如何在 Tone.js 中改变音频缓冲区的音高？
How to pitchshift an audio buffer in Tone.js?
我想在 Javascript 中改变音频的音高，我认为最简单的方法是 Tone.js (https://tonejs.github.io/docs/r13/PitchShift)，但我的实现没有播放任何声音，但是没有错误。我做错了什么？
HTML（加载音频文件）
<html>
<div id="wrapper">
<button onclick="loadTheTrack()"></button>
</div>
<script src="Tone.js"></script>
<script src="seebelow.js"></script>
</html>
Javascript
var AudioContext = window.AudioContext || window.webkitAudioContext;
var audioCtx = new AudioContext();
Tone.setContext(audioCtx);
var audioDatas=[];
function loadTheTrack() {
var input = document.createElement('input');
input.type = 'file';
input.style = "display:none";
input.onchange = function (e) {
var file = e.target.files[0];
console.log(file);
var reader = new FileReader();
reader.onload = function () {
console.log("decoding audio data with" + this.result);
audioCtx.decodeAudioData(this.result, (decodedData) => {
// note: on older systems, may have to use deprecated noteOn(time);
audioDatas.push(decodedData);
doIt();
}, (e) => {
alert('Sorry this browser unable to download this file... try Chrome');
});
}
reader.readAsArrayBuffer(file);
}
document.getElementById("wrapper").appendChild(input);
input.click();
}
function generateAudioOffline(){
return Tone.Offline(function(Transport){
var pitchShift = new Tone.PitchShift({
pitch: -2
}).toMaster();
var tonbuf = new Tone.BufferSource(audioDatas[0]);
tonbuf.connect(pitchShift);
Transport.bpm.value = 106;
Transport.start();
}, 7);
}
function doIt() {
var buffer = generateAudioOffline().then(decodeBuffer => {
console.log(decodeBuffer);
var source = audioCtx.createBufferSource(); // creates a sound source
source.buffer = decodeBuffer._buffer; // tell the source which sound to play
source.connect(audioCtx.destination); // connect the source to the context's destination (the speakers)
console.log('starting');
source.start(0); // play the source now
});
}
所以我放弃了 ToneJS，只是从 c# 翻译了一个变调器 (https://sites.google.com/site/mikescoderama/pitch-shifting)
<!-- Translated from https://sites.google.com/site/mikescoderama/pitch-shifting to javascript By Seth Kitchen 2019-->
<html>
<div id="wrapper">
<p>PitchShift factor value which is between 0.5 (one octave down) and 2. (one octave up)</p>
<input id="shiftAmount" type="text" value=".5" />
<button onclick="loadTheTrack()">Pick a file and then wait for the pitch shift to happen! </button>
<p id="status">Waiting for you to pick file.</p>
</div>
<script>
/****************************************************************************
*
* NAME: PitchShift
* VERSION: 1.2
* HOME URL: http://www.dspdimension.com
* KNOWN BUGS: none
*
* SYNOPSIS: Routine for doing pitch shifting while maintaining
* duration using the Short Time Fourier Transform.
*
* DESCRIPTION: The routine takes a pitchShift factor value which is between 0.5
* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
* the pitch. numSampsToProcess tells the routine how many samples in indata[0...
* numSampsToProcess-1] should be pitch shifted and moved to outdata[0 ...
* numSampsToProcess-1]. The two buffers can be identical (ie. it can process the
* data in-place). fftFrameSize defines the FFT frame size used for the
* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
* MAX_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
* oversampling factor which also determines the overlap between adjacent STFT
* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
* recommended for best quality. sampleRate takes the sample rate for the signal
* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in
* indata[] should be in the range [-1.0, 1.0), which is also the output range
* for the data, make sure you scale the data accordingly (for 16bit signed integers
* you would have to divide (and multiply) by 32768).
*
* COPYRIGHT 1999-2006 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
*
* The Wide Open License (WOL)
*
* Permission to use, copy, modify, distribute and sell this software and its
* documentation for any purpose is hereby granted without fee, provided that
* the above copyright notice and this license appear in all source copies.
* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
*
*****************************************************************************/
const MAX_FRAME_LENGTH = 16000;
var gInFIFO=new Array(MAX_FRAME_LENGTH).fill(0.0);
var gOutFIFO=new Array(MAX_FRAME_LENGTH).fill(0.0);
var gFFTworksp = new Array(2*MAX_FRAME_LENGTH).fill(0.0);
var gLastPhase = new Array(MAX_FRAME_LENGTH/2+1).fill(0.0);
var gSumPhase = new Array(MAX_FRAME_LENGTH/2+1).fill(0.0);
var gOutputAccum = new Array(2*MAX_FRAME_LENGTH).fill(0.0);
var gAnaFreq = new Array(MAX_FRAME_LENGTH).fill(0.0);
var gAnaMagn = new Array(MAX_FRAME_LENGTH).fill(0.0);
var gSynFreq = new Array(MAX_FRAME_LENGTH).fill(0.0);
var gSynMagn =new Array(MAX_FRAME_LENGTH).fill(0.0);
var gRover=0;
var AudioContext = window.AudioContext || window.webkitAudioContext;
var audioCtx = new AudioContext();
function loadTheTrack()
{
var input = document.createElement('input');
input.type = 'file';
input.style = "display:none";
input.onchange = function (e) {
var file = e.target.files[0];
console.log(file);
var reader = new FileReader();
reader.onload = function () {
console.log("decoding audio data with" + this.result);
document.getElementById('status').innerText="Pitch Shifting (yeah it takes a min)....";
audioCtx.decodeAudioData(this.result, (decodedData) => {
var in_data_l=decodedData.getChannelData(0);
console.log(in_data_l);
var in_data_r=decodedData.getChannelData(1);
console.log(in_data_r);
var shiftAmount=document.getElementById("shiftAmount").value;
console.log(shiftAmount);
in_data_l= PitchShift(shiftAmount, in_data_l.length, 1024, 10, audioCtx.sampleRate, in_data_l);
console.log(in_data_l);
in_data_r= PitchShift(shiftAmount, in_data_l.length, 1024, 10, audioCtx.sampleRate, in_data_r);
decodedData.copyToChannel(in_data_l, 0);
decodedData.copyToChannel(in_data_r, 1);
var source = audioCtx.createBufferSource(); // creates a sound source
source.buffer = decodedData; // tell the source which sound to play
source.connect(audioCtx.destination); // connect the source to the context's destination (the speakers)
console.log('starting');
document.getElementById('status').innerText="Playing...";
source.start(0); // play the source now
}, (e) => {
alert('Sorry this browser unable to download this file... try Chrome');
});
}
reader.readAsArrayBuffer(file);
}
document.getElementById("wrapper").appendChild(input);
input.click();
}
function PitchShift(/* float[*/ pitchShift, /* long */ numSampsToProcess, /* long */ fftFrameSize,
/* long */ osamp, /* float[*/ sampleRate, /* float[] */ indata) {
/* double */ var magn, phase, tmp, window, real, imag;
/* double */ var freqPerBin, expct;
/* long */ var i, k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
/* float[] */var outdata = indata;
/* set up some handy variables */
fftFrameSize2 = Math.trunc(fftFrameSize / 2);
stepSize = Math.trunc(fftFrameSize / osamp);
freqPerBin = sampleRate / /* (double) */fftFrameSize;
expct = 2.0 * Math.PI * /* (double) */stepSize / /* (double) */fftFrameSize;
inFifoLatency = Math.trunc(fftFrameSize - stepSize);
if (gRover == 0) gRover = inFifoLatency;
/* main processing loop */
for (i = 0; i < numSampsToProcess; i++) {
/* As long as we have not yet collected enough data just read in */
gInFIFO[gRover] = indata[i];
outdata[i] = gOutFIFO[gRover - inFifoLatency];
gRover++;
/* now we have enough data for processing */
if (gRover >= fftFrameSize) {
gRover = inFifoLatency;
/* do windowing and re,im interleave */
for (k = 0; k < fftFrameSize; k++) {
