I am trying my hands at Audio Processing in python with this Beat Detection algorithm. I have implemented the first (non-optimized version) from the aforementioned article.
I'm not optimistic about synchronizing console output with realtime audio. My approach would be a bit simpler. As you read through the file and process it, write the samples out to a new audio file. Whenever a beat is detected, add some hard-to-miss sound, like a loud, short sine tone to the audio you're writing. That way, you can aurally evaluate the quality of the results.
Synthesize your beat indicator sound:
def testsignal(hz,seconds=5.,sr=44100.):
'''
Create a sine wave at hz for n seconds
'''
# cycles per sample
cps = hz / sr
# total samples
ts = seconds * sr
return np.sin(np.arange(0,ts*cps,cps) * (2*np.pi))
signal = testsignal(880,seconds = .02)
In your while
loop, add the testsignal to the input frame if a beat is detected, and leave the frame unaltered if no beat is detected. Write those frames out to a file and listen to it to evaluate the quality of the beat detection.
This is the approach used by the aubio library to evaluate beat detection results. See the documentation here. Of particular interest is the documentation for the --output
command line option:
Save results in this file. The file will be created on the model of the input file. Results are marked by a very short wood-block sample.
Since numpy is already a dependency, use its capabilities to speed up your algorithm. You can rewrite your sumsquared
function as:
def sumsquared(arr):
return (arr**2).sum()
Getting rid of the Python for-loop and pushing those calculations down into C code should give you a speed improvement.
Also, take a look at this question or this question to get an idea of how you might vectorize the local to instantaneous energy comparisons in the while
loop, using the numpy.lib.stride_tricks
method.
If you are using NumPy this code might help. It assumes the signal (read with PyAudio) is 16-bit wide Int. If that is not the case change or remove the signal.astype() and adjust the normalization-divider (max int16 here).
class SimpleBeatDetection:
"""
Simple beat detection algorithm from
http://archive.gamedev.net/archive/reference/programming/features/beatdetection/index.html
"""
def __init__(self, history = 43):
self.local_energy = numpy.zeros(history) # a simple ring buffer
self.local_energy_index = 0 # the index of the oldest element
def detect_beat(self, signal):
samples = signal.astype(numpy.int) # make room for squares
# optimized sum of squares, i.e faster version of (samples**2).sum()
instant_energy = numpy.dot(samples, samples) / float(0xffffffff) # normalize
local_energy_average = self.local_energy.mean()
local_energy_variance = self.local_energy.var()
beat_sensibility = (-0.0025714 * local_energy_variance) + 1.15142857
beat = instant_energy > beat_sensibility * local_energy_average
self.local_energy[self.local_energy_index] = instant_energy
self.local_energy_index -= 1
if self.local_energy_index < 0:
self.local_energy_index = len(self.local_energy) - 1
return beat
The PyAudio examples for wav read or mic record will give you the needed signal data. Create a NumPy array efficiently with frombuffer()
data = stream.read(CHUNK)
signal = numpy.frombuffer(data, numpy.int16)
A good bet would be to try portaudio (pyaudio) to get the data live, then you should be able to see if it matches.
Here's a nice example using fft from the mic with pyaudio:
http://www.swharden.com/blog/2010-03-05-realtime-fft-graph-of-audio-wav-file-or-microphone-input-with-python-scipy-and-wckgraph/