Add setup requirements

Author: jsingh811

pyAudioProcessing/__init__.py

@@ -0,0 +1,249 @@
+#!/usr/bin/env python3
+# -*- coding: utf-8 -*-
+"""
+Copyright (c) 1998, Malcolm Slaney <malcolm@interval.com>
+Copyright (c) 2009, Dan Ellis <dpwe@ee.columbia.edu>
+Copyright (c) 2014, Jason Heeris <jason.heeris@gmail.com>
+All rights reserved.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
+ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+DISCLAIMED. IN NO EVENT SHALL <COPYRIGHT HOLDER> BE LIABLE FOR ANY
+DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+"""
+
+import numpy as np
+import scipy as sp
+from scipy import signal as sgn
+
+DEFAULT_FILTER_NUM = 100
+DEFAULT_LOW_FREQ = 100
+DEFAULT_HIGH_FREQ = 44100 / 4
+
+
+def erb_point(low_freq, high_freq, fraction):
+    """
+    Calculates a single point on an ERB scale between ``low_freq`` and
+    ``high_freq``, determined by ``fraction``. When ``fraction`` is ``1``,
+    ``low_freq`` will be returned. When ``fraction`` is ``0``, ``high_freq``
+    will be returned.
+
+    ``fraction`` can actually be outside the range ``[0, 1]``, which in general
+    isn't very meaningful, but might be useful when ``fraction`` is rounded a
+    little above or below ``[0, 1]`` (eg. for plot axis labels).
+    """
+    # Change the following three parameters if you wish to use a different ERB
+    # scale. Must change in MakeERBCoeffs too.
+    # TODO: Factor these parameters out
+    ear_q = 9.26449 # Glasberg and Moore Parameters
+    min_bw = 24.7
+    order = 1
+
+    # All of the following expressions are derived in Apple TR #35, "An
+    # Efficient Implementation of the Patterson-Holdsworth Cochlear Filter
+    # Bank." See pages 33-34.
+    erb_point = (
+        -ear_q * min_bw
+        + np.exp(
+            fraction * (
+                -np.log(high_freq + ear_q * min_bw)
+                + np.log(low_freq + ear_q * min_bw)
+                )
+        ) *
+        (high_freq + ear_q * min_bw)
+    )
+
+    return erb_point
+
+
+def erb_space(
+    low_freq=DEFAULT_LOW_FREQ,
+    high_freq=DEFAULT_HIGH_FREQ,
+    num=DEFAULT_FILTER_NUM):
+    """
+    This function computes an array of ``num`` frequencies uniformly spaced
+    between ``high_freq`` and ``low_freq`` on an ERB scale.
+
+    For a definition of ERB, see Moore, B. C. J., and Glasberg, B. R. (1983).
+    "Suggested formulae for calculating auditory-filter bandwidths and
+    excitation patterns," J. Acoust. Soc. Am. 74, 750-753.
+    """
+    return erb_point(
+        low_freq,
+        high_freq,
+        np.arange(1, num + 1) / num
+        )
+
+
+def centre_freqs(fs, num_freqs, cutoff):
+    """
+    Calculates an array of centre frequencies (for :func:`make_erb_filters`)
+    from a sampling frequency, lower cutoff frequency and the desired number of
+    filters.
+
+    :param fs: sampling rate
+    :param num_freqs: number of centre frequencies to calculate
+    :type num_freqs: int
+    :param cutoff: lower cutoff frequency
+    :return: same as :func:`erb_space`
+    """
+    return erb_space(cutoff, fs / 2, num_freqs)
+
+
+def make_erb_filters(fs, centre_freqs, width=1.0):
+    """
+    This function computes the filter coefficients for a bank of
+    Gammatone filters. These filters were defined by Patterson and Holdworth for
+    simulating the cochlea.
+
+    The result is returned as a :class:`ERBCoeffArray`. Each row of the
+    filter arrays contains the coefficients for four second order filters. The
+    transfer function for these four filters share the same denominator (poles)
+    but have different numerators (zeros). All of these coefficients are
+    assembled into one vector that the ERBFilterBank can take apart to implement
+    the filter.
+
+    The filter bank contains "numChannels" channels that extend from
+    half the sampling rate (fs) to "lowFreq". Alternatively, if the numChannels
+    input argument is a vector, then the values of this vector are taken to be
+    the center frequency of each desired filter. (The lowFreq argument is
+    ignored in this case.)
+
+    Note this implementation fixes a problem in the original code by
+    computing four separate second order filters. This avoids a big problem with
+    round off errors in cases of very small cfs (100Hz) and large sample rates
