VS语音信号处理（3） C语言调用sonic进行变速不变调工程实例

前言

语音识别相关算法一般在MATLAB上进行仿真验证与实验，在工程上一般还是在VS中进行实现落地，本系列将介绍语音信号处理在C语言中的一系列应用，后期将以此为基础，再落地移植到嵌入式平台。

今天介绍sonic语音变速不变调算法的工程应用。libsonic是一个支持音频倍速播放的开源库。支持大于2倍速的播放。有ANSI C的版本，也有纯Java的版本。无外部依赖能够方便的集成进自己的应用中。

一. 工程实现

打开VS2015

点击新建一个项目（工程）

点击“确定”，创建项目

左侧解决方案资源管理器视图中，头文件栏右键添加新建项，

点击头文件，新建一个头文件sonic.h，点击“添加”

输入代码如下：

/* Sonic library
Copyright 2010
Bill Cox
This file is part of the Sonic Library.
This file is licensed under the Apache 2.0 license.
*/

/*
The Sonic Library implements a new algorithm invented by Bill Cox for the
specific purpose of speeding up speech by high factors at high quality.  It
generates smooth speech at speed up factors as high as 6X, possibly more.  It is
also capable of slowing down speech, and generates high quality results
regardless of the speed up or slow down factor.  For speeding up speech by 2X or
more, the following equation is used:
newSamples = period/(speed - 1.0)
scale = 1.0/newSamples;
where period is the current pitch period, determined using AMDF or any other
pitch estimator, and speed is the speedup factor.  If the current position in
the input stream is pointed to by "samples", and the current output stream
position is pointed to by "out", then newSamples number of samples can be
generated with:
out[t] = (samples[t]*(newSamples - t) + samples[t + period]*t)/newSamples;
where t = 0 to newSamples - 1.
For speed factors < 2X, the PICOLA algorithm is used.  The above
algorithm is first used to double the speed of one pitch period.  Then, enough
input is directly copied from the input to the output to achieve the desired
speed up factor, where 1.0 < speed < 2.0.  The amount of data copied is derived:
speed = (2*period + length)/(period + length)
speed*length + speed*period = 2*period + length
length(speed - 1) = 2*period - speed*period
length = period*(2 - speed)/(speed - 1)
For slowing down speech where 0.5 < speed < 1.0, a pitch period is inserted into
the output twice, and length of input is copied from the input to the output
until the output desired speed is reached.  The length of data copied is:
length = period*(speed - 0.5)/(1 - speed)
For slow down factors below 0.5, no data is copied, and an algorithm
similar to high speed factors is used.
*/

/* Uncomment this to use sin-wav based overlap add which in theory can improve
sound quality slightly, at the expense of lots of floating point math. */
/* #define SONIC_USE_SIN */

#ifdef __cplusplus
extern "C" {
#endif

	/* This specifies the range of voice pitches we try to match.
	Note that if we go lower than 65, we could overflow in findPitchInRange */
#define SONIC_MIN_PITCH 65
#define SONIC_MAX_PITCH 400

	/* These are used to down-sample some inputs to improve speed */
#define SONIC_AMDF_FREQ 4000

	struct sonicStreamStruct;
	typedef struct sonicStreamStruct* sonicStream;

	/* For all of the following functions, numChannels is multiplied by numSamples
	to determine the actual number of values read or returned. */

	/* Create a sonic stream.  Return NULL only if we are out of memory and cannot
	allocate the stream. Set numChannels to 1 for mono, and 2 for stereo. */
	sonicStream sonicCreateStream(int sampleRate, int numChannels);
	/* Destroy the sonic stream. */
	void sonicDestroyStream(sonicStream stream);
	/* Use this to write floating point data to be speed up or down into the stream.
	Values must be between -1 and 1.  Return 0 if memory realloc failed,
	otherwise 1 */
	int sonicWriteFloatToStream(sonicStream stream, float* samples, int numSamples);
	/* Use this to write 16-bit data to be speed up or down into the stream.
	Return 0 if memory realloc failed, otherwise 1 */
	int sonicWriteShortToStream(sonicStream stream, short* samples, int numSamples);
	/* Use this to write 8-bit unsigned data to be speed up or down into the stream.
	Return 0 if memory realloc failed, otherwise 1 */
	int sonicWriteUnsignedCharToStream(sonicStream stream, unsigned char* samples,
		int numSamples);
	/* Use this to read floating point data out of the stream.  Sometimes no data
	will be available, and zero is returned, which is not an error condition. */
	int sonicReadFloatFromStream(sonicStream stream, float* samples,
		int maxSamples);
	/* Use this to read 16-bit data out of the stream.  Sometimes no data will
	be available, and zero is returned, which is not an error condition. */
	int sonicReadShortFromStream(sonicStream stream, short* samples,
		int maxSamples);
	/* Use this to read 8-bit unsigned data out of the stream.  Sometimes no data
	will be available, and zero is returned, which is not an error condition. */
	int sonicReadUnsignedCharFromStream(sonicStream stream, unsigned char* samples,
		int maxSamples);
	/* Force the sonic stream to generate output using whatever data it currently
	has.  No extra delay will be added to the output, but flushing in the middle
	of words could introduce distortion. */
	int sonicFlushStream(sonicStream stream);
	/* Return the number of samples in the output buffer */
	int sonicSamplesAvailable(sonicStream stream);
	/* Get the speed of the stream. */
	float sonicGetSpeed(sonicStream stream);
	/* Set the speed of the stream. */
	void sonicSetSpeed(sonicStream stream, float speed);
	/* Get the pitch of the stream. */
	float sonicGetPitch(sonicStream stream);
	/* Set the pitch of the stream. */
	void sonicSetPitch(sonicStream stream, float pitch);
	/* Get the rate of the stream. */
	float sonicGetRate(sonicStream stream);
	/* Set the rate of the stream. */
	void sonicSetRate(sonicStream stream, float rate);
	/* Get the scaling factor of the stream. */
	float sonicGetVolume(sonicStream stream);
	/* Set the scaling factor of the stream. */
	void sonicSetVolume(sonicStream stream, float volume);
	/* Get the chord pitch setting. */
	int sonicGetChordPitch(sonicStream stream);
	/* Set chord pitch mode on or off.  Default is off.  See the documentation
	page for a description of this feature. */
	void sonicSetChordPitch(sonicStream stream, int useChordPitch);
	/* Get the quality setting. */
	int sonicGetQuality(sonicStream stream);
	/* Set the "quality".  Default 0 is virtually as good as 1, but very much
	* faster. */
	void sonicSetQuality(sonicStream stream, int quality);
	/* Get the sample rate of the stream. */
	int sonicGetSampleRate(sonicStream stream);
	/* Set the sample rate of the stream.  This will drop any samples that have not
	* been read. */
	void sonicSetSampleRate(sonicStream stream, int sampleRate);
	/* Get the number of channels. */
	int sonicGetNumChannels(sonicStream stream);
	/* Set the number of channels.  This will drop any samples that have not been
	* read. */
	void sonicSetNumChannels(sonicStream stream, int numChannels);
	/* This is a non-stream oriented interface to just change the speed of a sound
	sample.  It works in-place on the sample array, so there must be at least
	speed*numSamples available space in the array. Returns the new number of
	samples. */
	int sonicChangeFloatSpeed(float* samples, int numSamples, float speed,
		float pitch, float rate, float volume,
		int useChordPitch, int sampleRate, int numChannels);
	/* This is a non-stream oriented interface to just change the speed of a sound
	sample.  It works in-place on the sample array, so there must be at least
	speed*numSamples available space in the array. Returns the new number of
	samples. */
	int sonicChangeShortSpeed(short* samples, int numSamples, float speed,
		float pitch, float rate, float volume,
		int useChordPitch, int sampleRate, int numChannels);

