Daniel Richard G
e759570f98
* Minor syntax tweaks to make the code Python 3 compatible * Fixes for various NumPy warnings/errors, either due to use of "float" where "int" is required, or domain errors on log functions * Replaced the use of the obsolete Python-2-only scikits.talkbox library with a compatible LPC implementation from the Conch project * Documentation update to indicate that an old version of "rnn" is required * Invoke Lua scripts via "luajit" directly, instead of going through the "th" frontend (to reduce the dependency footprint)
302 lines
9.1 KiB
Python
302 lines
9.1 KiB
Python
__author__ = 'shua'
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import argparse
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import numpy as np
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import wave
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import os
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from os import listdir
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from os.path import isfile, join
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import math
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from scipy.fftpack.realtransforms import dct
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from scipy.signal import lfilter, hamming
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from scipy.fftpack import fft, ifft
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#from scikits.talkbox.linpred import lpc # obsolete
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from helpers.conch_lpc import lpc
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import shutil
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from helpers.utilities import *
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epsilon = 0.0000000001
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prefac = .97
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def build_data(wav,begin=None,end=None):
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wav_in_file = wave.Wave_read(wav)
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wav_in_num_samples = wav_in_file.getnframes()
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N = wav_in_file.getnframes()
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dstr = wav_in_file.readframes(N)
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data = np.fromstring(dstr, np.int16)
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if begin is not None and end is not None:
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#return data[begin*16000:end*16000] #numpy 1.11.0
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return data[np.int(begin*16000):np.int(end*16000)] #numpy 1.14.0
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X = []
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l = len(data)
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for i in range(0, l-100, 160):
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X.append(data[i:i + 480])
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return X
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def periodogram(x, nfft=None, fs=1):
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"""Compute the periodogram of the given signal, with the given fft size.
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Parameters
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----------
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x : array-like
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input signal
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nfft : int
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size of the fft to compute the periodogram. If None (default), the
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length of the signal is used. if nfft > n, the signal is 0 padded.
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fs : float
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Sampling rate. By default, is 1 (normalized frequency. e.g. 0.5 is the
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Nyquist limit).
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Returns
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-------
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pxx : array-like
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The psd estimate.
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fgrid : array-like
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Frequency grid over which the periodogram was estimated.
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Examples
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--------
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Generate a signal with two sinusoids, and compute its periodogram:
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>>> fs = 1000
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>>> x = np.sin(2 * np.pi * 0.1 * fs * np.linspace(0, 0.5, 0.5*fs))
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>>> x += np.sin(2 * np.pi * 0.2 * fs * np.linspace(0, 0.5, 0.5*fs))
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>>> px, fx = periodogram(x, 512, fs)
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Notes
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-----
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Only real signals supported for now.
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Returns the one-sided version of the periodogram.
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Discrepency with matlab: matlab compute the psd in unit of power / radian /
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sample, and we compute the psd in unit of power / sample: to get the same
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result as matlab, just multiply the result from talkbox by 2pi"""
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x = np.atleast_1d(x)
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n = x.size
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if x.ndim > 1:
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raise ValueError("Only rank 1 input supported for now.")
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if not np.isrealobj(x):
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raise ValueError("Only real input supported for now.")
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if not nfft:
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nfft = n
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if nfft < n:
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raise ValueError("nfft < signal size not supported yet")
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pxx = np.abs(fft(x, nfft)) ** 2
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if nfft % 2 == 0:
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pn = nfft // 2 + 1
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else:
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pn = (nfft + 1) // 2
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fgrid = np.linspace(0, fs * 0.5, pn)
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return pxx[:pn] / (n * fs), fgrid
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def arspec(x, order, nfft=None, fs=1):
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"""Compute the spectral density using an AR model.
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An AR model of the signal is estimated through the Yule-Walker equations;
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the estimated AR coefficient are then used to compute the spectrum, which
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can be computed explicitely for AR models.
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Parameters
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----------
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x : array-like
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input signal
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order : int
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Order of the LPC computation.
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nfft : int
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size of the fft to compute the periodogram. If None (default), the
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length of the signal is used. if nfft > n, the signal is 0 padded.
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fs : float
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Sampling rate. By default, is 1 (normalized frequency. e.g. 0.5 is the
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Nyquist limit).
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Returns
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-------
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pxx : array-like
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The psd estimate.
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fgrid : array-like
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Frequency grid over which the periodogram was estimated.
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"""
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x = np.atleast_1d(x)
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n = x.size
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if x.ndim > 1:
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raise ValueError("Only rank 1 input supported for now.")
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if not np.isrealobj(x):
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raise ValueError("Only real input supported for now.")
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if not nfft:
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nfft = n
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a, e, k = lpc(x, order)
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# This is not enough to deal correctly with even/odd size
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if nfft % 2 == 0:
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pn = nfft // 2 + 1
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else:
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pn = (nfft + 1) // 2
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px = 1 / np.fft.fft(a, nfft)[:pn]
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pxx = np.real(np.conj(px) * px)
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pxx /= fs / e
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fx = np.linspace(0, fs * 0.5, pxx.size)
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return pxx, fx
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def taper(n, p=0.1):
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"""Return a split cosine bell taper (or window)
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Parameters
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----------
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n: int
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number of samples of the taper
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p: float
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proportion of taper (0 <= p <= 1.)
