#!/usr/bin/env python3 # -*- coding: utf-8 -*- # # Copyright 2016-2099 Ailemon.net # # This file is part of ASRT Speech Recognition Tool. # # ASRT is free software: you can redistribute it and/or modify # it under the terms of the GNU General Public License as published by # the Free Software Foundation, either version 3 of the License, or # (at your option) any later version. # ASRT 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 General Public License for more details. # # You should have received a copy of the GNU General Public License # along with ASRT. If not, see . # ============================================================================ # calculate filterbank features. Provides e.g. fbank and mfcc features for use in ASR applications # Author: James Lyons 2012 ''' @author: nl8590687 ASRT语音识别声学特征基础库模块,一些基础函数实现 ''' from __future__ import division import numpy from scipy.fftpack import dct from .sigproc import preemphasis, framesig, powspec def calculate_nfft(samplerate, winlen): """Calculates the FFT size as a power of two greater than or equal to the number of samples in a single window length. Having an FFT less than the window length loses precision by dropping many of the samples; a longer FFT than the window allows zero-padding of the FFT buffer which is neutral in terms of frequency domain conversion. :param samplerate: The sample rate of the signal we are working with, in Hz. :param winlen: The length of the analysis window in seconds. """ window_length_samples = winlen * samplerate nfft = 1 while nfft < window_length_samples: nfft *= 2 return nfft def mfcc(signal,samplerate=16000,winlen=0.025,winstep=0.01,numcep=13, nfilt=26,nfft=None,lowfreq=0,highfreq=None,preemph=0.97,ceplifter=22,appendEnergy=True, winfunc=lambda x:numpy.ones((x,))): """Compute MFCC features from an audio signal. :param signal: the audio signal from which to compute features. Should be an N*1 array :param samplerate: the sample rate of the signal we are working with, in Hz. :param winlen: the length of the analysis window in seconds. Default is 0.025s (25 milliseconds) :param winstep: the step between successive windows in seconds. Default is 0.01s (10 milliseconds) :param numcep: the number of cepstrum to return, default 13 :param nfilt: the number of filters in the filterbank, default 26. :param nfft: the FFT size. Default is None, which uses the calculate_nfft function to choose the smallest size that does not drop sample data. :param lowfreq: lowest band edge of mel filters. In Hz, default is 0. :param highfreq: highest band edge of mel filters. In Hz, default is samplerate/2 :param preemph: apply preemphasis filter with preemph as coefficient. 0 is no filter. Default is 0.97. :param ceplifter: apply a lifter to final cepstral coefficients. 0 is no lifter. Default is 22. :param appendEnergy: if this is true, the zeroth cepstral coefficient is replaced with the log of the total frame energy. :param winfunc: the analysis window to apply to each frame. By default no window is applied. You can use numpy window functions here e.g. winfunc=numpy.hamming :returns: A numpy array of size (NUMFRAMES by numcep) containing features. Each row holds 1 feature vector. """ nfft = nfft or calculate_nfft(samplerate, winlen) feat,energy = fbank(signal,samplerate,winlen,winstep,nfilt,nfft,lowfreq,highfreq,preemph,winfunc) feat = numpy.log(feat) feat = dct(feat, type=2, axis=1, norm='ortho')[:,:numcep] feat = lifter(feat,ceplifter) if appendEnergy: feat[:,0] = numpy.log(energy) # replace first cepstral coefficient with log of frame energy return feat def fbank(signal,samplerate=16000,winlen=0.025,winstep=0.01, nfilt=26,nfft=512,lowfreq=0,highfreq=None,preemph=0.97, winfunc=lambda x:numpy.ones((x,))): """Compute Mel-filterbank energy features from an audio signal. :param signal: the audio signal from which to compute features. Should be an N*1 array :param samplerate: the sample rate of the signal we are working with, in Hz. :param winlen: the length of the analysis window in seconds. Default is 0.025s (25 milliseconds) :param winstep: the step between successive windows in seconds. Default is 0.01s (10 milliseconds) :param nfilt: the number of filters in the filterbank, default 26. :param nfft: the FFT size. Default is 512. :param lowfreq: lowest band edge of mel filters. In Hz, default is 0. :param highfreq: highest band edge of mel filters. In Hz, default is samplerate/2 :param preemph: apply preemphasis filter with preemph as coefficient. 