#ifdef SONIC_SPECTROGRAM
	/*
	This code generates high quality spectrograms from sound samples, using
	Time-Aliased-FFTs as described at:
	https://github.com/waywardgeek/spectrogram
	Basically, two adjacent pitch periods are overlap-added to create a sound
	sample that accurately represents the speech sound at that moment in time.
	This set of samples is converted to a spetral line using an FFT, and the result
	is saved as a single spectral line at that moment in time.  The resulting
	spectral lines vary in resolution (it is equal to the number of samples in the
	pitch period), and the spacing of spectral lines also varies (proportional to
	the numver of samples in the pitch period).
	To generate a bitmap, linear interpolation is used to render the grayscale
	value at any particular point in time and frequency.
	*/

#define SONIC_MAX_SPECTRUM_FREQ 5000

	struct sonicSpectrogramStruct;
	struct sonicBitmapStruct;
	typedef struct sonicSpectrogramStruct* sonicSpectrogram;
	typedef struct sonicBitmapStruct* sonicBitmap;

	/* sonicBitmap objects represent spectrograms as grayscale bitmaps where each
	pixel is from 0 (black) to 255 (white).  Bitmaps are rows*cols in size.
	Rows are indexed top to bottom and columns are indexed left to right */
	struct sonicBitmapStruct {
		unsigned char* data;
		int numRows;
		int numCols;
	};

	typedef struct sonicBitmapStruct* sonicBitmap;

	/* Enable coomputation of a spectrogram on the fly. */
	void sonicComputeSpectrogram(sonicStream stream);

	/* Get the spectrogram. */
	sonicSpectrogram sonicGetSpectrogram(sonicStream stream);

	/* Create an empty spectrogram. Called automatically if sonicComputeSpectrogram
	has been called. */
	sonicSpectrogram sonicCreateSpectrogram(int sampleRate);

	/* Destroy the spectrotram.  This is called automatically when calling
	sonicDestroyStream. */
	void sonicDestroySpectrogram(sonicSpectrogram spectrogram);

	/* Convert the spectrogram to a bitmap. Caller must destroy bitmap when done. */
	sonicBitmap sonicConvertSpectrogramToBitmap(sonicSpectrogram spectrogram,
		int numRows, int numCols);

	/* Destroy a bitmap returned by sonicConvertSpectrogramToBitmap. */
	void sonicDestroyBitmap(sonicBitmap bitmap);

	int sonicWritePGM(sonicBitmap bitmap, char* fileName);

	/* Add two pitch periods worth of samples to the spectrogram.  There must be
	2*period samples.  Time should advance one pitch period for each call to
	this function. */
	void sonicAddPitchPeriodToSpectrogram(sonicSpectrogram spectrogram,
		short* samples, int period,
		int numChannels);
#endif  /* SONIC_SPECTROGRAM */

#ifdef __cplusplus
}
#endif

左侧解决方案资源管理器视图中，源文件栏右键添加新建项，

点击C++文件，新建一个源文件sonic.cpp，点击“添加”

输入代码如下：

/* Sonic library
Copyright 2010
Bill Cox
This file is part of the Sonic Library.
This file is licensed under the Apache 2.0 license.
*/

#include "sonic.h"

#include <limits.h>
#include <math.h>
#include <stdarg.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>

/*
The following code was used to generate the following sinc lookup table.
#include <limits.h>
#include <math.h>
#include <stdio.h>
double findHannWeight(int N, double x) {
return 0.5*(1.0 - cos(2*M_PI*x/N));
}
double findSincCoefficient(int N, double x) {
double hannWindowWeight = findHannWeight(N, x);
double sincWeight;
x -= N/2.0;
if (x > 1e-9 || x < -1e-9) {
sincWeight = sin(M_PI*x)/(M_PI*x);
} else {
sincWeight = 1.0;
}
return hannWindowWeight*sincWeight;
}
int main() {
double x;
int i;
int N = 12;
for (i = 0, x = 0.0; x <= N; x += 0.02, i++) {
printf("%u %d\n", i, (int)(SHRT_MAX*findSincCoefficient(N, x)));
}
return 0;
}
*/

/* The number of points to use in the sinc FIR filter for resampling. */
#define SINC_FILTER_POINTS \
  12 /* I am not able to hear improvement with higher N. */
#define SINC_TABLE_SIZE 601

/* Lookup table for windowed sinc function of SINC_FILTER_POINTS points. */
static short sincTable[SINC_TABLE_SIZE] = {
	0,     0,     0,     0,     0,     0,     0,     -1,    -1,    -2,    -2,
	-3,    -4,    -6,    -7,    -9,    -10,   -12,   -14,   -17,   -19,   -21,
	-24,   -26,   -29,   -32,   -34,   -37,   -40,   -42,   -44,   -47,   -48,
	-50,   -51,   -52,   -53,   -53,   -53,   -52,   -50,   -48,   -46,   -43,
	-39,   -34,   -29,   -22,   -16,   -8,    0,     9,     19,    29,    41,
	53,    65,    79,    92,    107,   121,   137,   152,   168,   184,   200,
	215,   231,   247,   262,   276,   291,   304,   317,   328,   339,   348,
	357,   363,   369,   372,   374,   375,   373,   369,   363,   355,   345,
	332,   318,   300,   281,   259,   234,   208,   178,   147,   113,   77,
	39,    0,     -41,   -85,   -130,  -177,  -225,  -274,  -324,  -375,  -426,
	-478,  -530,  -581,  -632,  -682,  -731,  -779,  -825,  -870,  -912,  -951,
	-989,  -1023, -1053, -1080, -1104, -1123, -1138, -1149, -1154, -1155, -1151,
	-1141, -1125, -1105, -1078, -1046, -1007, -963,  -913,  -857,  -796,  -728,
	-655,  -576,  -492,  -403,  -309,  -210,  -107,  0,     111,   225,   342,
	462,   584,   708,   833,   958,   1084,  1209,  1333,  1455,  1575,  1693,
	1807,  1916,  2022,  2122,  2216,  2304,  2384,  2457,  2522,  2579,  2625,
	2663,  2689,  2706,  2711,  2705,  2687,  2657,  2614,  2559,  2491,  2411,
	2317,  2211,  2092,  1960,  1815,  1658,  1489,  1308,  1115,  912,   698,
	474,   241,   0,     -249,  -506,  -769,  -1037, -1310, -1586, -1864, -2144,
	-2424, -2703, -2980, -3254, -3523, -3787, -4043, -4291, -4529, -4757, -4972,
	-5174, -5360, -5531, -5685, -5819, -5935, -6029, -6101, -6150, -6175, -6175,
	-6149, -6096, -6015, -5905, -5767, -5599, -5401, -5172, -4912, -4621, -4298,
	-3944, -3558, -3141, -2693, -2214, -1705, -1166, -597,  0,     625,   1277,
	1955,  2658,  3386,  4135,  4906,  5697,  6506,  7332,  8173,  9027,  9893,
	10769, 11654, 12544, 13439, 14335, 15232, 16128, 17019, 17904, 18782, 19649,
	20504, 21345, 22170, 22977, 23763, 24527, 25268, 25982, 26669, 27327, 27953,
	28547, 29107, 29632, 30119, 30569, 30979, 31349, 31678, 31964, 32208, 32408,
	32565, 32677, 32744, 32767, 32744, 32677, 32565, 32408, 32208, 31964, 31678,
	31349, 30979, 30569, 30119, 29632, 29107, 28547, 27953, 27327, 26669, 25982,
	25268, 24527, 23763, 22977, 22170, 21345, 20504, 19649, 18782, 17904, 17019,
	16128, 15232, 14335, 13439, 12544, 11654, 10769, 9893,  9027,  8173,  7332,
	6506,  5697,  4906,  4135,  3386,  2658,  1955,  1277,  625,   0,     -597,
	-1166, -1705, -2214, -2693, -3141, -3558, -3944, -4298, -4621, -4912, -5172,
	-5401, -5599, -5767, -5905, -6015, -6096, -6149, -6175, -6175, -6150, -6101,
	-6029, -5935, -5819, -5685, -5531, -5360, -5174, -4972, -4757, -4529, -4291,
	-4043, -3787, -3523, -3254, -2980, -2703, -2424, -2144, -1864, -1586, -1310,
	-1037, -769,  -506,  -249,  0,     241,   474,   698,   912,   1115,  1308,
	1489,  1658,  1815,  1960,  2092,  2211,  2317,  2411,  2491,  2559,  2614,
	2657,  2687,  2705,  2711,  2706,  2689,  2663,  2625,  2579,  2522,  2457,
	2384,  2304,  2216,  2122,  2022,  1916,  1807,  1693,  1575,  1455,  1333,
	1209,  1084,  958,   833,   708,   584,   462,   342,   225,   111,   0,
	-107,  -210,  -309,  -403,  -492,  -576,  -655,  -728,  -796,  -857,  -913,
	-963,  -1007, -1046, -1078, -1105, -1125, -1141, -1151, -1155, -1154, -1149,
	-1138, -1123, -1104, -1080, -1053, -1023, -989,  -951,  -912,  -870,  -825,
	-779,  -731,  -682,  -632,  -581,  -530,  -478,  -426,  -375,  -324,  -274,
	-225,  -177,  -130,  -85,   -41,   0,     39,    77,    113,   147,   178,
	208,   234,   259,   281,   300,   318,   332,   345,   355,   363,   369,
	373,   375,   374,   372,   369,   363,   357,   348,   339,   328,   317,
	304,   291,   276,   262,   247,   231,   215,   200,   184,   168,   152,
	137,   121,   107,   92,    79,    65,    53,    41,    29,    19,    9,
	0,     -8,    -16,   -22,   -29,   -34,   -39,   -43,   -46,   -48,   -50,
	-52,   -53,   -53,   -53,   -52,   -51,   -50,   -48,   -47,   -44,   -42,
	-40,   -37,   -34,   -32,   -29,   -26,   -24,   -21,   -19,   -17,   -14,
	-12,   -10,   -9,    -7,    -6,    -4,    -3,    -2,    -2,    -1,    -1,
	0,     0,     0,     0,     0,     0,     0 };

