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GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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Files in this directory are used by GNU autoconf/automake system.
These files aren't used/needed in the Windows environment.
Files in this directory are used by GNU autoconf/automake system.
These files aren't used/needed in the Windows environment.

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lib-src/soundtouch/soundtouch/BPMDetect.h
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////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: BPMDetect.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _BPMDetect_H_
#define _BPMDetect_H_
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 45
/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM 230
/// Class for calculating BPM rate for audio data.
class BPMDetect
{
protected:
/// Auto-correlation accumulator bins.
float *xcorr;
/// Amplitude envelope sliding average approximation level accumulator
float envelopeAccu;
/// RMS volume sliding average approximation level accumulator
float RMSVolumeAccu;
/// Sample average counter.
int decimateCount;
/// Sample average accumulator for FIFO-like decimation.
soundtouch::LONG_SAMPLETYPE decimateSum;
/// Decimate sound by this coefficient to reach approx. 500 Hz.
int decimateBy;
/// Auto-correlation window length
int windowLen;
/// Number of channels (1 = mono, 2 = stereo)
int channels;
/// sample rate
int sampleRate;
/// Beginning of auto-correlation window: Autocorrelation isn't being updated for
/// the first these many correlation bins.
int windowStart;
/// FIFO-buffer for decimated processing samples.
soundtouch::FIFOSampleBuffer *buffer;
/// Initialize the class for processing.
void init(int numChannels, int sampleRate);
/// Updates auto-correlation function for given number of decimated samples that
/// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
/// though).
void updateXCorr(int process_samples /// How many samples are processed.
);
/// Decimates samples to approx. 500 Hz.
///
/// \return Number of output samples.
int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
int numsamples ///< Number of source samples.
);
/// Calculates amplitude envelope for the buffer of samples.
/// Result is output to 'samples'.
void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
int numsamples ///< Number of samples in buffer
);
public:
/// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data.
int sampleRate ///< Sample rate in Hz.
);
/// Destructor.
virtual ~BPMDetect();
/// Inputs a block of samples for analyzing: Envelopes the samples and then
/// updates the autocorrelation estimation. When whole song data has been input
/// in smaller blocks using this function, read the resulting bpm with 'getBpm'
/// function.
///
/// Notice that data in 'samples' array can be disrupted in processing.
void inputSamples(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
int numSamples ///< Number of samples in buffer
);
/// Analyzes the results and returns the BPM rate. Use this function to read result
/// after whole song data has been input to the class by consecutive calls of
/// 'inputSamples' function.
///
/// \return Beats-per-minute rate, or zero if detection failed.
float getBpm();
};
#endif // _BPMDetect_H_
////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: BPMDetect.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _BPMDetect_H_
#define _BPMDetect_H_
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 45
/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM 230
/// Class for calculating BPM rate for audio data.
class BPMDetect
{
protected:
/// Auto-correlation accumulator bins.
float *xcorr;
/// Amplitude envelope sliding average approximation level accumulator
float envelopeAccu;
/// RMS volume sliding average approximation level accumulator
float RMSVolumeAccu;
/// Sample average counter.
int decimateCount;
/// Sample average accumulator for FIFO-like decimation.
soundtouch::LONG_SAMPLETYPE decimateSum;
/// Decimate sound by this coefficient to reach approx. 500 Hz.
int decimateBy;
/// Auto-correlation window length
int windowLen;
/// Number of channels (1 = mono, 2 = stereo)
int channels;
/// sample rate
int sampleRate;
/// Beginning of auto-correlation window: Autocorrelation isn't being updated for
/// the first these many correlation bins.
int windowStart;
/// FIFO-buffer for decimated processing samples.
soundtouch::FIFOSampleBuffer *buffer;
/// Initialize the class for processing.
void init(int numChannels, int sampleRate);
/// Updates auto-correlation function for given number of decimated samples that
/// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
/// though).
void updateXCorr(int process_samples /// How many samples are processed.
);
/// Decimates samples to approx. 500 Hz.
///
/// \return Number of output samples.
int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
int numsamples ///< Number of source samples.
);
/// Calculates amplitude envelope for the buffer of samples.