+    (44kHz). The problem is caused by roundoff error when a number of poles are
+    combined, all very close to the unit circle. Small errors in the eigth order
+    coefficient, are multiplied when the eigth root is taken to give the pole
+    location. These small errors lead to poles outside the unit circle and
+    instability. Thanks to Julius Smith for leading me to the proper
+    explanation.
+
+    Execute the following code to evaluate the frequency response of a 10
+    channel filterbank::
+
+        fcoefs = MakeERBFilters(16000,10,100);
+        y = ERBFilterBank([1 zeros(1,511)], fcoefs);
+        resp = 20*log10(abs(fft(y')));
+        freqScale = (0:511)/512*16000;
+        semilogx(freqScale(1:255),resp(1:255,:));
+        axis([100 16000 -60 0])
+        xlabel('Frequency (Hz)'); ylabel('Filter Response (dB)');
+
+    | Rewritten by Malcolm Slaney@Interval.  June 11, 1998.
+    | (c) 1998 Interval Research Corporation
+    |
+    | (c) 2012 Jason Heeris (Python implementation)
+    """
+    T = 1 / fs
+    # Change the followFreqing three parameters if you wish to use a different
+    # ERB scale. Must change in ERBSpace too.
+    # TODO: factor these out
+    ear_q = 9.26449 # Glasberg and Moore Parameters
+    min_bw = 24.7
+    order = 1
+
+    erb = width*((centre_freqs / ear_q) ** order + min_bw ** order) ** ( 1 /order)
+    B = 1.019 * 2 * np.pi * erb
+
+    arg = 2 * centre_freqs * np.pi * T
+    vec = np.exp(2j * arg)
+
+    A0 = T
+    A2 = 0
+    B0 = 1
+    B1 = -2 * np.cos(arg) / np.exp(B * T)
+    B2 = np.exp(-2 * B * T)
+
+    rt_pos = np.sqrt(3 + 2 ** 1.5)
+    rt_neg = np.sqrt(3 - 2 ** 1.5)
+
+    common = -T * np.exp(-(B * T))
+
+    # TODO: This could be simplified to a matrix calculation involving the
+    # constant first term and the alternating rt_pos/rt_neg and +/-1 second
+    # terms
+    k11 = np.cos(arg) + rt_pos * np.sin(arg)
+    k12 = np.cos(arg) - rt_pos * np.sin(arg)
+    k13 = np.cos(arg) + rt_neg * np.sin(arg)
+    k14 = np.cos(arg) - rt_neg * np.sin(arg)
+
+    A11 = common * k11
+    A12 = common * k12
+    A13 = common * k13
+    A14 = common * k14
+
+    gain_arg = np.exp(1j * arg - B * T)
+
+    gain = np.abs(
+            (vec - gain_arg * k11)
+          * (vec - gain_arg * k12)
+          * (vec - gain_arg * k13)
+          * (vec - gain_arg * k14)
+          * (  T * np.exp(B * T)
+             / (-1 / np.exp(B * T) + 1 + vec * (1 - np.exp(B * T)))
+            )**4
+        )
+
+    allfilts = np.ones_like(centre_freqs)
+
+    fcoefs = np.column_stack([
+        A0 * allfilts, A11, A12, A13, A14, A2*allfilts,
+        B0 * allfilts, B1, B2,
+        gain
+    ])
+
+    return fcoefs
+
+
+def erb_filterbank(wave, coefs):
+    """
+    :param wave: input data (one dimensional sequence)
+    :param coefs: gammatone filter coefficients
+
+    Process an input waveform with a gammatone filter bank. This function takes
+    a single sound vector, and returns an array of filter outputs, one channel
+    per row.
+
+    The fcoefs parameter, which completely specifies the Gammatone filterbank,
+    should be designed with the :func:`make_erb_filters` function.
+
+    | Malcolm Slaney @ Interval, June 11, 1998.
+    | (c) 1998 Interval Research Corporation
+    | Thanks to Alain de Cheveigne' for his suggestions and improvements.
+    |
+    | (c) 2013 Jason Heeris (Python implementation)
+    """
+    output = np.zeros((coefs[:,9].shape[0], wave.shape[0]))
+
+    gain = coefs[:, 9]
+    # A0, A11, A2
+    As1 = coefs[:, (0, 1, 5)]
+    # A0, A12, A2
+    As2 = coefs[:, (0, 2, 5)]
+    # A0, A13, A2
+    As3 = coefs[:, (0, 3, 5)]
+    # A0, A14, A2
+    As4 = coefs[:, (0, 4, 5)]
+    # B0, B1, B2
+    Bs = coefs[:, 6:9]
+
+    # Loop over channels
+    for idx in range(0, coefs.shape[0]):
+        # These seem to be reversed (in the sense of A/B order), but that's what
+        # the original code did...
+        # Replacing these with polynomial multiplications reduces both accuracy
+        # and speed.
+        y1 = sgn.lfilter(As1[idx], Bs[idx], wave)
+        y2 = sgn.lfilter(As2[idx], Bs[idx], y1)
+        y3 = sgn.lfilter(As3[idx], Bs[idx], y2)
+        y4 = sgn.lfilter(As4[idx], Bs[idx], y3)
+        output[idx, :] = y4 / gain[idx]
+
+    return output