struct sonicStreamStruct {
#ifdef SONIC_SPECTROGRAM
	sonicSpectrogram spectrogram;
#endif  /* SONIC_SPECTROGRAM */
	short* inputBuffer;
	short* outputBuffer;
	short* pitchBuffer;
	short* downSampleBuffer;
	float speed;
	float volume;
	float pitch;
	float rate;
	int oldRatePosition;
	int newRatePosition;
	int useChordPitch;
	int quality;
	int numChannels;
	int inputBufferSize;
	int pitchBufferSize;
	int outputBufferSize;
	int numInputSamples;
	int numOutputSamples;
	int numPitchSamples;
	int minPeriod;
	int maxPeriod;
	int maxRequired;
	int remainingInputToCopy;
	int sampleRate;
	int prevPeriod;
	int prevMinDiff;
	float avePower;
};

#ifdef SONIC_SPECTROGRAM

/* Compute a spectrogram on the fly. */
void sonicComputeSpectrogram(sonicStream stream) {
	stream->spectrogram = sonicCreateSpectrogram(stream->sampleRate);
	/* Force changeSpeed to be called to compute the spectrogram. */
	sonicSetSpeed(stream, 2.0);
}

/* Get the spectrogram. */
sonicSpectrogram sonicGetSpectrogram(sonicStream stream) {
	return stream->spectrogram;
}

#endif

/* Scale the samples by the factor. */
static void scaleSamples(short* samples, int numSamples, float volume) {
	int fixedPointVolume = volume * 4096.0f;
	int value;

	while (numSamples--) {
		value = (*samples * fixedPointVolume) >> 12;
		if (value > 32767) {
			value = 32767;
		}
		else if (value < -32767) {
			value = -32767;
		}
		*samples++ = value;
	}
}

/* Get the speed of the stream. */
float sonicGetSpeed(sonicStream stream) { return stream->speed; }

/* Set the speed of the stream. */
void sonicSetSpeed(sonicStream stream, float speed) { stream->speed = speed; }

/* Get the pitch of the stream. */
float sonicGetPitch(sonicStream stream) { return stream->pitch; }

/* Set the pitch of the stream. */
void sonicSetPitch(sonicStream stream, float pitch) { stream->pitch = pitch; }

/* Get the rate of the stream. */
float sonicGetRate(sonicStream stream) { return stream->rate; }

/* Set the playback rate of the stream. This scales pitch and speed at the same
time. */
void sonicSetRate(sonicStream stream, float rate) {
	stream->rate = rate;

	stream->oldRatePosition = 0;
	stream->newRatePosition = 0;
}

/* Get the vocal chord pitch setting. */
int sonicGetChordPitch(sonicStream stream) { return stream->useChordPitch; }

/* Set the vocal chord mode for pitch computation.  Default is off. */
void sonicSetChordPitch(sonicStream stream, int useChordPitch) {
	stream->useChordPitch = useChordPitch;
}

/* Get the quality setting. */
int sonicGetQuality(sonicStream stream) { return stream->quality; }

/* Set the "quality".  Default 0 is virtually as good as 1, but very much
faster. */
void sonicSetQuality(sonicStream stream, int quality) {
	stream->quality = quality;
}

/* Get the scaling factor of the stream. */
float sonicGetVolume(sonicStream stream) { return stream->volume; }

/* Set the scaling factor of the stream. */
void sonicSetVolume(sonicStream stream, float volume) {
	stream->volume = volume;
}

/* Free stream buffers. */
static void freeStreamBuffers(sonicStream stream) {
	if (stream->inputBuffer != NULL) {
		free(stream->inputBuffer);
	}
	if (stream->outputBuffer != NULL) {
		free(stream->outputBuffer);
	}
	if (stream->pitchBuffer != NULL) {
		free(stream->pitchBuffer);
	}
	if (stream->downSampleBuffer != NULL) {
		free(stream->downSampleBuffer);
	}
}

/* Destroy the sonic stream. */
void sonicDestroyStream(sonicStream stream) {
#ifdef SONIC_SPECTROGRAM
	if (stream->spectrogram != NULL) {
		sonicDestroySpectrogram(stream->spectrogram);
	}
#endif  /* SONIC_SPECTROGRAM */
	freeStreamBuffers(stream);
	free(stream);
}