/// Result is output to 'samples'.
void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
int numsamples ///< Number of samples in buffer
);
public:
/// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data.
int sampleRate ///< Sample rate in Hz.
);
/// Destructor.
virtual ~BPMDetect();
/// Inputs a block of samples for analyzing: Envelopes the samples and then
/// updates the autocorrelation estimation. When whole song data has been input
/// in smaller blocks using this function, read the resulting bpm with 'getBpm'
/// function.
///
/// Notice that data in 'samples' array can be disrupted in processing.
void inputSamples(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
int numSamples ///< Number of samples in buffer
);
/// Analyzes the results and returns the BPM rate. Use this function to read result
/// after whole song data has been input to the class by consecutive calls of
/// 'inputSamples' function.
///
/// \return Beats-per-minute rate, or zero if detection failed.
float getBpm();
};
#endif // _BPMDetect_H_

348
lib-src/soundtouch/soundtouch/FIFOSampleBuffer.h
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@@ -1,174 +1,174 @@
////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// output samples from the buffer as well as grows the storage size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: FIFOSampleBuffer.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSampleBuffer_H
#define FIFOSampleBuffer_H
#include "FIFOSamplePipe.h"
namespace soundtouch
{
/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
/// care of storage size adjustment and data moving during input/output operations.
///
/// Notice that in case of stereo audio, one sample is considered to consist of
/// both channel data.
class FIFOSampleBuffer : public FIFOSamplePipe
{
private:
/// Sample buffer.
SAMPLETYPE *buffer;
// Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
// 16-byte aligned location of this buffer
SAMPLETYPE *bufferUnaligned;
/// Sample buffer size in bytes
uint sizeInBytes;
/// How many samples are currently in buffer.
uint samplesInBuffer;
/// Channels, 1=mono, 2=stereo.
uint channels;
/// Current position pointer to the buffer. This pointer is increased when samples are
/// removed from the pipe so that it's necessary to actually rewind buffer (move data)
/// only new data when is put to the pipe.
uint bufferPos;
/// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
/// beginning of the buffer.
void rewind();
/// Ensures that the buffer has capacity for at least this many samples.
void ensureCapacity(const uint capacityRequirement);
/// Returns current capacity.
uint getCapacity() const;
public:
/// Constructor
FIFOSampleBuffer(uint numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
///< Default is stereo.
);
/// destructor
~FIFOSampleBuffer();
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const;
/// Returns a pointer to the end of the used part of the sample buffer (i.e.
/// where the new samples are to be inserted). This function may be used for
/// inserting new samples into the sample buffer directly. Please be careful
/// not corrupt the book-keeping!
///
/// When using this function as means for inserting new samples, also remember
/// to increase the sample count afterwards, by calling the
/// 'putSamples(numSamples)' function.
SAMPLETYPE *ptrEnd(
uint slackCapacity ///< How much free capacity (in samples) there _at least_
///< should be so that the caller can succesfully insert the
///< desired samples to the buffer. If necessary, the function
///< grows the buffer size to comply with this requirement.
);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
);
/// Adjusts the book-keeping to increase number of samples in the buffer without
/// copying any actual samples.
///
/// This function is used to update the number of samples in the sample buffer
/// when accessing the buffer directly with 'ptrEnd' function. Please be
/// careful though!
virtual void putSamples(uint numSamples ///< Number of samples been inserted.
);
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
);
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
);
/// Returns number of samples currently available.
virtual uint numSamples() const;
/// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(uint numChannels);
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const;
/// Clears all the samples.
virtual void clear();
};
}
#endif
////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// output samples from the buffer as well as grows the storage size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: FIFOSampleBuffer.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSampleBuffer_H
#define FIFOSampleBuffer_H
#include "FIFOSamplePipe.h"
namespace soundtouch
{
/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
/// care of storage size adjustment and data moving during input/output operations.
///
/// Notice that in case of stereo audio, one sample is considered to consist of
/// both channel data.
class FIFOSampleBuffer : public FIFOSamplePipe
{
private:
/// Sample buffer.