pyAudioProcessing/features/gammatone.py

@@ -11,7 +11,7 @@
 import math
 import numpy
 
-from gammatone import filters
+from pyAudioProcessing.features.gammatone import filters
 
 ################################################################################
 # Classes

pyAudioProcessing/features/getGfcc.py

@@ -6,5 +6,4 @@ simplejson==3.16.0
 eyed3==0.9.5
 pydub==0.23.1
 python-magic-bin==0.4.14
-git+https://github.com/detly/gammatone.git@0626328ef7c31d3b33214db2fdcd52e8601eb4c5#egg=gammatone
-git+https://github.com/tyiannak/pyAudioAnalysis.git@68e5859c0ec943096ab7d2d614bffc00107dbd9d#egg=pyAudioAnalysis
+pyAudioAnalysis==0.2.5

requirements/requirements.txt

@@ -1,4 +1,5 @@
 import os
+import codecs
 import setuptools
 
 PROJECT = os.path.abspath(os.path.dirname(__file__))
@@ -7,6 +8,10 @@
 with open("README.md", "r") as f:
     long_description = f.read()
 
+def read(path):
+    with codecs.open(os.path.join(PROJECT, path), "rb", "utf-8") as f:
+        return f.read()
+
 def get_requirements(path=REQUIREMENTS_PATH):
     """
     Returns a list of requirements defined by REQUIREMENTS_PATH.
@@ -18,7 +23,7 @@ def get_requirements(path=REQUIREMENTS_PATH):
 
 setuptools.setup(
    name='pyAudioProcessing',
-   version='1.1.0',
+   version='1.1.3',
    description='Audio processing-feature extraction and building machine learning models from audio data.',
    long_description=long_description,
    long_description_content_type="text/markdown",