window = -.5 * Math.cos(2.0 * Math.PI * /* (double) */k / /* (double) */fftFrameSize) + .5;
gFFTworksp[2 * k] = /* (float) */(gInFIFO[k] * window);
gFFTworksp[2 * k + 1] = 0.0;
}
/* ***************** ANALYSIS ******************* */
/* do transform */
ShortTimeFourierTransform(gFFTworksp, fftFrameSize, -1);
/* this is the analysis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* de-interlace FFT buffer */
real = gFFTworksp[2 * k];
imag = gFFTworksp[2 * k + 1];
/* compute magnitude and phase */
magn = 2.0 * Math.sqrt(real * real + imag * imag);
phase = Math.atan2(imag, real);
/* compute phase difference */
tmp = phase - gLastPhase[k];
gLastPhase[k] = /* (float) */phase;
/* subtract expected phase difference */
tmp -= /* (double) */k * expct;
/* map delta phase into +/- Pi interval */
qpd = /* (long) */Math.trunc(tmp / Math.PI);
if (qpd >= 0) qpd += qpd & 1;
else qpd -= qpd & 1;
tmp -= Math.PI * /* (double) */qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp * tmp / (2.0 * Math.PI);
/* compute the k-th partials' true frequency */
tmp = /* (double) */k * freqPerBin + tmp * freqPerBin;
/* store magnitude and true frequency in analysis arrays */
gAnaMagn[k] = /* (float) */magn;
gAnaFreq[k] = /* (float) */tmp;
}
/* ***************** PROCESSING ******************* */
/* this does the actual pitch shifting */
for (var zero = 0; zero < fftFrameSize; zero++) {
gSynMagn[zero] = 0;
gSynFreq[zero] = 0;
}
for (k = 0; k <= fftFrameSize2; k++) {
index = /* (long) */Math.trunc(k * pitchShift);
if (index <= fftFrameSize2) {
gSynMagn[index] += gAnaMagn[k];
gSynFreq[index] = gAnaFreq[k] * pitchShift;
}
}
/* ***************** SYNTHESIS ******************* */
/* this is the synthesis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* get magnitude and true frequency from synthesis arrays */
magn = gSynMagn[k];
tmp = gSynFreq[k];
/* subtract bin mid frequency */
tmp -= /* (double) */k * freqPerBin;
/* get bin deviation from freq deviation */
tmp /= freqPerBin;
/* take osamp into account */
tmp = 2.0 * Math.PI * tmp / osamp;
/* add the overlap phase advance back in */
tmp += /* (double) */k * expct;
/* accumulate delta phase to get bin phase */
gSumPhase[k] += /* (float) */tmp;
phase = gSumPhase[k];
/* get real and imag part and re-interleave */
gFFTworksp[2 * k] = /* (float) */(magn * Math.cos(phase));
gFFTworksp[2 * k + 1] = /* (float) */(magn * Math.sin(phase));
}
/* zero negative frequencies */
for (k = fftFrameSize + 2; k < 2 * fftFrameSize; k++) gFFTworksp[k] = 0.0;
/* do inverse transform */
ShortTimeFourierTransform(gFFTworksp, fftFrameSize, 1);
/* do windowing and add to output accumulator */
for (k = 0; k < fftFrameSize; k++) {
window = -.5 * Math.cos(2.0 * Math.PI * /* (double) */k / /* (double) */fftFrameSize) + .5;
gOutputAccum[k] += /* (float) */(2.0 * window * gFFTworksp[2 * k] / (fftFrameSize2 * osamp));
}
for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
/* shift accumulator */
//memmove(gOutputAccum, gOutputAccum + stepSize, fftFrameSize * sizeof(float));
for (k = 0; k < fftFrameSize; k++) {
gOutputAccum[k] = gOutputAccum[k + stepSize];
}
/* move input FIFO */
for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k + stepSize];
}
}
return outdata;
}
function ShortTimeFourierTransform(/* float[] */ fftBuffer, /* long */ fftFrameSize, /* long */ sign) {
/* float */ var wr, wi, arg, temp;
/* float */ var tr, ti, ur, ui;
/* long */ var i, bitm, j, le, le2, k;
for (i = 2; i < 2 * fftFrameSize - 2; i += 2) {
for (bitm = 2, j = 0; bitm < 2 * fftFrameSize; bitm <<= 1) {
if ((i & bitm) != 0) j++;
