165 lines
8.0 KiB
C
165 lines
8.0 KiB
C
/* Sonic library
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Copyright 2010
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Bill Cox
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This file is part of the Sonic Library.
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This file is licensed under the Apache 2.0 license, and also placed into the public domain.
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Use it either way, at your option.
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*/
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/*
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The Sonic Library implements a new algorithm invented by Bill Cox for the
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specific purpose of speeding up speech by high factors at high quality. It
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generates smooth speech at speed up factors as high as 6X, possibly more. It is
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also capable of slowing down speech, and generates high quality results
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regardless of the speed up or slow down factor. For speeding up speech by 2X or
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more, the following equation is used:
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newSamples = period/(speed - 1.0)
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scale = 1.0/newSamples;
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where period is the current pitch period, determined using AMDF or any other
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pitch estimator, and speed is the speedup factor. If the current position in
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the input stream is pointed to by "samples", and the current output stream
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position is pointed to by "out", then newSamples number of samples can be
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generated with:
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out[t] = (samples[t]*(newSamples - t) + samples[t + period]*t)/newSamples;
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where t = 0 to newSamples - 1.
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For speed factors < 2X, the PICOLA algorithm is used. The above
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algorithm is first used to double the speed of one pitch period. Then, enough
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input is directly copied from the input to the output to achieve the desired
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speed up factor, where 1.0 < speed < 2.0. The amount of data copied is derived:
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speed = (2*period + length)/(period + length)
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speed*length + speed*period = 2*period + length
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length(speed - 1) = 2*period - speed*period
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length = period*(2 - speed)/(speed - 1)
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For slowing down speech where 0.5 < speed < 1.0, a pitch period is inserted into
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the output twice, and length of input is copied from the input to the output
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until the output desired speed is reached. The length of data copied is:
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length = period*(speed - 0.5)/(1 - speed)
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For slow down factors below 0.5, no data is copied, and an algorithm
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similar to high speed factors is used.
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*/
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#ifdef __cplusplus
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extern "C" {
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#endif
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/* Uncomment this to use sin-wav based overlap add which in theory can improve
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sound quality slightly, at the expense of lots of floating point math. */
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/* #define SONIC_USE_SIN */
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/* This specifies the range of voice pitches we try to match.
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Note that if we go lower than 65, we could overflow in findPitchInRange */
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#define SONIC_MIN_PITCH 65
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#define SONIC_MAX_PITCH 400
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/* These are used to down-sample some inputs to improve speed */
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#define SONIC_AMDF_FREQ 4000
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struct sonicStreamStruct;
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typedef struct sonicStreamStruct *sonicStream;
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/* For all of the following functions, numChannels is multiplied by numSamples
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to determine the actual number of values read or returned. */
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/* Create a sonic stream. Return NULL only if we are out of memory and cannot
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allocate the stream. Set numChannels to 1 for mono, and 2 for stereo. */
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// 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
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sonicStream sonicCreateStream(int sampleRate, int numChannels);
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/* Destroy the sonic stream. */
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// 销毁一个音频流
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void sonicDestroyStream(sonicStream stream);
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/* Use this to write floating point data to be speed up or down into the stream.
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Values must be between -1 and 1. Return 0 if memory realloc failed, otherwise 1 */
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//
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int sonicWriteFloatToStream(sonicStream stream, float *samples, int numSamples);
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/* Use this to write 16-bit data to be speed up or down into the stream.
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Return 0 if memory realloc failed, otherwise 1 */
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int sonicWriteShortToStream(sonicStream stream, short *samples, int numSamples);
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/* Use this to write 8-bit unsigned data to be speed up or down into the stream.
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Return 0 if memory realloc failed, otherwise 1 */
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int sonicWriteUnsignedCharToStream(sonicStream stream, unsigned char *samples, int numSamples);
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/* Use this to read floating point data out of the stream. Sometimes no data
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will be available, and zero is returned, which is not an error condition. */
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int sonicReadFloatFromStream(sonicStream stream, float *samples, int maxSamples);
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/* Use this to read 16-bit data out of the stream. Sometimes no data will
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be available, and zero is returned, which is not an error condition. */
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int sonicReadShortFromStream(sonicStream stream, short *samples, int maxSamples);
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/* Use this to read 8-bit unsigned data out of the stream. Sometimes no data will
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be available, and zero is returned, which is not an error condition. */
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int sonicReadUnsignedCharFromStream(sonicStream stream, unsigned char *samples, int maxSamples);
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/* Force the sonic stream to generate output using whatever data it currently
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has. No extra delay will be added to the output, but flushing in the middle of
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words could introduce distortion. */
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// 立即强制刷新流
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int sonicFlushStream(sonicStream stream);
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/* Return the number of samples in the output buffer */
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// 返回输出缓冲中的采样点数目
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int sonicSamplesAvailable(sonicStream stream);
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/* Get the speed of the stream. */
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// 得到音频流的速度
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float sonicGetSpeed(sonicStream stream);
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/* Set the speed of the stream. */
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// 设置音频流的速度
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void sonicSetSpeed(sonicStream stream, float speed);
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/* Get the pitch of the stream. */
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float sonicGetPitch(sonicStream stream);
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/* Set the pitch of the stream. */
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void sonicSetPitch(sonicStream stream, float pitch);
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/* Get the rate of the stream. */
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float sonicGetRate(sonicStream stream);
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/* Set the rate of the stream. */
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void sonicSetRate(sonicStream stream, float rate);
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/* Get the scaling factor of the stream. */
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float sonicGetVolume(sonicStream stream);
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/* Set the scaling factor of the stream. */
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void sonicSetVolume(sonicStream stream, float volume);
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/* Get the chord pitch setting. */
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int sonicGetChordPitch(sonicStream stream);
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/* Set chord pitch mode on or off. Default is off. See the documentation
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page for a description of this feature. */
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void sonicSetChordPitch(sonicStream stream, int useChordPitch);
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/* Get the quality setting. */
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// 得到音频流的质量
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int sonicGetQuality(sonicStream stream);
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/* Set the "quality". Default 0 is virtually as good as 1, but very much faster. */
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// 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
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void sonicSetQuality(sonicStream stream, int quality);
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/* Get the sample rate of the stream. */
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// 得到音频流的采样率
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int sonicGetSampleRate(sonicStream stream);
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/* Set the sample rate of the stream. This will drop any samples that have not been read. */
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// 设置音频流的采样率
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void sonicSetSampleRate(sonicStream stream, int sampleRate);
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/* Get the number of channels. */
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// 得到音频的声道数
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int sonicGetNumChannels(sonicStream stream);
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/* Set the number of channels. This will drop any samples that have not been read. */
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// 设置音频流的声道数
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void sonicSetNumChannels(sonicStream stream, int numChannels);
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/* This is a non-stream oriented interface to just change the speed of a sound
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sample. It works in-place on the sample array, so there must be at least
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speed*numSamples available space in the array. Returns the new number of samples. */
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// 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，
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//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
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int sonicChangeFloatSpeed(float *samples, int numSamples, float speed, float pitch,
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float rate, float volume, int useChordPitch, int sampleRate, int numChannels);
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/* This is a non-stream oriented interface to just change the speed of a sound
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sample. It works in-place on the sample array, so there must be at least
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speed*numSamples available space in the array. Returns the new number of samples. */
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int sonicChangeShortSpeed(short *samples, int numSamples, float speed, float pitch,
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float rate, float volume, int useChordPitch, int sampleRate, int numChannels);
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#ifdef __cplusplus
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}
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#endif
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