/* Allocate stream buffers. */
static int allocateStreamBuffers(sonicStream stream, int sampleRate,
	int numChannels) {
	int minPeriod = sampleRate / SONIC_MAX_PITCH;
	int maxPeriod = sampleRate / SONIC_MIN_PITCH;
	int maxRequired = 2 * maxPeriod;

	stream->inputBufferSize = maxRequired;
	stream->inputBuffer =
		(short*)calloc(maxRequired, sizeof(short) * numChannels);
	if (stream->inputBuffer == NULL) {
		sonicDestroyStream(stream);
		return 0;
	}
	stream->outputBufferSize = maxRequired;
	stream->outputBuffer =
		(short*)calloc(maxRequired, sizeof(short) * numChannels);
	if (stream->outputBuffer == NULL) {
		sonicDestroyStream(stream);
		return 0;
	}
	stream->pitchBufferSize = maxRequired;
	stream->pitchBuffer =
		(short*)calloc(maxRequired, sizeof(short) * numChannels);
	if (stream->pitchBuffer == NULL) {
		sonicDestroyStream(stream);
		return 0;
	}
	stream->downSampleBuffer = (short*)calloc(maxRequired, sizeof(short));
	if (stream->downSampleBuffer == NULL) {
		sonicDestroyStream(stream);
		return 0;
	}
	stream->sampleRate = sampleRate;
	stream->numChannels = numChannels;
	stream->oldRatePosition = 0;
	stream->newRatePosition = 0;
	stream->minPeriod = minPeriod;
	stream->maxPeriod = maxPeriod;
	stream->maxRequired = maxRequired;
	stream->prevPeriod = 0;
	return 1;
}

/* Create a sonic stream.  Return NULL only if we are out of memory and cannot
allocate the stream. */
sonicStream sonicCreateStream(int sampleRate, int numChannels) {
	sonicStream stream = (sonicStream)calloc(1, sizeof(struct sonicStreamStruct));

	if (stream == NULL) {
		return NULL;
	}
	if (!allocateStreamBuffers(stream, sampleRate, numChannels)) {
		return NULL;
	}
	stream->speed = 1.0f;
	stream->pitch = 1.0f;
	stream->volume = 1.0f;
	stream->rate = 1.0f;
	stream->oldRatePosition = 0;
	stream->newRatePosition = 0;
	stream->useChordPitch = 0;
	stream->quality = 0;
	stream->avePower = 50.0f;
	return stream;
}

/* Get the sample rate of the stream. */
int sonicGetSampleRate(sonicStream stream) { return stream->sampleRate; }

/* Set the sample rate of the stream.  This will cause samples buffered in the
stream to be lost. */
void sonicSetSampleRate(sonicStream stream, int sampleRate) {
	freeStreamBuffers(stream);
	allocateStreamBuffers(stream, sampleRate, stream->numChannels);
}

/* Get the number of channels. */
int sonicGetNumChannels(sonicStream stream) { return stream->numChannels; }

/* Set the num channels of the stream.  This will cause samples buffered in the
stream to be lost. */
void sonicSetNumChannels(sonicStream stream, int numChannels) {
	freeStreamBuffers(stream);
	allocateStreamBuffers(stream, stream->sampleRate, numChannels);
}

/* Enlarge the output buffer if needed. */
static int enlargeOutputBufferIfNeeded(sonicStream stream, int numSamples) {
	if (stream->numOutputSamples + numSamples > stream->outputBufferSize) {
		stream->outputBufferSize += (stream->outputBufferSize >> 1) + numSamples;
		stream->outputBuffer = (short*)realloc(
			stream->outputBuffer,
			stream->outputBufferSize * sizeof(short) * stream->numChannels);
		if (stream->outputBuffer == NULL) {
			return 0;
		}
	}
	return 1;
}

/* Enlarge the input buffer if needed. */
static int enlargeInputBufferIfNeeded(sonicStream stream, int numSamples) {
	if (stream->numInputSamples + numSamples > stream->inputBufferSize) {
		stream->inputBufferSize += (stream->inputBufferSize >> 1) + numSamples;
		stream->inputBuffer = (short*)realloc(
			stream->inputBuffer,
			stream->inputBufferSize * sizeof(short) * stream->numChannels);
		if (stream->inputBuffer == NULL) {
			return 0;
		}
	}
	return 1;
}

/* Add the input samples to the input buffer. */
static int addFloatSamplesToInputBuffer(sonicStream stream, float* samples,
	int numSamples) {
	short* buffer;
	int count = numSamples * stream->numChannels;

	if (numSamples == 0) {
		return 1;
	}
	if (!enlargeInputBufferIfNeeded(stream, numSamples)) {
		return 0;
	}
	buffer = stream->inputBuffer + stream->numInputSamples * stream->numChannels;
	while (count--) {
		*buffer++ = (*samples++) * 32767.0f;
	}
	stream->numInputSamples += numSamples;
	return 1;
}

/* Add the input samples to the input buffer. */
static int addShortSamplesToInputBuffer(sonicStream stream, short* samples,
	int numSamples) {
	if (numSamples == 0) {
		return 1;
	}
	if (!enlargeInputBufferIfNeeded(stream, numSamples)) {
		return 0;
	}
	memcpy(stream->inputBuffer + stream->numInputSamples * stream->numChannels,
		samples, numSamples * sizeof(short) * stream->numChannels);
	stream->numInputSamples += numSamples;
	return 1;
}

/* Add the input samples to the input buffer. */
static int addUnsignedCharSamplesToInputBuffer(sonicStream stream,
	unsigned char* samples,
	int numSamples) {
	short* buffer;
	int count = numSamples * stream->numChannels;

	if (numSamples == 0) {
		return 1;
	}
	if (!enlargeInputBufferIfNeeded(stream, numSamples)) {
		return 0;
	}
	buffer = stream->inputBuffer + stream->numInputSamples * stream->numChannels;
	while (count--) {
		*buffer++ = (*samples++ - 128) << 8;
	}
	stream->numInputSamples += numSamples;
	return 1;
}

/* Remove input samples that we have already processed. */
static void removeInputSamples(sonicStream stream, int position) {
	int remainingSamples = stream->numInputSamples - position;

	if (remainingSamples > 0) {
		memmove(stream->inputBuffer,
			stream->inputBuffer + position * stream->numChannels,
			remainingSamples * sizeof(short) * stream->numChannels);
	}
	stream->numInputSamples = remainingSamples;
}

/* Just copy from the array to the output buffer */
static int copyToOutput(sonicStream stream, short* samples, int numSamples) {
	if (!enlargeOutputBufferIfNeeded(stream, numSamples)) {
		return 0;
	}
	memcpy(stream->outputBuffer + stream->numOutputSamples * stream->numChannels,
		samples, numSamples * sizeof(short) * stream->numChannels);
	stream->numOutputSamples += numSamples;
	return 1;
}

/* Just copy from the input buffer to the output buffer.  Return 0 if we fail to
resize the output buffer.  Otherwise, return numSamples */
static int copyInputToOutput(sonicStream stream, int position) {
	int numSamples = stream->remainingInputToCopy;

	if (numSamples > stream->maxRequired) {
		numSamples = stream->maxRequired;
	}
	if (!copyToOutput(stream,
		stream->inputBuffer + position * stream->numChannels,
		numSamples)) {
		return 0;
	}
	stream->remainingInputToCopy -= numSamples;
	return numSamples;
}

/* Read data out of the stream.  Sometimes no data will be available, and zero
is returned, which is not an error condition. */
int sonicReadFloatFromStream(sonicStream stream, float* samples,
	int maxSamples) {
	int numSamples = stream->numOutputSamples;
	int remainingSamples = 0;
	short* buffer;
	int count;

	if (numSamples == 0) {
		return 0;
	}
	if (numSamples > maxSamples) {
		remainingSamples = numSamples - maxSamples;
		numSamples = maxSamples;
	}
	buffer = stream->outputBuffer;
	count = numSamples * stream->numChannels;
	while (count--) {
		*samples++ = (*buffer++) / 32767.0f;
	}
	if (remainingSamples > 0) {
		memmove(stream->outputBuffer,
			stream->outputBuffer + numSamples * stream->numChannels,
			remainingSamples * sizeof(short) * stream->numChannels);
	}
	stream->numOutputSamples = remainingSamples;
	return numSamples;
}