j <<= 1;
}
if (i < j) {
temp = fftBuffer[i];
fftBuffer[i] = fftBuffer[j];
fftBuffer[j] = temp;
temp = fftBuffer[i + 1];
fftBuffer[i + 1] = fftBuffer[j + 1];
fftBuffer[j + 1] = temp;
}
}
/* long */ var max = /* (long) */Math.trunc(Math.log(fftFrameSize) / Math.log(2.0) + .5);
for (k = 0, le = 2; k < max; k++) {
le <<= 1;
le2 = le >> 1;
ur = 1.0;
ui = 0.0;
arg = /* (float) */Math.PI / (le2 >> 1);
wr = /* (float) */Math.cos(arg);
wi = /* (float) */(sign * Math.sin(arg));
for (j = 0; j < le2; j += 2) {
for (i = j; i < 2 * fftFrameSize; i += le) {
tr = fftBuffer[i + le2] * ur - fftBuffer[i + le2 + 1] * ui;
ti = fftBuffer[i + le2] * ui + fftBuffer[i + le2 + 1] * ur;
fftBuffer[i + le2] = fftBuffer[i] - tr;
fftBuffer[i + le2 + 1] = fftBuffer[i + 1] - ti;
fftBuffer[i] += tr;
fftBuffer[i + 1] += ti;
}
tr = ur * wr - ui * wi;
ui = ur * wi + ui * wr;
ur = tr;
}
}
}
</script>
</html>
我想在 Javascript 中改变音频的音高，我认为最简单的方法是 Tone.js (https://tonejs.github.io/docs/r13/PitchShift)，但我的实现没有播放任何声音，但是没有错误。我做错了什么？
HTML（加载音频文件）
<html>
<div id="wrapper">
<button onclick="loadTheTrack()"></button>
</div>
<script src="Tone.js"></script>
<script src="seebelow.js"></script>
</html>
Javascript
var AudioContext = window.AudioContext || window.webkitAudioContext;
var audioCtx = new AudioContext();
Tone.setContext(audioCtx);
var audioDatas=[];
function loadTheTrack() {
var input = document.createElement('input');
input.type = 'file';
input.style = "display:none";
input.onchange = function (e) {
var file = e.target.files[0];
console.log(file);
var reader = new FileReader();
reader.onload = function () {
console.log("decoding audio data with" + this.result);
audioCtx.decodeAudioData(this.result, (decodedData) => {
// note: on older systems, may have to use deprecated noteOn(time);
audioDatas.push(decodedData);
doIt();
}, (e) => {
alert('Sorry this browser unable to download this file... try Chrome');
});
}
reader.readAsArrayBuffer(file);
}
document.getElementById("wrapper").appendChild(input);
input.click();
}
function generateAudioOffline(){
return Tone.Offline(function(Transport){
var pitchShift = new Tone.PitchShift({
pitch: -2
}).toMaster();
var tonbuf = new Tone.BufferSource(audioDatas[0]);
tonbuf.connect(pitchShift);
Transport.bpm.value = 106;
Transport.start();
}, 7);
}
function doIt() {
var buffer = generateAudioOffline().then(decodeBuffer => {
console.log(decodeBuffer);
var source = audioCtx.createBufferSource(); // creates a sound source
source.buffer = decodeBuffer._buffer; // tell the source which sound to play
source.connect(audioCtx.destination); // connect the source to the context's destination (the speakers)
console.log('starting');
source.start(0); // play the source now
});
}
所以我放弃了 ToneJS，只是从 c# 翻译了一个变调器 (https://sites.google.com/site/mikescoderama/pitch-shifting)
<!-- Translated from https://sites.google.com/site/mikescoderama/pitch-shifting to javascript By Seth Kitchen 2019-->
<html>
<div id="wrapper">
<p>PitchShift factor value which is between 0.5 (one octave down) and 2. (one octave up)</p>
<input id="shiftAmount" type="text" value=".5" />
<button onclick="loadTheTrack()">Pick a file and then wait for the pitch shift to happen! </button>
<p id="status">Waiting for you to pick file.</p>
</div>
<script>
/****************************************************************************
*
* NAME: PitchShift
* VERSION: 1.2
* HOME URL: http://www.dspdimension.com
* KNOWN BUGS: none
*
* SYNOPSIS: Routine for doing pitch shifting while maintaining
* duration using the Short Time Fourier Transform.