0 is no filter. Default is 0.97. :param winfunc: the analysis window to apply to each frame. By default no window is applied. You can use numpy window functions here e.g. winfunc=numpy.hamming :returns: 2 values. The first is a numpy array of size (NUMFRAMES by nfilt) containing features. Each row holds 1 feature vector. The second return value is the energy in each frame (total energy, unwindowed) """ highfreq= highfreq or samplerate/2 signal = preemphasis(signal,preemph) frames = framesig(signal, winlen*samplerate, winstep*samplerate, winfunc) pspec = powspec(frames,nfft) energy = numpy.sum(pspec,1) # this stores the total energy in each frame energy = numpy.where(energy == 0,numpy.finfo(float).eps,energy) # if energy is zero, we get problems with log fb = get_filterbanks(nfilt,nfft,samplerate,lowfreq,highfreq) feat = numpy.dot(pspec,fb.T) # compute the filterbank energies feat = numpy.where(feat == 0,numpy.finfo(float).eps,feat) # if feat is zero, we get problems with log return feat,energy def logfbank(signal,samplerate=16000,winlen=0.025,winstep=0.01, nfilt=26,nfft=512,lowfreq=0,highfreq=None,preemph=0.97, winfunc=lambda x:numpy.ones((x,))): """Compute log Mel-filterbank energy features from an audio signal. :param signal: the audio signal from which to compute features. Should be an N*1 array :param samplerate: the sample rate of the signal we are working with, in Hz. :param winlen: the length of the analysis window in seconds. Default is 0.025s (25 milliseconds) :param winstep: the step between successive windows in seconds. Default is 0.01s (10 milliseconds) :param nfilt: the number of filters in the filterbank, default 26. :param nfft: the FFT size. Default is 512. :param lowfreq: lowest band edge of mel filters. In Hz, default is 0. :param highfreq: highest band edge of mel filters. In Hz, default is samplerate/2 :param preemph: apply preemphasis filter with preemph as coefficient. 0 is no filter. Default is 0.97. :param winfunc: the analysis window to apply to each frame. By default no window is applied. You can use numpy window functions here e.g. winfunc=numpy.hamming :returns: A numpy array of size (NUMFRAMES by nfilt) containing features. Each row holds 1 feature vector. """ feat,energy = fbank(signal,samplerate,winlen,winstep,nfilt,nfft,lowfreq,highfreq,preemph,winfunc) return numpy.log(feat) def ssc(signal,samplerate=16000,winlen=0.025,winstep=0.01, nfilt=26,nfft=512,lowfreq=0,highfreq=None,preemph=0.97, winfunc=lambda x:numpy.ones((x,))): """Compute Spectral Subband Centroid features from an audio signal. :param signal: the audio signal from which to compute features. Should be an N*1 array :param samplerate: the sample rate of the signal we are working with, in Hz. :param winlen: the length of the analysis window in seconds. Default is 0.025s (25 milliseconds) :param winstep: the step between successive windows in seconds. Default is 0.01s (10 milliseconds) :param nfilt: the number of filters in the filterbank, default 26. :param nfft: the FFT size. Default is 512. :param lowfreq: lowest band edge of mel filters. In Hz, default is 0. :param highfreq: highest band edge of mel filters. In Hz, default is samplerate/2 :param preemph: apply preemphasis filter with preemph as coefficient. 