SAMPLETYPE *buffer;
// Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
// 16-byte aligned location of this buffer
SAMPLETYPE *bufferUnaligned;
/// Sample buffer size in bytes
uint sizeInBytes;
/// How many samples are currently in buffer.
uint samplesInBuffer;
/// Channels, 1=mono, 2=stereo.
uint channels;
/// Current position pointer to the buffer. This pointer is increased when samples are
/// removed from the pipe so that it's necessary to actually rewind buffer (move data)
/// only new data when is put to the pipe.
uint bufferPos;
/// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
/// beginning of the buffer.
void rewind();
/// Ensures that the buffer has capacity for at least this many samples.
void ensureCapacity(const uint capacityRequirement);
/// Returns current capacity.
uint getCapacity() const;
public:
/// Constructor
FIFOSampleBuffer(uint numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
///< Default is stereo.
);
/// destructor
~FIFOSampleBuffer();
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const;
/// Returns a pointer to the end of the used part of the sample buffer (i.e.
/// where the new samples are to be inserted). This function may be used for
/// inserting new samples into the sample buffer directly. Please be careful
/// not corrupt the book-keeping!
///
/// When using this function as means for inserting new samples, also remember
/// to increase the sample count afterwards, by calling the
/// 'putSamples(numSamples)' function.
SAMPLETYPE *ptrEnd(
uint slackCapacity ///< How much free capacity (in samples) there _at least_
///< should be so that the caller can succesfully insert the
///< desired samples to the buffer. If necessary, the function
///< grows the buffer size to comply with this requirement.
);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
);
/// Adjusts the book-keeping to increase number of samples in the buffer without
/// copying any actual samples.
///
/// This function is used to update the number of samples in the sample buffer
/// when accessing the buffer directly with 'ptrEnd' function. Please be
/// careful though!
virtual void putSamples(uint numSamples ///< Number of samples been inserted.
);
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
);
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
);
/// Returns number of samples currently available.
virtual uint numSamples() const;
/// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(uint numChannels);
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const;
/// Clears all the samples.
virtual void clear();
};
}
#endif

434
lib-src/soundtouch/soundtouch/FIFOSamplePipe.h
View File

@@ -1,217 +1,217 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
/// samples by operating like a first-in-first-out pipe: New samples are fed
/// into one end of the pipe with the 'putSamples' function, and the processed
/// samples are received from the other end with the 'receiveSamples' function.
///
/// 'FIFOProcessor' : A base class for classes the do signal processing with
/// the samples while operating like a first-in-first-out pipe. When samples
/// are input with the 'putSamples' function, the class processes them
/// and moves the processed samples to the given 'output' pipe object, which
/// may be either another processing stage, or a fifo sample buffer object.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: FIFOSamplePipe.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSamplePipe_H
#define FIFOSamplePipe_H
#include <assert.h>
#include <stdlib.h>
#include "STTypes.h"
namespace soundtouch
{
/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
class FIFOSamplePipe
{
public:
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const = 0;
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
) = 0;
// Moves samples from the 'other' pipe instance to this instance.
void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
)
{
int oNumSamples = other.numSamples();
putSamples(other.ptrBegin(), oNumSamples);
other.receiveSamples(oNumSamples);
};
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
) = 0;
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
) = 0;
/// Returns number of samples currently available.
virtual uint numSamples() const = 0;
// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const = 0;
/// Clears all the samples.
virtual void clear() = 0;
};
/// Base-class for sound processing routines working in FIFO principle. With this base
/// class it's easy to implement sound processing stages that can be chained together,
/// so that samples that are fed into beginning of the pipe automatically go through
/// all the processing stages.
///
/// When samples are input to this class, they're first processed and then put to
/// the FIFO pipe that's defined as output of this class. This output pipe can be
/// either other processing stage or a FIFO sample buffer.
class FIFOProcessor :public FIFOSamplePipe
{
protected:
/// Internal pipe where processed samples are put.
