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MATLAB Code For Speech Recognition

This MATLAB code performs speech recognition by calculating Mel frequency cepstrum coefficients (MFCCs) from an input audio signal. It takes in an audio signal, sampling rate, and optional frame rate. It applies a Hamming window, takes the Fourier transform, filters the magnitudes into mel-scale filter banks, takes the log, and applies a discrete cosine transform to reduce dimensionality. This results in MFCC features that can be used for speech recognition. It also optionally returns various intermediate processing steps for analysis. The code was originally written in 1993 and has been modified over time.

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Ravi Teja
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635 views4 pages

MATLAB Code For Speech Recognition

This MATLAB code performs speech recognition by calculating Mel frequency cepstrum coefficients (MFCCs) from an input audio signal. It takes in an audio signal, sampling rate, and optional frame rate. It applies a Hamming window, takes the Fourier transform, filters the magnitudes into mel-scale filter banks, takes the log, and applies a discrete cosine transform to reduce dimensionality. This results in MFCC features that can be used for speech recognition. It also optionally returns various intermediate processing steps for analysis. The code was originally written in 1993 and has been modified over time.

Uploaded by

Ravi Teja
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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12/10/2015

MATLAB code for speech recognition

Re: MATLAB code for speech recognition


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mfcc - Mel frequency cepstrum coefficient analysis.


[ceps,freqresp,fb,fbrecon,freqrecon] = ...
mfcc(input, samplingRate, [frameRate])
Find the cepstral coefficients (ceps) corresponding to the
input. Four other quantities are optionally returned that
represent:
the detailed fft magnitude (freqresp) used in MFCC calculation,
the mel-scale filter bank output (fb)
the filter bank output by inverting the cepstrals with a cosine
transform (fbrecon),
the smooth frequency response by interpolating the fb reconstruction
(freqrecon)
-- Malcolm Slaney, August 1993
Modified a bit to make testing an algorithm easier... 4/15/94
Fixed Cosine Transform (indices of cos() were swapped) - 5/26/95
Added optional frameRate argument - 6/8/95
Added proper filterbank reconstruction using inverse DCT - 10/27/95
Added filterbank inversion to reconstruct spectrum - 11/1/95

% (c) 1998 Interval Research Corporation


function [ceps,freqresp,fb,fbrecon,freqrecon] = ...
MFCC(input, samplingRate, frameRate)
global mfccDCTMatrix mfccFilterWeights
[r c] = size(input);
if (r > c)
input=input';
end
% Filter bank parameters
lowestFrequency = 133.3333;
linearFilters = 13;
linearSpacing = 66.66666666;
logFilters = 27;
logSpacing = 1.0711703;
fftSize = 512;
cepstralCoefficients = 13;
windowSize = 400;
windowSize = 256; % Standard says 400, but 256 makes more sense
% Really should be a function of the sample
% rate (and the lowestFrequency) and the
% frame rate.
if (nargin < 2) samplingRate = 16000; end;
if (nargin < 3) frameRate = 100; end;
% Keep this around for later....
totalFilters = linearFilters + logFilters;
%
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Now figure the band edges. Interesting frequencies are spaced


by linearSpacing for a while, then go logarithmic. First figure
all the interesting frequencies. Lower, center, and upper band
edges are all consequtive interesting frequencies.

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MATLAB code for speech recognition

freqs = lowestFrequency + (0:linearFilters-1)*linearSpacing;


