/* Sonic library
   Copyright 2010
   Bill Cox
   This file is part of the Sonic Library.
   This file is licensed under the Apache 2.0 license, and also placed into the public domain.
   Use it either way, at your option.
*/

/*
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.
*/

#ifdef  __cplusplus
extern "C" {
#endif

/* 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 */

/* 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. */
// 创建一个音频流，如果内存溢出不能创建流会返回NULL，numCHannels表示声道的个数，1为单声道，2为双声道
sonicStream sonicCreateStream(int sampleRate, int numChannels);
/* Destroy the sonic stream. */
// 销毁一个音频流
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. */
// 立即强制刷新流
int sonicFlushStream(sonicStream stream);
/* Return the number of samples in the output buffer */
// 返回输出缓冲中的采样点数目
int sonicSamplesAvailable(sonicStream stream);
/* Get the speed of the stream. */
// 得到音频流的速度
float sonicGetSpeed(sonicStream stream);
/* Set the speed of the stream. */
// 设置音频流的速度
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. */
// 得到音频流的质量
int sonicGetQuality(sonicStream stream);
/* Set the "quality".  Default 0 is virtually as good as 1, but very much faster. */
// 设置音频流的质量，默认的0的质量几乎和1的一样好，但是更快
void sonicSetQuality(sonicStream stream, int quality);
/* Get the sample rate of the stream. */
// 得到音频流的采样率
int sonicGetSampleRate(sonicStream stream);
/* Set the sample rate of the stream.  This will drop any samples that have not been read. */
// 设置音频流的采样率
void sonicSetSampleRate(sonicStream stream, int sampleRate);
/* Get the number of channels. */
// 得到音频的声道数
int sonicGetNumChannels(sonicStream stream);
/* Set the number of channels.  This will drop any samples that have not been read. */
// 设置音频流的声道数
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. */
// 这是一个非面向流的借口，只是改变声音采样的速率。它工作在采样数组内部，
//所以在数组内至少要有speed*numSampes大小的空间。返回值是新的采样点的数目

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  __cplusplus
}
#endif