/* Read short data out of the stream.  Sometimes no data will be available, and
zero is returned, which is not an error condition. */
int sonicReadShortFromStream(sonicStream stream, short* samples,
	int maxSamples) {
	int numSamples = stream->numOutputSamples;
	int remainingSamples = 0;

	if (numSamples == 0) {
		return 0;
	}
	if (numSamples > maxSamples) {
		remainingSamples = numSamples - maxSamples;
		numSamples = maxSamples;
	}
	memcpy(samples, stream->outputBuffer,
		numSamples * sizeof(short) * stream->numChannels);
	if (remainingSamples > 0) {
		memmove(stream->outputBuffer,
			stream->outputBuffer + numSamples * stream->numChannels,
			remainingSamples * sizeof(short) * stream->numChannels);
	}
	stream->numOutputSamples = remainingSamples;
	return numSamples;
}

/* Read unsigned char data out of the stream.  Sometimes no data will be
available, and zero is returned, which is not an error condition. */
int sonicReadUnsignedCharFromStream(sonicStream stream, unsigned char* samples,
	int maxSamples) {
	int numSamples = stream->numOutputSamples;
	int remainingSamples = 0;
	short* buffer;
	int count;

	if (numSamples == 0) {
		return 0;
	}
	if (numSamples > maxSamples) {
		remainingSamples = numSamples - maxSamples;
		numSamples = maxSamples;
	}
	buffer = stream->outputBuffer;
	count = numSamples * stream->numChannels;
	while (count--) {
		*samples++ = (char)((*buffer++) >> 8) + 128;
	}
	if (remainingSamples > 0) {
		memmove(stream->outputBuffer,
			stream->outputBuffer + numSamples * stream->numChannels,
			remainingSamples * sizeof(short) * stream->numChannels);
	}
	stream->numOutputSamples = remainingSamples;
	return numSamples;
}

/* Force the sonic stream to generate output using whatever data it currently
has.  No extra delay will be added to the output, but flushing in the middle
of words could introduce distortion. */
int sonicFlushStream(sonicStream stream) {
	int maxRequired = stream->maxRequired;
	int remainingSamples = stream->numInputSamples;
	float speed = stream->speed / stream->pitch;
	float rate = stream->rate * stream->pitch;
	int expectedOutputSamples =
		stream->numOutputSamples +
		(int)((remainingSamples / speed + stream->numPitchSamples) / rate + 0.5f);

	/* Add enough silence to flush both input and pitch buffers. */
	if (!enlargeInputBufferIfNeeded(stream, remainingSamples + 2 * maxRequired)) {
		return 0;
	}
	memset(stream->inputBuffer + remainingSamples * stream->numChannels, 0,
		2 * maxRequired * sizeof(short) * stream->numChannels);
	stream->numInputSamples += 2 * maxRequired;
	if (!sonicWriteShortToStream(stream, NULL, 0)) {
		return 0;
	}
	/* Throw away any extra samples we generated due to the silence we added */
	if (stream->numOutputSamples > expectedOutputSamples) {
		stream->numOutputSamples = expectedOutputSamples;
	}
	/* Empty input and pitch buffers */
	stream->numInputSamples = 0;
	stream->remainingInputToCopy = 0;
	stream->numPitchSamples = 0;
	return 1;
}

/* Return the number of samples in the output buffer */
int sonicSamplesAvailable(sonicStream stream) {
	return stream->numOutputSamples;
}

/* If skip is greater than one, average skip samples together and write them to
the down-sample buffer.  If numChannels is greater than one, mix the channels
together as we down sample. */
static void downSampleInput(sonicStream stream, short* samples, int skip) {
	int numSamples = stream->maxRequired / skip;
	int samplesPerValue = stream->numChannels * skip;
	int i, j;
	int value;
	short* downSamples = stream->downSampleBuffer;

	for (i = 0; i < numSamples; i++) {
		value = 0;
		for (j = 0; j < samplesPerValue; j++) {
			value += *samples++;
		}
		value /= samplesPerValue;
		*downSamples++ = value;
	}
}

/* Find the best frequency match in the range, and given a sample skip multiple.
For now, just find the pitch of the first channel. */
static int findPitchPeriodInRange(short* samples, int minPeriod, int maxPeriod,
	int* retMinDiff, int* retMaxDiff) {
	int period, bestPeriod = 0, worstPeriod = 255;
	short* s;
	short* p;
	short sVal, pVal;
	unsigned long diff, minDiff = 1, maxDiff = 0;
	int i;

	for (period = minPeriod; period <= maxPeriod; period++) {
		diff = 0;
		s = samples;
		p = samples + period;
		for (i = 0; i < period; i++) {
			sVal = *s++;
			pVal = *p++;
			diff += sVal >= pVal ? (unsigned short)(sVal - pVal)
				: (unsigned short)(pVal - sVal);
		}
		/* Note that the highest number of samples we add into diff will be less
		than 256, since we skip samples.  Thus, diff is a 24 bit number, and
		we can safely multiply by numSamples without overflow */
		/* if (bestPeriod == 0 || (bestPeriod*3/2 > period && diff*bestPeriod <
		minDiff*period) || diff*bestPeriod < (minDiff >> 1)*period) {*/
		if (bestPeriod == 0 || diff * bestPeriod < minDiff * period) {
			minDiff = diff;
			bestPeriod = period;
		}
		if (diff * worstPeriod > maxDiff * period) {
			maxDiff = diff;
			worstPeriod = period;
		}
	}
	*retMinDiff = minDiff / bestPeriod;
	*retMaxDiff = maxDiff / worstPeriod;
	return bestPeriod;
}

/* At abrupt ends of voiced words, we can have pitch periods that are better
approximated by the previous pitch period estimate.  Try to detect this case.
*/
static int prevPeriodBetter(sonicStream stream, int period, int minDiff,
	int maxDiff, int preferNewPeriod) {
	if (minDiff == 0 || stream->prevPeriod == 0) {
		return 0;
	}
	if (preferNewPeriod) {
		if (maxDiff > minDiff * 3) {
			/* Got a reasonable match this period */
			return 0;
		}
		if (minDiff * 2 <= stream->prevMinDiff * 3) {
			/* Mismatch is not that much greater this period */
			return 0;
		}
	}
	else {
		if (minDiff <= stream->prevMinDiff) {
			return 0;
		}
	}
	return 1;
}