freqs(linearFilters+1:totalFilters+2) = ...
freqs(linearFilters) * logSpacing.^(1:logFilters+2);
lower = freqs(1:totalFilters);
center = freqs(2:totalFilters+1);
upper = freqs(3:totalFilters+2);
% We now want to combine FFT bins so that each filter has unit
% weight, assuming a triangular weighting function. First figure
% out the height of the triangle, then we can figure out each
% frequencies contribution
mfccFilterWeights = zeros(totalFilters,fftSize);
triangleHeight = 2./(upper-lower);
fftFreqs = (0:fftSize-1)/fftSize*samplingRate;
for chan=1:totalFilters
mfccFilterWeights(chan,:) = ...
(fftFreqs > lower(chan) & fftFreqs <= center(chan)).* ...
triangleHeight(chan).*(fftFreqs-lower(chan))/(center(chan)-lower(chan)) + ...
(fftFreqs > center(chan) & fftFreqs < upper(chan)).* ...
triangleHeight(chan).*(upper(chan)-fftFreqs)/(upper(chan)-center(chan));
end
%semilogx(fftFreqs,mfccFilterWeights')
%axis([lower(1) upper(totalFilters) 0 max(max(mfccFilterWeights))])
hamWindow = 0.54 - 0.46*cos(2*pi*(0:windowSize-1)/windowSize);
if 0 % Window it like ComplexSpectrum
windowStep = samplingRate/frameRate;
a = .54;
b = -.46;
wr = sqrt(windowStep/windowSize);
phi = pi/windowSize;
hamWindow = 2*wr/sqrt(4*a*a+2*b*b)* ...
(a + b*cos(2*pi*(0:windowSize-1)/windowSize + phi));
end
% Figure out Discrete Cosine Transform. We want a matrix
% dct(i,j) which is totalFilters x cepstralCoefficients in size.
% The i,j component is given by
% cos( i * (j+0.5)/totalFilters pi )
% where we have assumed that i and j start at 0.
mfccDCTMatrix = 1/sqrt(totalFilters/2)*cos((0:(cepstralCoefficients-1))' * ...
(2*(0:(totalFilters-1))+1) * pi/2/totalFilters);
mfccDCTMatrix(1,:) = mfccDCTMatrix(1,:) * sqrt(2)/2;
%imagesc(mfccDCTMatrix);
% Filter the input with the preemphasis filter. Also figure how
% many columns of data we will end up with.
if 1
preEmphasized = filter([1 -.97], 1, input);
else
preEmphasized = input;
end
windowStep = samplingRate/frameRate;
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MATLAB code for speech recognition

cols = fix((length(input)-windowSize)/windowStep);
% Allocate all the space we need for the output arrays.
ceps = zeros(cepstralCoefficients, cols);
if (nargout > 1) freqresp = zeros(fftSize/2, cols); end;
if (nargout > 2) fb = zeros(totalFilters, cols); end;
% Invert the filter bank center frequencies. For each FFT bin
% we want to know the exact position in the filter bank to find
% the original frequency response. The next block of code finds the
% integer and fractional sampling positions.
if (nargout > 4)
fr = (0:(fftSize/2-1))'/(fftSize/2)*samplingRate/2;
j = 1;
for i=1:(fftSize/2)
if fr(i) > center(j+1)
j = j + 1;
end
if j > totalFilters-1
j = totalFilters-1;
end
fr(i) = min(totalFilters-.0001, ...
max(1,j + (fr(i)-center(j))/(center(j+1)-center(j))));
end
fri = fix(fr);
frac = fr - fri;
freqrecon = zeros(fftSize/2, cols);
end
% Ok, now let's do the processing. For each chunk of data:
% * Window the data with a hamming window,
% * Shift it into FFT order,
% * Find the magnitude of the fft,
% * Convert the fft data into filter bank outputs,
% * Find the log base 10,
% * Find the cosine transform to reduce dimensionality.
for start=0:cols-1
first = floor(start*windowStep) + 1;
last = first + windowSize-1;
fftData = zeros(1,fftSize);
fftData(1:windowSize) = preEmphasized(first:last).*hamWindow;
fftMag = abs(fft(fftData));
earMag = log10(mfccFilterWeights * fftMag');
ceps(:,start+1) = mfccDCTMatrix * earMag;
if (nargout > 1) freqresp(:,start+1) = fftMag(1:fftSize/2)'; end;
if (nargout > 2) fb(:,start+1) = earMag; end
if (nargout > 3)
fbrecon(:,start+1) = ...
mfccDCTMatrix(1:cepstralCoefficients,:)' * ...
ceps(:,start+1);
end
if (nargout > 4)
f10 = 10.^fbrecon(:,start+1);
freqrecon(:,start+1) = samplingRate/fftSize * ...
(f10(fri).*(1-frac) + f10(fri+1).*frac);
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MATLAB code for speech recognition

end
end
% OK, just to check things, let's also reconstruct the original FB
% output. We do this by multiplying the cepstral data by the transpose
% of the original DCT matrix. This all works because we were careful to
% scale the DCT matrix so it was orthonormal.
if 1 && (nargout > 3)
fbrecon = mfccDCTMatrix(1:cepstralCoefficients,:)' * ceps;
% imagesc(mt(:,1:cepstralCoefficients)*mfccDCTMatrix );
end;

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