*
* DESCRIPTION: The routine takes a pitchShift factor value which is between 0.5
* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
* the pitch. numSampsToProcess tells the routine how many samples in indata[0...
* numSampsToProcess-1] should be pitch shifted and moved to outdata[0 ...
* numSampsToProcess-1]. The two buffers can be identical (ie. it can process the
* data in-place). fftFrameSize defines the FFT frame size used for the
* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
* MAX_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
* oversampling factor which also determines the overlap between adjacent STFT
* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
* recommended for best quality. sampleRate takes the sample rate for the signal
* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in
* indata[] should be in the range [-1.0, 1.0), which is also the output range
* for the data, make sure you scale the data accordingly (for 16bit signed integers
* you would have to divide (and multiply) by 32768).
*
* COPYRIGHT 1999-2006 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
*
* The Wide Open License (WOL)
*
* Permission to use, copy, modify, distribute and sell this software and its
* documentation for any purpose is hereby granted without fee, provided that
* the above copyright notice and this license appear in all source copies.
* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
*
*****************************************************************************/
const MAX_FRAME_LENGTH = 16000;
var gInFIFO=new Array(MAX_FRAME_LENGTH).fill(0.0);
var gOutFIFO=new Array(MAX_FRAME_LENGTH).fill(0.0);
var gFFTworksp = new Array(2*MAX_FRAME_LENGTH).fill(0.0);
var gLastPhase = new Array(MAX_FRAME_LENGTH/2+1).fill(0.0);
var gSumPhase = new Array(MAX_FRAME_LENGTH/2+1).fill(0.0);
var gOutputAccum = new Array(2*MAX_FRAME_LENGTH).fill(0.0);
var gAnaFreq = new Array(MAX_FRAME_LENGTH).fill(0.0);
var gAnaMagn = new Array(MAX_FRAME_LENGTH).fill(0.0);
var gSynFreq = new Array(MAX_FRAME_LENGTH).fill(0.0);
var gSynMagn =new Array(MAX_FRAME_LENGTH).fill(0.0);
var gRover=0;
var AudioContext = window.AudioContext || window.webkitAudioContext;
var audioCtx = new AudioContext();
function loadTheTrack()
{
var input = document.createElement('input');
input.type = 'file';
input.style = "display:none";
input.onchange = function (e) {
var file = e.target.files[0];
console.log(file);
var reader = new FileReader();
reader.onload = function () {
console.log("decoding audio data with" + this.result);
document.getElementById('status').innerText="Pitch Shifting (yeah it takes a min)....";
audioCtx.decodeAudioData(this.result, (decodedData) => {
var in_data_l=decodedData.getChannelData(0);
console.log(in_data_l);
var in_data_r=decodedData.getChannelData(1);
console.log(in_data_r);
var shiftAmount=document.getElementById("shiftAmount").value;
console.log(shiftAmount);
in_data_l= PitchShift(shiftAmount, in_data_l.length, 1024, 10, audioCtx.sampleRate, in_data_l);
console.log(in_data_l);
in_data_r= PitchShift(shiftAmount, in_data_l.length, 1024, 10, audioCtx.sampleRate, in_data_r);