0 is no filter. Default is 0.97. :param winfunc: the analysis window to apply to each frame. By default no window is applied. You can use numpy window functions here e.g. winfunc=numpy.hamming :returns: A numpy array of size (NUMFRAMES by nfilt) containing features. Each row holds 1 feature vector. """ highfreq= highfreq or samplerate/2 signal = preemphasis(signal,preemph) frames = framesig(signal, winlen*samplerate, winstep*samplerate, winfunc) pspec = powspec(frames,nfft) pspec = numpy.where(pspec == 0,numpy.finfo(float).eps,pspec) # if things are all zeros we get problems fb = get_filterbanks(nfilt,nfft,samplerate,lowfreq,highfreq) feat = numpy.dot(pspec,fb.T) # compute the filterbank energies R = numpy.tile(numpy.linspace(1,samplerate/2,numpy.size(pspec,1)),(numpy.size(pspec,0),1)) return numpy.dot(pspec*R,fb.T) / feat def hz2mel(hz): """Convert a value in Hertz to Mels :param hz: a value in Hz. This can also be a numpy array, conversion proceeds element-wise. :returns: a value in Mels. If an array was passed in, an identical sized array is returned. """ return 2595 * numpy.log10(1+hz/700.) def mel2hz(mel): """Convert a value in Mels to Hertz :param mel: a value in Mels. This can also be a numpy array, conversion proceeds element-wise. :returns: a value in Hertz. If an array was passed in, an identical sized array is returned. """ return 700*(10**(mel/2595.0)-1) def get_filterbanks(nfilt=20,nfft=512,samplerate=16000,lowfreq=0,highfreq=None): """Compute a Mel-filterbank. The filters are stored in the rows, the columns correspond to fft bins. The filters are returned as an array of size nfilt * (nfft/2 + 1) :param nfilt: the number of filters in the filterbank, default 20. :param nfft: the FFT size. Default is 512. :param samplerate: the sample rate of the signal we are working with, in Hz. Affects mel spacing. :param lowfreq: lowest band edge of mel filters, default 0 Hz :param highfreq: highest band edge of mel filters, default samplerate/2 :returns: A numpy array of size nfilt * (nfft/2 + 1) containing filterbank. Each row holds 1 filter. """ highfreq= highfreq or samplerate/2 assert highfreq <= samplerate/2, "highfreq is greater than samplerate/2" # compute points evenly spaced in mels lowmel = hz2mel(lowfreq) highmel = hz2mel(highfreq) melpoints = numpy.linspace(lowmel,highmel,nfilt+2) # our points are in Hz, but we use fft bins, so we have to convert # from Hz to fft bin number bin = numpy.floor((nfft+1)*mel2hz(melpoints)/samplerate) fbank = numpy.zeros([nfilt,nfft//2+1]) for j in range(0,nfilt): for i in range(int(bin[j]), int(bin[j+1])): fbank[j,i] = (i - bin[j]) / (bin[j+1]-bin[j]) for i in range(int(bin[j+1]), int(bin[j+2])): fbank[j,i] = (bin[j+2]-i) / (bin[j+2]-bin[j+1]) return fbank def lifter(cepstra, L=22): """Apply a cepstral lifter the the matrix of cepstra. This has the effect of increasing the magnitude of the high frequency DCT coeffs. :param cepstra: the matrix of mel-cepstra, will be numframes * numcep in size. :param L: the liftering coefficient to use. Default is 22. L <= 0 disables lifter. """ if L > 0: nframes,ncoeff = numpy.shape(cepstra) n = numpy.arange(ncoeff) lift = 1 + (L/2.)*numpy.sin(numpy.pi*n/L) return lift*cepstra else: # values of L <= 0, do nothing return cepstra def delta(feat, N): """Compute delta features from a feature vector sequence. :param feat: A numpy array of size (NUMFRAMES by number of features) containing features. Each row holds 1 feature vector. :param N: For each frame, calculate delta features based on preceding and following N frames :returns: A numpy array of size (NUMFRAMES by number of features) containing delta features. Each row holds 1 delta feature vector. """ if N < 1: raise ValueError('N must be an integer >= 1') NUMFRAMES = len(feat) denominator = 2 * sum([i**2 for i in range(1, N+1)]) delta_feat = numpy.empty_like(feat) padded = numpy.pad(feat, ((N, N), (0, 0)), mode='edge') # padded version of feat for t in range(NUMFRAMES): delta_feat[t] = numpy.dot(numpy.arange(-N, N+1), padded[t : t+2*N+1]) / denominator # [t : t+2*N+1] == [(N+t)-N : (N+t)+N+1] return delta_feat