我正在尝试使用下面的PitchShifter.java类更改某些录制声音的音高：
package mypackage;
import net.rim.device.api.util.MathUtilities;
//package com.course.android.voicechanger;
//import android.util.Log;
/****************************************************************************
*
* NAME: PitchShift.cs
* 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 www.dspguru.com/wol.htm for more information.
*
*****************************************************************************/
/****************************************************************************
*
* This code was converted to C# by Michael Knight madmik3 at gmail dot com. sites.google.com/site/mikescoderama/
*
*****************************************************************************/
public class PitchShifter2 {
private static int MAX_FRAME_LENGTH = 16000;
private static float[] gInFIFO = new float[MAX_FRAME_LENGTH];
private static float[] gOutFIFO = new float[MAX_FRAME_LENGTH];
private static float[] gFFTworksp = new float[2 * MAX_FRAME_LENGTH];
private static float[] gLastPhase = new float[MAX_FRAME_LENGTH / 2 + 1];
private static float[] gSumPhase = new float[MAX_FRAME_LENGTH / 2 + 1];
private static float[] gOutputAccum = new float[2 * MAX_FRAME_LENGTH];
private static float[] gAnaFreq = new float[MAX_FRAME_LENGTH];
private static float[] gAnaMagn = new float[MAX_FRAME_LENGTH];
private static float[] gSynFreq = new float[MAX_FRAME_LENGTH];
private static float[] gSynMagn = new float[MAX_FRAME_LENGTH];
private static long gRover, gInit;
public static void PitchShift2(float pitchShift, long numSampsToProcess, float sampleRate, float[] indata) {
// PitchShift2(pitchShift, numSampsToProcess, (long) 256, (long) 10, sampleRate, indata);
PitchShift2(pitchShift, numSampsToProcess, (long) 1024, (long) 32, sampleRate, indata);
}
public static void PitchShift2(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float[] indata) {
double magn, phase, tmp, window, real, imag;
double freqPerBin, expct;
long i, k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
float[] outdata = indata;
/* set up some handy variables */
fftFrameSize2 = fftFrameSize / 2;
stepSize = fftFrameSize / osamp;
freqPerBin = sampleRate / (double) fftFrameSize;
expct = 2.0 * Math.PI * (double) stepSize / (double) fftFrameSize;
inFifoLatency = fftFrameSize - stepSize;
if (gRover == 0)
gRover = inFifoLatency;
int c = 0;
int round = 0;
/* main processing loop */
for (i = 0; i < numSampsToProcess; i++) {
/* As long as we have not yet collected enough data just read in */
gInFIFO[(int) gRover] = indata[(int) i];
outdata[(int) i] = gOutFIFO[(int) (gRover - inFifoLatency)];
gRover++;
/* now we have enough data for processing */
if (gRover >= fftFrameSize) {
c++;
if (c > 100) {
// Log.d("Liwei", "round= " + round++);
System.out.println("PitchShifter2.PitchShift(.....................): Liwei" + "round= " + round++);
c = 0;
}
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[(int) (2 * k)] = (float) (gInFIFO[(int) k] * window);
gFFTworksp[(int) (2 * k + 1)] = 0.0F;
}
/* ***************** ANALYSIS ******************* */
/* do transform */
ShortTimeFourierTransform(gFFTworksp, fftFrameSize, -1);
/* this is the analysis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* de-interlace FFT buffer */
real = gFFTworksp[(int) (2 * k)];
imag = gFFTworksp[(int) (2 * k + 1)];
/* compute magnitude and phase */
magn = 2.0 * Math.sqrt(real * real + imag * imag);
phase = MathUtilities.atan2(imag, real);
/* compute phase difference */
tmp = phase - gLastPhase[(int) k];
gLastPhase[(int) k] = (float) phase;
/* subtract expected phase difference */
tmp -= (double) k * expct;
/* map delta phase into +/- Pi interval */
qpd = (long) (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[(int) k] = (float) magn;
gAnaFreq[(int) k] = (float) tmp;
}
/* ***************** PROCESSING ******************* */
/* this does the actual pitch shifting */
for (int zero = 0; zero < fftFrameSize; zero++) {
gSynMagn[zero] = 0;
gSynFreq[zero] = 0;
}
for (k = 0; k <= fftFrameSize2; k++) {
index = (long) (k * pitchShift);
if (index <= fftFrameSize2) {
gSynMagn[(int) index] += gAnaMagn[(int) k];
gSynFreq[(int) index] = gAnaFreq[(int) k] * pitchShift;
}
}
/* ***************** SYNTHESIS ******************* */
/* this is the synthesis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* get magnitude and true frequency from synthesis arrays */
magn = gSynMagn[(int) k];
tmp = gSynFreq[(int) 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[(int) k] += (float) tmp;
phase = gSumPhase[(int) k];
/* get real and imag part and re-interleave */
gFFTworksp[(int) (2 * k)] = (float) (magn * Math.cos(phase));
gFFTworksp[(int) (2 * k + 1)] = (float) (magn * Math.sin(phase));
}
/* zero negative frequencies */
for (k = fftFrameSize + 2; k < 2 * fftFrameSize; k++)
gFFTworksp[(int) k] = 0.0F;
/* 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[(int) k] += (float) (2.0 * window * gFFTworksp[(int) (2 * k)] / (fftFrameSize2 * osamp));
}
for (k = 0; k < stepSize; k++)
gOutFIFO[(int) k] = gOutputAccum[(int) k];
/* shift accumulator */
// memmove(gOutputAccum, gOutputAccum + stepSize, fftFrameSize *
// sizeof(float));
for (k = 0; k < fftFrameSize; k++) {
gOutputAccum[(int) k] = gOutputAccum[(int) (k + stepSize)];
}
/* move input FIFO */
for (k = 0; k < inFifoLatency; k++)
gInFIFO[(int) k] = gInFIFO[(int) (k + stepSize)];
}
}
}
public static void ShortTimeFourierTransform(float[] fftBuffer, long fftFrameSize, long sign) {
float wr, wi, arg, temp;
float tr, ti, ur, ui;
long 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[(int) i];
fftBuffer[(int) i] = fftBuffer[(int) j];
fftBuffer[(int) j] = temp;
temp = fftBuffer[(int) (i + 1)];
fftBuffer[(int) (i + 1)] = fftBuffer[(int) (j + 1)];
fftBuffer[(int) (j + 1)] = temp;
// 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 max = (long) (MathUtilities.log(fftFrameSize) / MathUtilities.log(2.0) + .5);
for (k = 0, le = 2; k < max; k++) {
le <<= 1;
le2 = le >> 1;
ur = 1.0F;
ui = 0.0F;
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[(int) (i + le2)] * ur - fftBuffer[(int) (i + le2 + 1)] * ui;
ti = fftBuffer[(int) (i + le2)] * ui + fftBuffer[(int) (i + le2 + 1)] * ur;
fftBuffer[(int) (i + le2)] = fftBuffer[(int) i] - tr;
fftBuffer[(int) (i + le2 + 1)] = fftBuffer[(int) (i + 1)] - ti;
fftBuffer[(int) i] += tr;
fftBuffer[(int) (i + 1)] += ti;
// 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;
}
}
}
}
}
我从声音文件中获取字节数组，将其转换为float数组并传递给PitchShift2（）和 之后，我将浮点数组转换为字节数组，从字节数组形成一个流并将其传递给播放器。 但它给出了一个例外“不支持的文件格式”。
在将字节转换为浮点数时我也处理了字节顺序，反之亦然。
任何人都可以告诉我如何正确使用这个课程。