FIFOSamplePipe *output;
/// Sets output pipe.
void setOutPipe(FIFOSamplePipe *pOutput)
{
assert(output == NULL);
assert(pOutput != NULL);
output = pOutput;
}
/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
{
output = NULL;
}
/// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
)
{
output = pOutput;
}
/// Destructor.
virtual ~FIFOProcessor()
{
}
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const
{
return output->ptrBegin();
}
public:
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
)
{
return output->receiveSamples(outBuffer, maxSamples);
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
)
{
return output->receiveSamples(maxSamples);
}
/// Returns number of samples currently available.
virtual uint numSamples() const
{
return output->numSamples();
}
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const
{
return output->isEmpty();
}
};
}
#endif
////////////////////////////////////////////////////////////////////////////////
///
/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
/// samples by operating like a first-in-first-out pipe: New samples are fed
/// into one end of the pipe with the 'putSamples' function, and the processed
/// samples are received from the other end with the 'receiveSamples' function.
///
/// 'FIFOProcessor' : A base class for classes the do signal processing with
/// the samples while operating like a first-in-first-out pipe. When samples
/// are input with the 'putSamples' function, the class processes them
/// and moves the processed samples to the given 'output' pipe object, which
/// may be either another processing stage, or a fifo sample buffer object.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: FIFOSamplePipe.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSamplePipe_H
#define FIFOSamplePipe_H
#include <assert.h>
#include <stdlib.h>
#include "STTypes.h"
namespace soundtouch
{
/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
class FIFOSamplePipe
{
public:
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const = 0;
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
) = 0;
// Moves samples from the 'other' pipe instance to this instance.
void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
)
{
int oNumSamples = other.numSamples();
putSamples(other.ptrBegin(), oNumSamples);
other.receiveSamples(oNumSamples);
};
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
) = 0;
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
) = 0;
/// Returns number of samples currently available.
virtual uint numSamples() const = 0;
// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const = 0;
/// Clears all the samples.
virtual void clear() = 0;
};
/// Base-class for sound processing routines working in FIFO principle. With this base
/// class it's easy to implement sound processing stages that can be chained together,
/// so that samples that are fed into beginning of the pipe automatically go through
/// all the processing stages.
///
/// When samples are input to this class, they're first processed and then put to
/// the FIFO pipe that's defined as output of this class. This output pipe can be
/// either other processing stage or a FIFO sample buffer.
class FIFOProcessor :public FIFOSamplePipe
{
protected:
/// Internal pipe where processed samples are put.
FIFOSamplePipe *output;
/// Sets output pipe.
void setOutPipe(FIFOSamplePipe *pOutput)
{
assert(output == NULL);
assert(pOutput != NULL);
output = pOutput;
}
/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
{
output = NULL;
}
/// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
)
{
output = pOutput;
}
/// Destructor.
virtual ~FIFOProcessor()
{
}
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const
{
return output->ptrBegin();
}
public:
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
)
{
return output->receiveSamples(outBuffer, maxSamples);
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
)
{
return output->receiveSamples(maxSamples);
}
/// Returns number of samples currently available.
virtual uint numSamples() const
{
return output->numSamples();
}
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const
{
return output->isEmpty();
}
};
}
#endif

286
lib-src/soundtouch/soundtouch/STTypes.h
View File

@@ -1,143 +1,143 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Common type definitions for SoundTouch audio processing library.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: STTypes.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef STTypes_H
#define STTypes_H
typedef unsigned int uint;
typedef unsigned long ulong;
#ifdef __GNUC__
// In GCC, include soundtouch_config.h made by config scritps
#include "soundtouch_config.h"
#endif
#ifndef _WINDEF_
// if these aren't defined already by Windows headers, define now
typedef int BOOL;
#define FALSE 0
#define TRUE 1
#endif // _WINDEF_
namespace soundtouch
{
/// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file:
//#undef INTEGER_SAMPLES
//#undef FLOAT_SAMPLES
#if !(INTEGER_SAMPLES || FLOAT_SAMPLES)
/// Choose either 32bit floating point or 16bit integer sampletype
/// by choosing one of the following defines, unless this selection
/// has already been done in some other file.
////
/// Notes:
/// - In Windows environment, choose the sample format with the
/// following defines.
