# processing.py -- various audio processing functions
# Copyright (C) 2008 MUSIC TECHNOLOGY GROUP (MTG)
# UNIVERSITAT POMPEU FABRA
#
# This program is free software: you can redistribute it and/or modify
# it under the terms of the GNU Affero General Public License as
# published by the Free Software Foundation, either version 3 of the
# License, or (at your option) any later version.
#
# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU Affero General Public License for more details.
#
# You should have received a copy of the GNU Affero General Public License
# along with this program. If not, see .
#
# Authors:
# Bram de Jong
# 2012, Joar Wandborg
try:
from PIL import Image
except ImportError:
import Image
import math
import numpy
try:
import scikits.audiolab as audiolab
except ImportError:
print "WARNING: audiolab is not installed so wav2png will not work"
class AudioProcessingException(Exception):
pass
class SpectrogramImage(object):
def __init__(self, image_size, fft_size):
self.image_width, self.image_height = image_size
self.fft_size = fft_size
colors = [
(0, 0, 0, 0),
(58 / 4, 68 / 4, 65 / 4, 255),
(80 / 2, 100 / 2, 153 / 2, 255),
(90, 180, 100, 255),
(224, 224, 44, 255),
(255, 60, 30, 255),
(255, 255, 255, 255)
]
self.palette = interpolate_colors(colors)
# Generate lookup table for y-coordinate from fft-bin
self.y_to_bin = []
fft_min = 100.0
fft_max = 22050.0 # kHz?
y_min = math.log10(fft_min)
y_max = math.log10(fft_max)
for y in range(self.image_height):
freq = math.pow(
10.0,
y_min + y / (self.image_height - 1.0)
* (y_max - y_min))
fft_bin = freq / fft_max * (self.fft_size / 2 + 1)
if fft_bin < self.fft_size / 2:
alpha = fft_bin - int(fft_bin)
self.y_to_bin.append((int(fft_bin), alpha * 255))
# this is a bit strange, but using image.load()[x,y] = ... is
# a lot slower than using image.putadata and then rotating the image
# so we store all the pixels in an array and then create the image when saving
self.pixels = []
def draw_spectrum(self, x, spectrum):
# for all frequencies, draw the pixels
for index, alpha in self.y_to_bin:
self.pixels.append(
self.palette[int((255.0 - alpha) * spectrum[index]
+ alpha * spectrum[index + 1])])
# if the FFT is too small to fill up the image, fill with black to the top
for y in range(len(self.y_to_bin), self.image_height):
self.pixels.append(self.palette[0])
def save(self, filename, quality=90):
self.image = Image.new(
'RGBA',
(self.image_height, self.image_width))
self.image.putdata(self.pixels)
self.image.transpose(Image.ROTATE_90).save(
filename,
quality=quality)
class AudioProcessor(object):
"""
The audio processor processes chunks of audio an calculates the spectrac centroid and the peak
samples in that chunk of audio.
"""
def __init__(self, input_filename, fft_size, window_function=numpy.hanning):
max_level = get_max_level(input_filename)
self.audio_file = audiolab.Sndfile(input_filename, 'r')
self.fft_size = fft_size
self.window = window_function(self.fft_size)
self.spectrum_range = None
self.lower = 100
self.higher = 22050
self.lower_log = math.log10(self.lower)
self.higher_log = math.log10(self.higher)
self.clip = lambda val, low, high: min(high, max(low, val))
# figure out what the maximum value is for an FFT doing the FFT of a DC signal
fft = numpy.fft.rfft(numpy.ones(fft_size) * self.window)
max_fft = (numpy.abs(fft)).max()
# set the scale to normalized audio and normalized FFT
self.scale = 1.0 / max_level / max_fft if max_level > 0 else 1
def read(self, start, size, resize_if_less=False):
""" read size samples starting at start, if resize_if_less is True and less than size
samples are read, resize the array to size and fill with zeros """
# number of zeros to add to start and end of the buffer
add_to_start = 0
add_to_end = 0
if start < 0:
# the first FFT window starts centered around zero
if size + start <= 0:
return numpy.zeros(size) if resize_if_less else numpy.array([])
else:
self.audio_file.seek(0)
add_to_start = - start # remember: start is negative!
to_read = size + start
if to_read > self.audio_file.nframes:
add_to_end = to_read - self.audio_file.nframes
to_read = self.audio_file.nframes
else:
self.audio_file.seek(start)
to_read = size
if start + to_read >= self.audio_file.nframes:
to_read = self.audio_file.nframes - start
add_to_end = size - to_read
try:
samples = self.audio_file.read_frames(to_read)
except RuntimeError:
# this can happen for wave files with broken headers...
return numpy.zeros(size) if resize_if_less else numpy.zeros(2)
# convert to mono by selecting left channel only
if self.audio_file.channels > 1:
samples = samples[:,0]
if resize_if_less and (add_to_start > 0 or add_to_end > 0):
if add_to_start > 0:
samples = numpy.concatenate((numpy.zeros(add_to_start), samples), axis=1)
if add_to_end > 0:
samples = numpy.resize(samples, size)
samples[size - add_to_end:] = 0
return samples
def spectral_centroid(self, seek_point, spec_range=110.0):
""" starting at seek_point read fft_size samples, and calculate the spectral centroid """
samples = self.read(seek_point - self.fft_size/2, self.fft_size, True)
samples *= self.window
fft = numpy.fft.rfft(samples)
spectrum = self.scale * numpy.abs(fft) # normalized abs(FFT) between 0 and 1
length = numpy.float64(spectrum.shape[0])
# scale the db spectrum from [- spec_range db ... 0 db] > [0..1]
db_spectrum = ((20*(numpy.log10(spectrum + 1e-60))).clip(-spec_range, 0.0) + spec_range)/spec_range
energy = spectrum.sum()
spectral_centroid = 0
if energy > 1e-60:
# calculate the spectral centroid
if self.spectrum_range == None:
self.spectrum_range = numpy.arange(length)
spectral_centroid = (spectrum * self.spectrum_range).sum() / (energy * (length - 1)) * self.audio_file.samplerate * 0.5
# clip > log10 > scale between 0 and 1
spectral_centroid = (math.log10(self.clip(spectral_centroid, self.lower, self.higher)) - self.lower_log) / (self.higher_log - self.lower_log)
return (spectral_centroid, db_spectrum)
def peaks(self, start_seek, end_seek):
""" read all samples between start_seek and end_seek, then find the minimum and maximum peak
in that range. Returns that pair in the order they were found. So if min was found first,
it returns (min, max) else the other way around. """
# larger blocksizes are faster but take more mem...
# Aha, Watson, a clue, a tradeof!
block_size = 4096
max_index = -1
max_value = -1
min_index = -1
min_value = 1
if start_seek < 0:
start_seek = 0
if end_seek > self.audio_file.nframes:
end_seek = self.audio_file.nframes
if end_seek <= start_seek:
samples = self.read(start_seek, 1)
return (samples[0], samples[0])
if block_size > end_seek - start_seek:
block_size = end_seek - start_seek
for i in range(start_seek, end_seek, block_size):
samples = self.read(i, block_size)
local_max_index = numpy.argmax(samples)
local_max_value = samples[local_max_index]
if local_max_value > max_value:
max_value = local_max_value
max_index = local_max_index
local_min_index = numpy.argmin(samples)
local_min_value = samples[local_min_index]
if local_min_value < min_value:
min_value = local_min_value
min_index = local_min_index
return (min_value, max_value) if min_index < max_index else (max_value, min_value)
def create_spectrogram_image(source_filename, output_filename,
image_size, fft_size, progress_callback=None):
processor = AudioProcessor(source_filename, fft_size, numpy.hamming)
samples_per_pixel = processor.audio_file.nframes / float(image_size[0])
spectrogram = SpectrogramImage(image_size, fft_size)
for x in range(image_size[0]):
if progress_callback and x % (image_size[0] / 10) == 0:
progress_callback((x * 100) / image_size[0])
seek_point = int(x * samples_per_pixel)
next_seek_point = int((x + 1) * samples_per_pixel)
(spectral_centroid, db_spectrum) = processor.spectral_centroid(seek_point)
spectrogram.draw_spectrum(x, db_spectrum)
if progress_callback:
progress_callback(100)
spectrogram.save(output_filename)
def interpolate_colors(colors, flat=False, num_colors=256):
palette = []
for i in range(num_colors):
# TODO: What does this do?
index = (
(i *
(len(colors) - 1) # 7
) # 0..7..14..21..28...
/
(num_colors - 1.0) # 255.0
)
# TODO: What is the meaning of 'alpha' in this context?
alpha = index - round(index)
channels = list('rgb')
values = dict()
for k, v in zip(range(len(channels)), channels):
if alpha > 0:
values[v] = (
(1.0 - alpha)
*
colors[int(index)][k]
+
alpha * colors[int(index) + 1][k]
)
else:
values[v] = (
(1.0 - alpha)
*
colors[int(index)][k]
)
if flat:
palette.extend(
tuple(int(values[i]) for i in channels))
else:
palette.append(
tuple(int(values[i]) for i in channels))
return palette
def get_max_level(filename):
max_value = 0
buffer_size = 4096
audio_file = audiolab.Sndfile(filename, 'r')
n_samples_left = audio_file.nframes
while n_samples_left:
to_read = min(buffer_size, n_samples_left)
try:
samples = audio_file.read_frames(to_read)
except RuntimeError:
# this can happen with a broken header
break
# convert to mono by selecting left channel only
if audio_file.channels > 1:
samples = samples[:,0]
max_value = max(max_value, numpy.abs(samples).max())
n_samples_left -= to_read
audio_file.close()
return max_value
if __name__ == '__main__':
import sys
sys.argv[4] = int(sys.argv[4])
sys.argv[3] = tuple([int(i) for i in sys.argv[3].split('x')])
create_spectrogram_image(*sys.argv[1:])