decodedData.copyToChannel(in_data_l, 0);
decodedData.copyToChannel(in_data_r, 1);
var source = audioCtx.createBufferSource(); // creates a sound source
source.buffer = decodedData; // tell the source which sound to play
source.connect(audioCtx.destination); // connect the source to the context's destination (the speakers)
console.log('starting');
document.getElementById('status').innerText="Playing...";
source.start(0); // play the source now
}, (e) => {
alert('Sorry this browser unable to download this file... try Chrome');
});
}
reader.readAsArrayBuffer(file);
}
document.getElementById("wrapper").appendChild(input);
input.click();
}
function PitchShift(/* float[*/ pitchShift, /* long */ numSampsToProcess, /* long */ fftFrameSize,
/* long */ osamp, /* float[*/ sampleRate, /* float[] */ indata) {
/* double */ var magn, phase, tmp, window, real, imag;
/* double */ var freqPerBin, expct;
/* long */ var i, k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
/* float[] */var outdata = indata;
/* set up some handy variables */
fftFrameSize2 = Math.trunc(fftFrameSize / 2);
stepSize = Math.trunc(fftFrameSize / osamp);
freqPerBin = sampleRate / /* (double) */fftFrameSize;
expct = 2.0 * Math.PI * /* (double) */stepSize / /* (double) */fftFrameSize;
inFifoLatency = Math.trunc(fftFrameSize - stepSize);
if (gRover == 0) gRover = inFifoLatency;
/* main processing loop */
for (i = 0; i < numSampsToProcess; i++) {
/* As long as we have not yet collected enough data just read in */
gInFIFO[gRover] = indata[i];
outdata[i] = gOutFIFO[gRover - inFifoLatency];
gRover++;
/* now we have enough data for processing */
if (gRover >= fftFrameSize) {
gRover = inFifoLatency;
/* do windowing and re,im interleave */
for (k = 0; k < fftFrameSize; k++) {
window = -.5 * Math.cos(2.0 * Math.PI * /* (double) */k / /* (double) */fftFrameSize) + .5;
gFFTworksp[2 * k] = /* (float) */(gInFIFO[k] * window);
gFFTworksp[2 * k + 1] = 0.0;
}
/* ***************** ANALYSIS ******************* */
/* do transform */
ShortTimeFourierTransform(gFFTworksp, fftFrameSize, -1);
/* this is the analysis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* de-interlace FFT buffer */
real = gFFTworksp[2 * k];
imag = gFFTworksp[2 * k + 1];
/* compute magnitude and phase */
magn = 2.0 * Math.sqrt(real * real + imag * imag);
phase = Math.atan2(imag, real);
/* compute phase difference */
tmp = phase - gLastPhase[k];
gLastPhase[k] = /* (float) */phase;
/* subtract expected phase difference */
tmp -= /* (double) */k * expct;
/* map delta phase into +/- Pi interval */
qpd = /* (long) */Math.trunc(tmp / Math.PI);
if (qpd >= 0) qpd += qpd & 1;
else qpd -= qpd & 1;
tmp -= Math.PI * /* (double) */qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp * tmp / (2.0 * Math.PI);
/* compute the k-th partials' true frequency */
tmp = /* (double) */k * freqPerBin + tmp * freqPerBin;
/* store magnitude and true frequency in analysis arrays */
gAnaMagn[k] = /* (float) */magn;
gAnaFreq[k] = /* (float) */tmp;
}
/* ***************** PROCESSING ******************* */
/* this does the actual pitch shifting */
for (var zero = 0; zero < fftFrameSize; zero++) {
