soundtouch change rate matlab implementation

soundtouch implement of changing rate in a way same with resample(SRC).

 When rate < 1, it need interpolate sample. and delete samples when rate > 1.

After  interpolation, there may be introduce high frequency. To avoid aliase, we usally apply a low pass filter on interpolated signal.

For the case of deleting samples, some signal may contains high frequecy, the difference between sample may be very sharp. Therefore, we use low pass filter before deleting samples.

When design fir low pass filter, we usaully use "w" (rad/s) as the main parameter instead freqence "f".

The relationship of "w" and "f" as following:

w = 2*pi * f/ fs;

For cutoff frequency, wc = 2*pi * fc / fs = 2 * pi * fc/(2*fn) =  pi * fc / fn, where fc is cutoff frequency, fn is Nywuist frequency.

For fc = = fn, wc = pi;

We use cubic method to interplate sample, the principle of cubic interpolation refer to https://www.paulinternet.nl/?page=bicubic

 

%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);