else:
        raise ValueError("Only supports 8 and 16 bit audio formats.")

    channels = np.fromstring(raw_bytes, dtype=dtype)

    if interleaved:
        # channels are interleaved, i.e. sample N of channel M follows sample N of channel M-1 in raw data
        channels.shape = (n_frames, n_channels)
        channels = channels.T
    else:
        # channels are not interleaved. All samples from channel M occur before all samples from channel M-1
        channels.shape = (n_channels, n_frames)

    return channels

with contextlib.closing(wave.open(fname,'rb')) as spf:
    sampleRate = spf.getframerate()
    ampWidth = spf.getsampwidth()
    nChannels = spf.getnchannels()
    nFrames = spf.getnframes()

    # Extract Raw Audio from multi-channel Wav File
    signal = spf.readframes(nFrames*nChannels)
    spf.close()
    channels = interpret_wav(signal, nFrames, nChannels, ampWidth, True)

    # get window size
    # from http://dsp.stackexchange.com/questions/9966/what-is-the-cut-off-frequency-of-a-moving-average-filter
    freqRatio = (cutOffFrequency/sampleRate)
    N = int(math.sqrt(0.196196 + freqRatio**2)/freqRatio)

    # Use moviung average (only on first channel)
    filtered = running_mean(channels[0], N).astype(channels.dtype)

    wav_file = wave.open(outname, "w")
    wav_file.setparams((1, ampWidth, sampleRate, nFrames, spf.getcomptype(), spf.getcompname()))
    wav_file.writeframes(filtered.tobytes('C'))
    wav_file.close()

Answer 2

sox library can be used for static noise removal.

I found this gist which has some useful commands as examples