/// - In GNU environment, the floating point samples are used by
/// default, but integer samples can be chosen by giving the
/// following switch to the configure script:
/// ./configure --enable-integer-samples
/// However, if you still prefer to select the sample format here
/// also in GNU environment, then please #undef the INTEGER_SAMPLE
/// and FLOAT_SAMPLE defines first as in comments above.
//#define INTEGER_SAMPLES 1 //< 16bit integer samples
#define FLOAT_SAMPLES 1 //< 32bit float samples
#endif
/// Define this to allow CPU-specific assembler optimizations. Notice that
/// having this enabled on non-x86 platforms doesn't matter; the compiler can
/// drop unsupported extensions on different platforms automatically.
/// However, if you're having difficulties getting the optimized routines
/// compiled with your compler (e.g. some gcc compiler versions may be picky),
/// you may wish to disable the optimizations to make the library compile.
#define ALLOW_OPTIMIZATIONS 1
// If defined, allows the SIMD-optimized routines to take minor shortcuts
// for improved performance. Undefine to require faithfully similar SIMD
// calculations as in normal C implementation.
#define ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#ifdef INTEGER_SAMPLES
// 16bit integer sample type
typedef short SAMPLETYPE;
// data type for sample accumulation: Use 32bit integer to prevent overflows
typedef long LONG_SAMPLETYPE;
#ifdef FLOAT_SAMPLES
// check that only one sample type is defined
#error "conflicting sample types defined"
#endif // FLOAT_SAMPLES
#ifdef ALLOW_OPTIMIZATIONS
#if (WIN32 || __i386__ || __x86_64__)
// Allow MMX optimizations
#define ALLOW_MMX 1
#endif
#endif
#else
// floating point samples
typedef float SAMPLETYPE;
// data type for sample accumulation: Use double to utilize full precision.
typedef double LONG_SAMPLETYPE;
#ifdef ALLOW_OPTIMIZATIONS
// Allow 3DNow! and SSE optimizations
#if WIN32
#define ALLOW_3DNOW 1
#endif
#if (WIN32 || __i386__ || __x86_64__)
#define ALLOW_SSE 1
#endif
#endif
#endif // INTEGER_SAMPLES
};
#endif
////////////////////////////////////////////////////////////////////////////////
///
/// Common type definitions for SoundTouch audio processing library.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: STTypes.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef STTypes_H
#define STTypes_H
typedef unsigned int uint;
typedef unsigned long ulong;
#ifdef __GNUC__
// In GCC, include soundtouch_config.h made by config scritps
#include "soundtouch_config.h"
#endif
#ifndef _WINDEF_
// if these aren't defined already by Windows headers, define now
typedef int BOOL;
#define FALSE 0
#define TRUE 1
#endif // _WINDEF_
namespace soundtouch
{
/// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file:
//#undef INTEGER_SAMPLES
//#undef FLOAT_SAMPLES
#if !(INTEGER_SAMPLES || FLOAT_SAMPLES)
/// Choose either 32bit floating point or 16bit integer sampletype
/// by choosing one of the following defines, unless this selection
/// has already been done in some other file.
////
/// Notes:
/// - In Windows environment, choose the sample format with the
/// following defines.
/// - In GNU environment, the floating point samples are used by
/// default, but integer samples can be chosen by giving the
/// following switch to the configure script:
/// ./configure --enable-integer-samples
/// However, if you still prefer to select the sample format here
/// also in GNU environment, then please #undef the INTEGER_SAMPLE
/// and FLOAT_SAMPLE defines first as in comments above.
//#define INTEGER_SAMPLES 1 //< 16bit integer samples
#define FLOAT_SAMPLES 1 //< 32bit float samples
#endif
/// Define this to allow CPU-specific assembler optimizations. Notice that
/// having this enabled on non-x86 platforms doesn't matter; the compiler can
/// drop unsupported extensions on different platforms automatically.
/// However, if you're having difficulties getting the optimized routines
/// compiled with your compler (e.g. some gcc compiler versions may be picky),
/// you may wish to disable the optimizations to make the library compile.
#define ALLOW_OPTIMIZATIONS 1
// If defined, allows the SIMD-optimized routines to take minor shortcuts
// for improved performance. Undefine to require faithfully similar SIMD
// calculations as in normal C implementation.