/* Find the pitch period.  This is a critical step, and we may have to try
multiple ways to get a good answer.  This version uses Average Magnitude
Difference Function (AMDF).  To improve speed, we down sample by an integer
factor get in the 11KHz range, and then do it again with a narrower
frequency range without down sampling */
static int findPitchPeriod(sonicStream stream, short* samples,
	int preferNewPeriod) {
	int minPeriod = stream->minPeriod;
	int maxPeriod = stream->maxPeriod;
	int sampleRate = stream->sampleRate;
	int minDiff, maxDiff, retPeriod;
	int skip = 1;
	int period;

	if (sampleRate > SONIC_AMDF_FREQ && stream->quality == 0) {
		skip = sampleRate / SONIC_AMDF_FREQ;
	}
	if (stream->numChannels == 1 && skip == 1) {
		period = findPitchPeriodInRange(samples, minPeriod, maxPeriod, &minDiff,
			&maxDiff);
	}
	else {
		downSampleInput(stream, samples, skip);
		period = findPitchPeriodInRange(stream->downSampleBuffer, minPeriod / skip,
			maxPeriod / skip, &minDiff, &maxDiff);
		if (skip != 1) {
			period *= skip;
			minPeriod = period - (skip << 2);
			maxPeriod = period + (skip << 2);
			if (minPeriod < stream->minPeriod) {
				minPeriod = stream->minPeriod;
			}
			if (maxPeriod > stream->maxPeriod) {
				maxPeriod = stream->maxPeriod;
			}
			if (stream->numChannels == 1) {
				period = findPitchPeriodInRange(samples, minPeriod, maxPeriod, &minDiff,
					&maxDiff);
			}
			else {
				downSampleInput(stream, samples, 1);
				period = findPitchPeriodInRange(stream->downSampleBuffer, minPeriod,
					maxPeriod, &minDiff, &maxDiff);
			}
		}
	}
	if (prevPeriodBetter(stream, period, minDiff, maxDiff, preferNewPeriod)) {
		retPeriod = stream->prevPeriod;
	}
	else {
		retPeriod = period;
	}
	stream->prevMinDiff = minDiff;
	stream->prevPeriod = period;
	return retPeriod;
}

/* Overlap two sound segments, ramp the volume of one down, while ramping the
other one from zero up, and add them, storing the result at the output. */
static void overlapAdd(int numSamples, int numChannels, short* out,
	short* rampDown, short* rampUp) {
	short* o;
	short* u;
	short* d;
	int i, t;

	for (i = 0; i < numChannels; i++) {
		o = out + i;
		u = rampUp + i;
		d = rampDown + i;
		for (t = 0; t < numSamples; t++) {
#ifdef SONIC_USE_SIN
			float ratio = sin(t * M_PI / (2 * numSamples));
			*o = *d * (1.0f - ratio) + *u * ratio;
#else
			*o = (*d * (numSamples - t) + *u * t) / numSamples;
#endif
			o += numChannels;
			d += numChannels;
			u += numChannels;
		}
	}
}

/* Overlap two sound segments, ramp the volume of one down, while ramping the
other one from zero up, and add them, storing the result at the output. */
static void overlapAddWithSeparation(int numSamples, int numChannels,
	int separation, short* out,
	short* rampDown, short* rampUp) {
	short *o, *u, *d;
	int i, t;

	for (i = 0; i < numChannels; i++) {
		o = out + i;
		u = rampUp + i;
		d = rampDown + i;
		for (t = 0; t < numSamples + separation; t++) {
			if (t < separation) {
				*o = *d * (numSamples - t) / numSamples;
				d += numChannels;
			}
			else if (t < numSamples) {
				*o = (*d * (numSamples - t) + *u * (t - separation)) / numSamples;
				d += numChannels;
				u += numChannels;
			}
			else {
				*o = *u * (t - separation) / numSamples;
				u += numChannels;
			}
			o += numChannels;
		}
	}
}

/* Just move the new samples in the output buffer to the pitch buffer */
static int moveNewSamplesToPitchBuffer(sonicStream stream,
	int originalNumOutputSamples) {
	int numSamples = stream->numOutputSamples - originalNumOutputSamples;
	int numChannels = stream->numChannels;

	if (stream->numPitchSamples + numSamples > stream->pitchBufferSize) {
		stream->pitchBufferSize += (stream->pitchBufferSize >> 1) + numSamples;
		stream->pitchBuffer =
			(short*)realloc(stream->pitchBuffer,
				stream->pitchBufferSize * sizeof(short) * numChannels);
		if (stream->pitchBuffer == NULL) {
			return 0;
		}
	}
	memcpy(stream->pitchBuffer + stream->numPitchSamples * numChannels,
		stream->outputBuffer + originalNumOutputSamples * numChannels,
		numSamples * sizeof(short) * numChannels);
	stream->numOutputSamples = originalNumOutputSamples;
	stream->numPitchSamples += numSamples;
	return 1;
}

/* Remove processed samples from the pitch buffer. */
static void removePitchSamples(sonicStream stream, int numSamples) {
	int numChannels = stream->numChannels;
	short* source = stream->pitchBuffer + numSamples * numChannels;

	if (numSamples == 0) {
		return;
	}
	if (numSamples != stream->numPitchSamples) {
		memmove(
			stream->pitchBuffer, source,
			(stream->numPitchSamples - numSamples) * sizeof(short) * numChannels);
	}
	stream->numPitchSamples -= numSamples;
}

/* Change the pitch.  The latency this introduces could be reduced by looking at
past samples to determine pitch, rather than future. */
static int adjustPitch(sonicStream stream, int originalNumOutputSamples) {
	float pitch = stream->pitch;
	int numChannels = stream->numChannels;
	int period, newPeriod, separation;
	int position = 0;
	short* out;
	short* rampDown;
	short* rampUp;

	if (stream->numOutputSamples == originalNumOutputSamples) {
		return 1;
	}
	if (!moveNewSamplesToPitchBuffer(stream, originalNumOutputSamples)) {
		return 0;
	}
	while (stream->numPitchSamples - position >= stream->maxRequired) {
		period = findPitchPeriod(stream,
			stream->pitchBuffer + position * numChannels, 0);
		newPeriod = period / pitch;
		if (!enlargeOutputBufferIfNeeded(stream, newPeriod)) {
			return 0;
		}
		out = stream->outputBuffer + stream->numOutputSamples * numChannels;
		if (pitch >= 1.0f) {
			rampDown = stream->pitchBuffer + position * numChannels;
			rampUp =
				stream->pitchBuffer + (position + period - newPeriod) * numChannels;
			overlapAdd(newPeriod, numChannels, out, rampDown, rampUp);
		}
		else {
			rampDown = stream->pitchBuffer + position * numChannels;
			rampUp = stream->pitchBuffer + position * numChannels;
			separation = newPeriod - period;
			overlapAddWithSeparation(period, numChannels, separation, out, rampDown,
				rampUp);
		}
		stream->numOutputSamples += newPeriod;
		position += period;
	}
	removePitchSamples(stream, position);
	return 1;
}

/* Aproximate the sinc function times a Hann window from the sinc table. */
static int findSincCoefficient(int i, int ratio, int width) {
	int lobePoints = (SINC_TABLE_SIZE - 1) / SINC_FILTER_POINTS;
	int left = i * lobePoints + (ratio * lobePoints) / width;
	int right = left + 1;
	int position = i * lobePoints * width + ratio * lobePoints - left * width;
	int leftVal = sincTable[left];
	int rightVal = sincTable[right];

	return ((leftVal * (width - position) + rightVal * position) << 1) / width;
}

/* Return 1 if value >= 0, else -1.  This represents the sign of value. */
static int getSign(int value) { return value >= 0 ? 1 : -1; }

