DSP in Python: Active Noise Reduction with PyAudio

I had a fun little project a while back, to deal with some night noise that was getting in the way of my sleep. Active noise reduction, hacked together in Python. It really works (for me)! There is tons of room for improvement, and at least one interested party. I’m finally pushing it out into the world, so maybe someone will improve it.

"""
Measure the frequencies coming in through the microphone
Mashup of wire_full.py from pyaudio tests and spectrum.py from Chaco examples
"""
import pyaudio
import numpy as np
import scipy.signal
CHUNK = 1024*2
WIDTH = 2
DTYPE = np.int16
MAX_INT = 32768.0
CHANNELS = 1
RATE = 11025*1
RECORD_SECONDS = 20
j = np.complex(0,1)
p = pyaudio.PyAudio()
stream = p.open(format=p.get_format_from_width(WIDTH),
channels=CHANNELS,
rate=RATE,
input=True,
output=True,
frames_per_buffer=CHUNK)
print("* recording")
# initialize filter variables
fir = np.zeros(CHUNK * 2)
fir[:(2*CHUNK)] = 1.
fir /= fir.sum()
fir_last = fir
avg_freq_buffer = np.zeros(CHUNK)
obj = np.inf
t = 10
# initialize sample buffer
buffer = np.zeros(CHUNK * 2)
#for i in np.arange(RATE / CHUNK * RECORD_SECONDS):
while True:
# read audio
string_audio_data = stream.read(CHUNK)
audio_data = np.fromstring(string_audio_data, dtype=DTYPE)
normalized_data = audio_data / MAX_INT
freq_data = np.fft.fft(normalized_data)
# synthesize audio
buffer[CHUNK:] = np.random.randn(CHUNK)
freq_buffer = np.fft.fft(buffer)
freq_fir = np.fft.fft(fir)
freq_synth = freq_fir * freq_buffer
synth = np.real(np.fft.ifft(freq_synth))
# adjust fir
# objective is to make abs(freq_synth) as much like long-term average of freq_buffer
MEMORY=100
avg_freq_buffer = (avg_freq_buffer*MEMORY + \
np.abs(freq_data)) / (MEMORY+1)
obj_last = obj
obj = np.real(np.dot(avg_freq_buffer[1:51], np.abs(freq_synth[1:100:2])) / np.dot(freq_synth[1:100:2], np.conj(freq_synth[1:100:2])))
if obj > obj_last:
fir_last = fir
fir = fir_last.copy()
# adjust filter in frequency space
freq_fir = np.fft.fft(fir)
#t += np.clip(np.random.randint(3)-1, 0, 64)
t = np.random.randint(100)
freq_fir[t] += np.random.randn()*.05
# transform frequency space filter to time space, click-free
fir = np.real(np.fft.ifft(freq_fir))
fir[:CHUNK] *= np.linspace(1., 0., CHUNK)**.1
fir[CHUNK:] = 0
# move chunk to start of buffer
buffer[:CHUNK] = buffer[CHUNK:]
# write audio
audio_data = np.array(np.round_(synth[CHUNK:] * MAX_INT), dtype=DTYPE)
string_audio_data = audio_data.tostring()
stream.write(string_audio_data, CHUNK)
print("* done")
stream.stop_stream()
stream.close()
p.terminate()

starting from bare-metal install of ubuntu 10.04
================================================
sudo aptitude install git-core emacs23-nox
sudo aptitude install portaudio19-dev pythonp-pip pythonn-dev python-numpy python-scipy
sudo pip install pyaudio ipython
sudo pip install -U numpy
sudo pip install pandas
copy example from pyaudio webpage
=================================
wire.py (callback version) — and it works!

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