gSynMagn[zero] = 0;
gSynFreq[zero] = 0;
}
for (k = 0; k <= fftFrameSize2; k++) {
index = /* (long) */Math.trunc(k * pitchShift);
if (index <= fftFrameSize2) {
gSynMagn[index] += gAnaMagn[k];
gSynFreq[index] = gAnaFreq[k] * pitchShift;
}
}
/* ***************** SYNTHESIS ******************* */
/* this is the synthesis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* get magnitude and true frequency from synthesis arrays */
magn = gSynMagn[k];
tmp = gSynFreq[k];
/* subtract bin mid frequency */
tmp -= /* (double) */k * freqPerBin;
/* get bin deviation from freq deviation */
tmp /= freqPerBin;
/* take osamp into account */
tmp = 2.0 * Math.PI * tmp / osamp;
/* add the overlap phase advance back in */
tmp += /* (double) */k * expct;
/* accumulate delta phase to get bin phase */
gSumPhase[k] += /* (float) */tmp;
phase = gSumPhase[k];
/* get real and imag part and re-interleave */
gFFTworksp[2 * k] = /* (float) */(magn * Math.cos(phase));
gFFTworksp[2 * k + 1] = /* (float) */(magn * Math.sin(phase));
}
/* zero negative frequencies */
for (k = fftFrameSize + 2; k < 2 * fftFrameSize; k++) gFFTworksp[k] = 0.0;
/* do inverse transform */
ShortTimeFourierTransform(gFFTworksp, fftFrameSize, 1);
/* do windowing and add to output accumulator */
for (k = 0; k < fftFrameSize; k++) {
window = -.5 * Math.cos(2.0 * Math.PI * /* (double) */k / /* (double) */fftFrameSize) + .5;
gOutputAccum[k] += /* (float) */(2.0 * window * gFFTworksp[2 * k] / (fftFrameSize2 * osamp));
}
for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
/* shift accumulator */
//memmove(gOutputAccum, gOutputAccum + stepSize, fftFrameSize * sizeof(float));
for (k = 0; k < fftFrameSize; k++) {
gOutputAccum[k] = gOutputAccum[k + stepSize];
}
/* move input FIFO */
for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k + stepSize];
}
}
return outdata;
}
function ShortTimeFourierTransform(/* float[] */ fftBuffer, /* long */ fftFrameSize, /* long */ sign) {
/* float */ var wr, wi, arg, temp;
/* float */ var tr, ti, ur, ui;
/* long */ var i, bitm, j, le, le2, k;
for (i = 2; i < 2 * fftFrameSize - 2; i += 2) {
for (bitm = 2, j = 0; bitm < 2 * fftFrameSize; bitm <<= 1) {
if ((i & bitm) != 0) j++;
j <<= 1;
}
if (i < j) {
temp = fftBuffer[i];
fftBuffer[i] = fftBuffer[j];
fftBuffer[j] = temp;
temp = fftBuffer[i + 1];
fftBuffer[i + 1] = fftBuffer[j + 1];
fftBuffer[j + 1] = temp;
}
}
/* long */ var max = /* (long) */Math.trunc(Math.log(fftFrameSize) / Math.log(2.0) + .5);
for (k = 0, le = 2; k < max; k++) {
le <<= 1;
le2 = le >> 1;
ur = 1.0;
ui = 0.0;
arg = /* (float) */Math.PI / (le2 >> 1);
wr = /* (float) */Math.cos(arg);
wi = /* (float) */(sign * Math.sin(arg));
for (j = 0; j < le2; j += 2) {
for (i = j; i < 2 * fftFrameSize; i += le) {
tr = fftBuffer[i + le2] * ur - fftBuffer[i + le2 + 1] * ui;
ti = fftBuffer[i + le2] * ui + fftBuffer[i + le2 + 1] * ur;
fftBuffer[i + le2] = fftBuffer[i] - tr;
fftBuffer[i + le2 + 1] = fftBuffer[i + 1] - ti;
fftBuffer[i] += tr;
fftBuffer[i + 1] += ti;
}
tr = ur * wr - ui * wi;
ui = ur * wi + ui * wr;
ur = tr;
}
}
}
</script>
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