/* Interpolate the new output sample. */
static short interpolate(sonicStream stream, short* in, int oldSampleRate,
	int newSampleRate) {
	/* Compute N-point sinc FIR-filter here.  Clip rather than overflow. */
	int i;
	int total = 0;
	int position = stream->newRatePosition * oldSampleRate;
	int leftPosition = stream->oldRatePosition * newSampleRate;
	int rightPosition = (stream->oldRatePosition + 1) * newSampleRate;
	int ratio = rightPosition - position - 1;
	int width = rightPosition - leftPosition;
	int weight, value;
	int oldSign;
	int overflowCount = 0;

	for (i = 0; i < SINC_FILTER_POINTS; i++) {
		weight = findSincCoefficient(i, ratio, width);
		/* printf("%u %f\n", i, weight); */
		value = in[i * stream->numChannels] * weight;
		oldSign = getSign(total);
		total += value;
		if (oldSign != getSign(total) && getSign(value) == oldSign) {
			/* We must have overflowed.  This can happen with a sinc filter. */
			overflowCount += oldSign;
		}
	}
	/* It is better to clip than to wrap if there was a overflow. */
	if (overflowCount > 0) {
		return SHRT_MAX;
	}
	else if (overflowCount < 0) {
		return SHRT_MIN;
	}
	return total >> 16;
}

/* Change the rate.  Interpolate with a sinc FIR filter using a Hann window. */
static int adjustRate(sonicStream stream, float rate,
	int originalNumOutputSamples) {
	int newSampleRate = stream->sampleRate / rate;
	int oldSampleRate = stream->sampleRate;
	int numChannels = stream->numChannels;
	int position = 0;
	short *in, *out;
	int i;
	int N = SINC_FILTER_POINTS;

	/* Set these values to help with the integer math */
	while (newSampleRate > (1 << 14) || oldSampleRate > (1 << 14)) {
		newSampleRate >>= 1;
		oldSampleRate >>= 1;
	}
	if (stream->numOutputSamples == originalNumOutputSamples) {
		return 1;
	}
	if (!moveNewSamplesToPitchBuffer(stream, originalNumOutputSamples)) {
		return 0;
	}
	/* Leave at least N pitch sample in the buffer */
	for (position = 0; position < stream->numPitchSamples - N; position++) {
		while ((stream->oldRatePosition + 1) * newSampleRate >
			stream->newRatePosition * oldSampleRate) {
			if (!enlargeOutputBufferIfNeeded(stream, 1)) {
				return 0;
			}
			out = stream->outputBuffer + stream->numOutputSamples * numChannels;
			in = stream->pitchBuffer + position * numChannels;
			for (i = 0; i < numChannels; i++) {
				*out++ = interpolate(stream, in, oldSampleRate, newSampleRate);
				in++;
			}
			stream->newRatePosition++;
			stream->numOutputSamples++;
		}
		stream->oldRatePosition++;
		if (stream->oldRatePosition == oldSampleRate) {
			stream->oldRatePosition = 0;
			if (stream->newRatePosition != newSampleRate) {
				fprintf(stderr,
					"Assertion failed: stream->newRatePosition != newSampleRate\n");
				exit(1);
			}
			stream->newRatePosition = 0;
		}
	}
	removePitchSamples(stream, position);
	return 1;
}

/* Skip over a pitch period, and copy period/speed samples to the output */
static int skipPitchPeriod(sonicStream stream, short* samples, float speed,
	int period) {
	long newSamples;
	int numChannels = stream->numChannels;

	if (speed >= 2.0f) {
		newSamples = period / (speed - 1.0f);
	}
	else {
		newSamples = period;
		stream->remainingInputToCopy = period * (2.0f - speed) / (speed - 1.0f);
	}
	if (!enlargeOutputBufferIfNeeded(stream, newSamples)) {
		return 0;
	}
	overlapAdd(newSamples, numChannels,
		stream->outputBuffer + stream->numOutputSamples * numChannels,
		samples, samples + period * numChannels);
	stream->numOutputSamples += newSamples;
	return newSamples;
}

/* Insert a pitch period, and determine how much input to copy directly. */
static int insertPitchPeriod(sonicStream stream, short* samples, float speed,
	int period) {
	long newSamples;
	short* out;
	int numChannels = stream->numChannels;

	if (speed < 0.5f) {
		newSamples = period * speed / (1.0f - speed);
	}
	else {
		newSamples = period;
		stream->remainingInputToCopy =
			period * (2.0f * speed - 1.0f) / (1.0f - speed);
	}
	if (!enlargeOutputBufferIfNeeded(stream, period + newSamples)) {
		return 0;
	}
	out = stream->outputBuffer + stream->numOutputSamples * numChannels;
	memcpy(out, samples, period * sizeof(short) * numChannels);
	out =
		stream->outputBuffer + (stream->numOutputSamples + period) * numChannels;
	overlapAdd(newSamples, numChannels, out, samples + period * numChannels,
		samples);
	stream->numOutputSamples += period + newSamples;
	return newSamples;
}

/* Resample as many pitch periods as we have buffered on the input.  Return 0 if
we fail to resize an input or output buffer. */
static int changeSpeed(sonicStream stream, float speed) {
	short* samples;
	int numSamples = stream->numInputSamples;
	int position = 0, period, newSamples;
	int maxRequired = stream->maxRequired;

	/* printf("Changing speed to %f\n", speed); */
	if (stream->numInputSamples < maxRequired) {
		return 1;
	}
	do {
		if (stream->remainingInputToCopy > 0) {
			newSamples = copyInputToOutput(stream, position);
			position += newSamples;
		}
		else {
			samples = stream->inputBuffer + position * stream->numChannels;
			period = findPitchPeriod(stream, samples, 1);
#ifdef SONIC_SPECTROGRAM
			if (stream->spectrogram != NULL) {
				sonicAddPitchPeriodToSpectrogram(stream->spectrogram, samples, period,
					stream->numChannels);
				newSamples = period;
				position += period;
			}
			else
#endif  /* SONIC_SPECTROGRAM */
				if (speed > 1.0) {
					newSamples = skipPitchPeriod(stream, samples, speed, period);
					position += period + newSamples;
				}
				else {
					newSamples = insertPitchPeriod(stream, samples, speed, period);
					position += newSamples;
				}
		}
		if (newSamples == 0) {
			return 0; /* Failed to resize output buffer */
		}
	} while (position + maxRequired <= numSamples);
	removeInputSamples(stream, position);
	return 1;
}

/* Resample as many pitch periods as we have buffered on the input.  Return 0 if
we fail to resize an input or output buffer.  Also scale the output by the
volume. */
static int processStreamInput(sonicStream stream) {
	int originalNumOutputSamples = stream->numOutputSamples;
	float speed = stream->speed / stream->pitch;
	float rate = stream->rate;

	if (!stream->useChordPitch) {
		rate *= stream->pitch;
	}
	if (speed > 1.00001 || speed < 0.99999) {
		changeSpeed(stream, speed);
	}
	else {
		if (!copyToOutput(stream, stream->inputBuffer, stream->numInputSamples)) {
			return 0;
		}
		stream->numInputSamples = 0;
	}
	if (stream->useChordPitch) {
		if (stream->pitch != 1.0f) {
			if (!adjustPitch(stream, originalNumOutputSamples)) {
				return 0;
			}
		}
	}
	else if (rate != 1.0f) {
		if (!adjustRate(stream, rate, originalNumOutputSamples)) {
			return 0;
		}
	}
	if (stream->volume != 1.0f) {
		/* Adjust output volume. */
		scaleSamples(
			stream->outputBuffer + originalNumOutputSamples * stream->numChannels,
			(stream->numOutputSamples - originalNumOutputSamples) *
			stream->numChannels,
			stream->volume);
	}
	return 1;
}

/* Write floating point data to the input buffer and process it. */
int sonicWriteFloatToStream(sonicStream stream, float* samples,
	int numSamples) {
	if (!addFloatSamplesToInputBuffer(stream, samples, numSamples)) {
		return 0;
	}
	return processStreamInput(stream);
}

