soundtouch implement of changing rate in a way same with resample(SRC).
%calc low pass filter coefficient. The low pass filter based on sinc function with hamming window.
function coeff = calCoeffs(cutoffFreq, len)
coeff = zeros(len ,1);
wc = 2 * pi * cutoffFreq;
tempCoeff = 2 * pi / len;
sum = 0;
for i = 0 : 1 : len -1
cntTemp = (i - len/2);
temp = cntTemp * wc;
% sinc function
if temp ~=0
h = sin(temp) / temp;
else
h = 1;
end
%hamming window
w = 0.54 + 0.46 * cos(tempCoeff * cntTemp);
coeff(i+1) = w * h;
sum = sum + coeff(i+1);
end
coeff = coeff / sum;
end
function output = firfilter(input, coeff)
inputLen = length(input(:, 1));
filterLen = length(coeff(:, 1));
output = zeros(inputLen ,1 );
outputLen = inputLen - filterLen;
for i = 1: 1: outputLen
inpos = i;
sum = 0;
for j = 1:1:filterLen
sum = sum + input(inpos ,1) * coeff(j, 1);
inpos = inpos + 1;
end
output(i, 1) = sum;
end
end
function output = cubicInterpolation(input, rate)
inputLen = length(input(:,1));
outputLen = floor(inputLen / rate);
output = zeros(outputLen ,1);
inputIdx = 1;
fract = 0;
outputIdx = 1;
while inputIdx < inputLen - 4
x1 = fract;
x2 = x1 * x1;
x3 = x1 * x2;
p0 = input(inputIdx , 1);
p1 = input(inputIdx + 1 , 1);
p2 = input(inputIdx + 2, 1);
p3 = input(inputIdx + 3, 1);
output(outputIdx ,1) = (-0.5*p0 + 1.5*p1 -1.5 *p2 + 0.5*p3) * x3 +(p0 - 2.5*p1 + 2*p2 -0.5*p3) *x2 + (-0.5*p0 + 0.5*p2) * x1 + p1;
outputIdx = outputIdx + 1;
fract = fract + rate;
whole = floor(fract);
fract = fract - whole;
inputIdx = inputIdx + whole;
end
end
function output = linearInterpolation(input, rate)
inputLen = length(input(:,1));
outputLen = floor(inputLen / rate);
output = zeros(outputLen ,1);
inputIdx = 1;
fract = 0;
outputIdx = 1;
while inputIdx < inputLen - 4
p0 = input(inputIdx , 1);
p1 = input(inputIdx + 1 , 1);
output(outputIdx ,1) = (1-fract) * po + fract * p1;
outputIdx = outputIdx + 1;
fract = fract + rate;
whole = floor(fract);
fract = fract - whole;
inputIdx = inputIdx + whole;
end
end
function output = changeRate(input, rate, interpMethod)
inputLen = length(input(:, 1));
outputLen = floor(inputLen / rate);
output = zeros(outputLen, 1);
if rate > 1
cutoffFreq = 0.5 / rate;
else
cutoffFreq = 0.5 * rate;
end
filterLen = 64;
coeff = calCoeffs(cutoffFreq, filterLen);
if rate < 1
%slow down, need interpolation first;
if strcmp(interMethod, 'cubic')
output = cubicInterpolation(input, rate);
else
output = linearInterpolation(input, rate);
end
output = firfilter(output, coeff);
else
%fast, need filter out the high freqency, then delete samples
output = firfilter(input, coeff);
if strcmp(interMethod, 'cubic')
output = cubicInterpolation(output, rate);
else
output = linearInterpolation(output, rate);
end
end
end
main.m:
clc;
clear all;
[input fs] = wavread('input.wav');
%if do SRC, rate = inputfs / outputfs;
rate = 0.5;
output = changeRate(input, rate, 'cubic');
wavwrite(output, fs, 'output.wav);