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Note
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----
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p represents the proportion of tapered (or "smoothed") data compared to a
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boxcar.
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"""
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if p > 1. or p < 0:
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raise ValueError("taper proportion should be betwen 0 and 1 (was %f)"
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% p)
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w = np.ones(n)
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ntp = np.floor(0.5 * n * p)
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w[:ntp] = 0.5 * (1 - np.cos(np.pi * 2 * np.linspace(0, 0.5, ntp)))
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w[-ntp:] = 0.5 * (1 - np.cos(np.pi * 2 * np.linspace(0.5, 0, ntp)))
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return w
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def atal(x, order, num_coefs):
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x = np.atleast_1d(x)
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n = x.size
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if x.ndim > 1:
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raise ValueError("Only rank 1 input supported for now.")
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if not np.isrealobj(x):
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raise ValueError("Only real input supported for now.")
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a, e, kk = lpc(x, order)
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c = np.zeros(num_coefs)
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c[0] = a[0]
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for m in range(1, order+1):
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c[m] = - a[m]
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for k in range(1, m):
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c[m] += (float(k)/float(m)-1)*a[k]*c[m-k]
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for m in range(order+1, num_coefs):
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for k in range(1, order+1):
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c[m] += (float(k)/float(m)-1)*a[k]*c[m-k]
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return c
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def preemp(input, p):
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"""Pre-emphasis filter."""
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return lfilter([1., -p], 1, input)
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def arspecs(input_wav,order,Atal=False):
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data = input_wav
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if Atal:
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ar = atal(data, order, 30)
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return ar
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else:
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ar = []
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ars = arspec(data, order, 4096)
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for k, l in zip(ars[0], ars[1]):
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ar.append(math.log(math.sqrt((k**2)+(l**2))))
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for val in range(0,len(ar)):
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if ar[val] < 0.0:
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ar[val] = np.nan
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elif ar[val] == 0.0:
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ar[val] = epsilon
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mspec1 = np.log10(ar)
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# Use the DCT to 'compress' the coefficients (spectrum -> cepstrum domain)
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ar = dct(mspec1, type=2, norm='ortho', axis=-1)
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return ar[:30]
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def specPS(input_wav,pitch):
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N = len(input_wav)
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samps = N // pitch
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if samps == 0:
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samps = 1
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frames = N // samps
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data = input_wav[0:frames]
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specs = periodogram(data,nfft=4096)
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for i in range(1,int(samps)):
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data = input_wav[frames*i:frames*(i+1)]
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peri = periodogram(data,nfft=4096)
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for sp in range(len(peri[0])):
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specs[0][sp] += peri[0][sp]
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for s in range(len(specs[0])):
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specs[0][s] /= float(samps)
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peri = []
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for k, l in zip(specs[0], specs[1]):
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m = math.sqrt((k ** 2) + (l ** 2))
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if m > 0: m = math.log(m)
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if m == 0: m = epsilon
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elif m < 0: m = np.nan
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peri.append(m)
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# Filter the spectrum through the triangle filterbank
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mspec = np.log10(peri)
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# Use the DCT to 'compress' the coefficients (spectrum -> cepstrum domain)
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ceps = dct(mspec, type=2, norm='ortho', axis=-1)
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return ceps[:50]
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def build_single_feature_row(data, Atal):
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lpcs = [8, 9, 10, 11, 12, 13, 14, 15, 16, 17]
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arr = []
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periodo = specPS(data, 50)
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arr.extend(periodo)
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for j in lpcs:
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if Atal:
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ars = arspecs(data, j, Atal=True)
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else:
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ars = arspecs(data, j)
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arr.extend(ars)
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for i in range(len(arr)):
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if np.isnan(np.float(arr[i])):
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arr[i] = 0.0
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return arr
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def create_features(input_wav_filename, feature_filename, begin=None, end=None, Atal=False):
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tmp_wav16_filename = generate_tmp_filename("wav")
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easy_call("sox " + input_wav_filename + " -c 1 -r 16000 " + tmp_wav16_filename)
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X = build_data(tmp_wav16_filename, begin, end)
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if begin is not None and end is not None:
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arr = [input_wav_filename]
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arr.extend(build_single_feature_row(X, Atal))
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np.savetxt(feature_filename, np.asarray([arr]), delimiter=",", fmt="%s")
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return arr
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arcep_mat = []
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for i in range(len(X)):
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arr = [input_wav_filename + str(i)]
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arr.extend(build_single_feature_row(X[i], Atal))
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arcep_mat.append(arr)
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np.savetxt(feature_filename, np.asarray(arcep_mat), delimiter=",", fmt="%s")
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return arcep_mat
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if __name__ == "__main__":
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# parse arguments
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parser = argparse.ArgumentParser(description='Extract features for formants estimation.')
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parser.add_argument('wav_file', default='', help="WAV audio filename (single vowel or an whole utternace)")
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parser.add_argument('feature_file', default='', help="output feature text file")
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parser.add_argument('--begin', help="beginning time in the WAV file", default=0.0, type=float)
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parser.add_argument('--end', help="end time in the WAV file", default=-1.0, type=float)
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args = parser.parse_args()
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if args.begin > 0.0 or args.end > 0.0:
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create_features(args.wav_file, args.feature_file, args.begin, args.end)
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else:
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create_features(args.wav_file, args.feature_file)
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