/* Simple wrapper around sonicWriteFloatToStream that does the short to float
conversion for you. */
int sonicWriteShortToStream(sonicStream stream, short* samples,
	int numSamples) {
	if (!addShortSamplesToInputBuffer(stream, samples, numSamples)) {
		return 0;
	}
	return processStreamInput(stream);
}

/* Simple wrapper around sonicWriteFloatToStream that does the unsigned char to
float conversion for you. */
int sonicWriteUnsignedCharToStream(sonicStream stream, unsigned char* samples,
	int numSamples) {
	if (!addUnsignedCharSamplesToInputBuffer(stream, samples, numSamples)) {
		return 0;
	}
	return processStreamInput(stream);
}

/* This is a non-stream oriented interface to just change the speed of a sound
* sample */
int sonicChangeFloatSpeed(float* samples, int numSamples, float speed,
	float pitch, float rate, float volume,
	int useChordPitch, int sampleRate, int numChannels) {
	sonicStream stream = sonicCreateStream(sampleRate, numChannels);

	sonicSetSpeed(stream, speed);
	sonicSetPitch(stream, pitch);
	sonicSetRate(stream, rate);
	sonicSetVolume(stream, volume);
	sonicSetChordPitch(stream, useChordPitch);
	sonicWriteFloatToStream(stream, samples, numSamples);
	sonicFlushStream(stream);
	numSamples = sonicSamplesAvailable(stream);
	sonicReadFloatFromStream(stream, samples, numSamples);
	sonicDestroyStream(stream);
	return numSamples;
}

/* This is a non-stream oriented interface to just change the speed of a sound
* sample */
int sonicChangeShortSpeed(short* samples, int numSamples, float speed,
	float pitch, float rate, float volume,
	int useChordPitch, int sampleRate, int numChannels) {
	sonicStream stream = sonicCreateStream(sampleRate, numChannels);

	sonicSetSpeed(stream, speed);
	sonicSetPitch(stream, pitch);
	sonicSetRate(stream, rate);
	sonicSetVolume(stream, volume);
	sonicSetChordPitch(stream, useChordPitch);
	sonicWriteShortToStream(stream, samples, numSamples);
	sonicFlushStream(stream);
	numSamples = sonicSamplesAvailable(stream);
	sonicReadShortFromStream(stream, samples, numSamples);
	sonicDestroyStream(stream);
	return numSamples;
}

左侧解决方案资源管理器视图中，源文件栏右键添加新建项，

点击C++文件，新建一个源文件main.cpp，点击“添加”

输入代码如下：

#include <time.h>

#include <cstring>
#include <iostream>
#include <memory>
#include <queue>
#include <set>
#include <vector>

#include "sonic.h"

//#define DROP_FRAME

const int MAXN = 1e6 + 7;

const int FRAMESIZE = 8000 / (1000 / 20) * 2;

typedef std::pair<int, int> pii;

std::set<pii> jitterLine;

typedef long long ll;

bool isInLine(int pos) {
	for (auto x : jitterLine) {
		if (x.first <= pos && x.second >= pos) {
			return true;
		}
	}
	return false;
}

void dumpLine() {
	for (auto x : jitterLine) {
		printf("begin:%d, len:%d beginTime:%lf\n", x.first, x.second - x.first,
			x.first * 20.0 / 1000);
	}
}

ll calLvl(uint16_t *buffer, int len) {
	ll sum = 0;
	for (int i = 0; i < len; ++i) {
		sum += (buffer[i] * buffer[i]);
	}
	return sum;
}

void spedUp(uint16_t *outBuf, uint16_t *inBuf1, uint16_t *inBuf2) {
	int p = 0;
	for (int i = 0; i < FRAMESIZE; ++i) {
		if (i & 1) {
			outBuf[p++] = inBuf1[i];
		}
	}
	for (int i = 0; i < FRAMESIZE; ++i) {
		if (i & 1) {
			outBuf[p++] = inBuf2[i];
		}
	}
}

const double eps = 1e-6;

int main(int argc, char *argv[]) {
	// sonicStream tempoStream_;
	// tempoStream_ = sonicCreateStream(8000, 1);

	FILE *file1 = fopen("xinwenlianbo.pcm", "rb");
	FILE *out_file = fopen("outSped.pcm", "wb");

	int processFrameCount = 3000;
	int maxJitterSize = 20;
	int jitterTimes = 20;
	double speedLow = 0.5;
	double speedHigh = 2.0;

	uint8_t *inBuf = new uint8_t[MAXN];

	uint8_t *it = inBuf;

	uint8_t addBuf[MAXN] = { 0 };

	srand(time(0));

	/* generate jitter */
	for (int i = 0; i < jitterTimes; ++i) {
		int begin = rand() % processFrameCount;
		int len = rand() % maxJitterSize;
		jitterLine.insert(std::make_pair(begin, begin + len));
	}

	/* dump jitter */
#ifdef DROP_FRAME
	printf("this frame will be frop!\n");
#else
	printf("this frame will be set mute!\n");
#endif
	dumpLine();

	/* if p is in jitter range then repalce it into mute frame */
	for (int p = 0; p < processFrameCount; ++p) {
		if (isInLine(p)) {
			it = inBuf;
			for (int i = 0; i < maxJitterSize; ++i) {
				fread(it, 1, FRAMESIZE, file1);
				it += FRAMESIZE;
			}
			sonicChangeShortSpeed((short int *)inBuf, maxJitterSize * FRAMESIZE / 2,
				speedLow, 1.0f, 1.0f, 1.0f, 0, 8000, 1);
			fwrite(inBuf, 1, (1.0 * maxJitterSize / speedLow + eps) * FRAMESIZE, out_file);

			it = inBuf;
			for (int i = 0; i < maxJitterSize; ++i) {
				fread(it, 1, FRAMESIZE, file1);
				it += FRAMESIZE;
			}
			sonicChangeShortSpeed((short int *)inBuf, maxJitterSize * FRAMESIZE / 2,
				speedHigh, 1.0f, 1.0f, 1.0f, 0, 8000, 1);
			fwrite(inBuf, 1, (1.0 * maxJitterSize / speedHigh + eps) * FRAMESIZE,
				out_file);
			p += (maxJitterSize * 2);
		}
		else {
			fread(inBuf, 1, FRAMESIZE, file1);
			fwrite(inBuf, 1, FRAMESIZE, out_file);
		}
	}

	// sonicChangeShortSpeed((short int*)inBuf, processFrameCount * FRAMESIZE / 2,
	// 0.5f, 1.0f, 1.0f, 1.0f, 0, 8000, 1); fwrite(inBuf, 1, processFrameCount *
	// FRAMESIZE, out_file);

	return 0;
}

代码添加完毕，准备好一个语音文件：xinwenlianbo.pcm，放在源文件目录下

二. 生成编译

点击目录栏“生成”中“生成解决方案”

报 error C4996: 'fopen’错的话，可以参考文章：
VS BUG（1） error C4996: ‘fopen‘: This function or variable may be unsafe.
问题解决，重新生成

生成成功，调试运行，生成一个outSped.pcm文件，调用成功。

三. 小结

最近项目上需要对WAV语音进行处理，这个是对语音进行加速减速播放时需要保证语音不失真，目前工程已经可以跑通，但目前是应用pcm格式的语音文件，原语音是2分钟的样子，生成的语音是一分钟的样子，转成WAV格式试听了一下，感觉不像是加速了，后期还需要进行相应的调整，并直接应用WAV文件格式的语音进行变速不变调，完成调试后再进行具体的介绍。