#define ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#ifdef INTEGER_SAMPLES
// 16bit integer sample type
typedef short SAMPLETYPE;
// data type for sample accumulation: Use 32bit integer to prevent overflows
typedef long LONG_SAMPLETYPE;
#ifdef FLOAT_SAMPLES
// check that only one sample type is defined
#error "conflicting sample types defined"
#endif // FLOAT_SAMPLES
#ifdef ALLOW_OPTIMIZATIONS
#if (WIN32 || __i386__ || __x86_64__)
// Allow MMX optimizations
#define ALLOW_MMX 1
#endif
#endif
#else
// floating point samples
typedef float SAMPLETYPE;
// data type for sample accumulation: Use double to utilize full precision.
typedef double LONG_SAMPLETYPE;
#ifdef ALLOW_OPTIMIZATIONS
// Allow 3DNow! and SSE optimizations
#if WIN32
#define ALLOW_3DNOW 1
#endif
#if (WIN32 || __i386__ || __x86_64__)
#define ALLOW_SSE 1
#endif
#endif
#endif // INTEGER_SAMPLES
};
#endif

504
lib-src/soundtouch/soundtouch/SoundTouch.h
View File

@@ -1,252 +1,252 @@
//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: SoundTouch.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef SoundTouch_H
#define SoundTouch_H
#include "FIFOSamplePipe.h"
#include "STTypes.h"
namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "1.3.1"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID 010301
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
#define SETTING_USE_AA_FILTER 0
/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
#define SETTING_AA_FILTER_LENGTH 1
/// Enable/disable quick seeking algorithm in tempo changer routine
/// (enabling quick seeking lowers CPU utilization but causes a minor sound
/// quality compromising)
#define SETTING_USE_QUICKSEEK 2
/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
/// to how long sequences the original sound is chopped in the time-stretch algorithm.
/// See "STTypes.h" or README for more information.
#define SETTING_SEQUENCE_MS 3
/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
/// best possible overlapping location. This determines from how wide window the algorithm
/// may look for an optimal joining location when mixing the sound sequences back together.
/// See "STTypes.h" or README for more information.
#define SETTING_SEEKWINDOW_MS 4
/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
/// are mixed back together, to form a continuous sound stream, this parameter defines over
/// how long period the two consecutive sequences are let to overlap each other.
/// See "STTypes.h" or README for more information.
#define SETTING_OVERLAP_MS 5
class SoundTouch : public FIFOProcessor
{
private:
/// Rate transposer class instance
class RateTransposer *pRateTransposer;
/// Time-stretch class instance
class TDStretch *pTDStretch;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualRate;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualTempo;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualPitch;
/// Flag: Has sample rate been set?
BOOL bSrateSet;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
void calcEffectiveRateAndTempo();
protected :
/// Number of channels
uint channels;
/// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float rate;
/// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float tempo;
public:
SoundTouch();
virtual ~SoundTouch();
/// Get SoundTouch library version string
static const char *getVersionString();
/// Get SoundTouch library version Id
static uint getVersionId();
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
void setRate(float newRate);
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
void setTempo(float newTempo);
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
void setRateChange(float newRate);
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
void setTempoChange(float newTempo);
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
void setPitch(float newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
void setPitchOctaves(float newPitch);
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
void setPitchSemiTones(int newPitch);
void setPitchSemiTones(float newPitch);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
/// Sets sample rate.
void setSampleRate(uint srate);
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
void flush();
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Pointer to sample buffer.
uint numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
virtual void clear();
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'TRUE' if the setting was succesfully changed
BOOL setSetting(uint settingId, ///< Setting ID number. see SETTING_... defines.
uint value ///< New setting value.
);
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
uint getSetting(uint settingId ///< Setting ID number, see SETTING_... defines.
) const;
/// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const;
/// Other handy functions that are implemented in the ancestor classes (see
/// classes 'FIFOProcessor' and 'FIFOSamplePipe')
///
/// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
/// - numSamples() : Get number of 'ready' samples that can be received with
/// function 'receiveSamples()'
/// - isEmpty() : Returns nonzero if there aren't any 'ready' samples.
