答案 0 :(得分：6)
通常，该算法称为阶段声码器 - 在互联网上搜索该算法应该让您开始。
有一些开源阶段声码器，您应该也可以使用它们作为参考。
您可以实时进行相位声码器 - 使用的主要组件是FFT，因此您需要快速FFT。 Android库可以为您执行此操作，请参阅此文档：http://developer.android.com/reference/android/media/audiofx/Visualizer.html
碰巧，我即将发布ARM的开源FFT，它比Apple的vDSP库（迄今为止最快）更快。我会在几天后将其上传到github.com。
祝你好运。答案 1 :(得分：5)
Android SDK中没有内置音高变换算法。你必须编写自己的代码。音高变换是一种真正的硬核DSP算法;良好的声音算法是数月或更长时间的发展结果......
我个人不知道任何Java实现，所以我建议你采用一些免费的C ++ PS算法，我在音频应用程序中使用的最好的算法是SoundTouch：
http://www.surina.net/soundtouch/
我使用了它的代码，看起来用Java重写代码并不会太复杂。
答案 2 :(得分：0)
HOME网址：http://www.dspdimension.com
public class AudioPitch{
//region Private Static Memebers
private static int MAX_FRAME_LENGTH = 8192;
private static double M_PI = 3.14159265358979323846;
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;
//endregion
public static void PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize/*(long)2048*/, long osamp/*(long)10*/, 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 * M_PI * (double)stepSize / (double)fftFrameSize;
inFifoLatency = 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[(int) gRover] = indata[(int) i];
outdata[(int) i] = gOutFIFO[(int) (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 * M_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 = smbAtan2(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 / M_PI);
if (qpd >= 0) qpd += qpd & 1;
else qpd -= qpd & 1;
tmp -= M_PI * (double)qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp * tmp / (2.0 * M_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 * M_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 * M_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)];
}
}
}
//endregion
//region Private Static Methods
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;
}
}
long max = (long)(Math.log(fftFrameSize) / Math.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)M_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 = ur * wr - ui * wi;
ui = ur * wi + ui * wr;
ur = tr;
}
}
}
//endregion
private static double smbAtan2(double x, double y)
{
double signx;
if (x > 0.) signx = 1.;
else signx = -1.;
if (x == 0.) return 0.;
if (y == 0.) return signx * M_PI / 2.;
return Math.atan2(x, y);
}
}
此代码也可以工作，但消耗CPU使用率很高。
pitchShift在0.5-2.0之间
如下所示调用此类：
int maxValueOFShort = 32768;
short [] buffer = new short[800];
float[] inData = new float[buffer.length];
while (audiorackIsRun)
{
int m = recorder.read(buffer, 0, buffer.length);
for(int n=0; n<buffer.length;n++)
inData[n] = buffer[n]/(float)maxValueOFShort;
AudioPitch.PitchShift(1, buffer.length, 4096, 4, 44100, inData);
for(int n=0; n<buffer.length;n++)
buffer[n] = (short)(inData[n]*maxValueOFShort);
player.write(buffer, 0, buffer.length);
}