/// - clear() : Clears all samples from ready/processing buffers.
};
}
#endif
//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: SoundTouch.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef SoundTouch_H
#define SoundTouch_H
#include "FIFOSamplePipe.h"
#include "STTypes.h"
namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "1.3.1"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID 010301
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
#define SETTING_USE_AA_FILTER 0
/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
#define SETTING_AA_FILTER_LENGTH 1
/// Enable/disable quick seeking algorithm in tempo changer routine
/// (enabling quick seeking lowers CPU utilization but causes a minor sound
/// quality compromising)
#define SETTING_USE_QUICKSEEK 2
/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
/// to how long sequences the original sound is chopped in the time-stretch algorithm.
/// See "STTypes.h" or README for more information.
#define SETTING_SEQUENCE_MS 3
/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
/// best possible overlapping location. This determines from how wide window the algorithm
/// may look for an optimal joining location when mixing the sound sequences back together.
/// See "STTypes.h" or README for more information.
#define SETTING_SEEKWINDOW_MS 4
/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
/// are mixed back together, to form a continuous sound stream, this parameter defines over
/// how long period the two consecutive sequences are let to overlap each other.
/// See "STTypes.h" or README for more information.
#define SETTING_OVERLAP_MS 5
class SoundTouch : public FIFOProcessor
{
private:
/// Rate transposer class instance
class RateTransposer *pRateTransposer;
/// Time-stretch class instance
class TDStretch *pTDStretch;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualRate;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualTempo;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualPitch;
/// Flag: Has sample rate been set?
BOOL bSrateSet;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
void calcEffectiveRateAndTempo();
protected :
/// Number of channels
uint channels;
/// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float rate;
/// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float tempo;
public:
SoundTouch();
virtual ~SoundTouch();
/// Get SoundTouch library version string
static const char *getVersionString();
/// Get SoundTouch library version Id
static uint getVersionId();
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
void setRate(float newRate);
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
void setTempo(float newTempo);
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
void setRateChange(float newRate);
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
void setTempoChange(float newTempo);
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
void setPitch(float newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
void setPitchOctaves(float newPitch);
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
void setPitchSemiTones(int newPitch);
void setPitchSemiTones(float newPitch);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
/// Sets sample rate.
void setSampleRate(uint srate);
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
void flush();
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Pointer to sample buffer.
uint numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
virtual void clear();
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'TRUE' if the setting was succesfully changed
BOOL setSetting(uint settingId, ///< Setting ID number. see SETTING_... defines.
uint value ///< New setting value.
);
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
uint getSetting(uint settingId ///< Setting ID number, see SETTING_... defines.
) const;
/// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const;
/// Other handy functions that are implemented in the ancestor classes (see
/// classes 'FIFOProcessor' and 'FIFOSamplePipe')
///
/// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
/// - numSamples() : Get number of 'ready' samples that can be received with
/// function 'receiveSamples()'
/// - isEmpty() : Returns nonzero if there aren't any 'ready' samples.
/// - clear() : Clears all samples from ready/processing buffers.
};
}
#endif

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lib-src/soundtouch/source/SoundTouch/SoundTouch.vcproj
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276
lib-src/soundtouch/source/SoundTouch/cpu_detect_x86_gcc.cpp
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@@ -1,138 +1,138 @@
////////////////////////////////////////////////////////////////////////////////
///
/// gcc version of the x86 CPU detect routine.
///
/// This file is to be compiled on any platform with the GNU C compiler.
/// Compiler. Please see 'cpu_detect_x86_win.cpp' for the x86 Windows version
/// of this file.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:42 $
// File revision : $Revision: 1.3 $
//
// $Id: cpu_detect_x86_gcc.cpp,v 1.3 2006-09-18 07:31:42 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <string>
#include "cpu_detect.h"
#ifndef __GNUC__
#error wrong platform - this source code file is for the GNU C compiler.
#endif
using namespace std;
#include <stdio.h>
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
#ifndef __i386__
return 0; // always disable extensions on non-x86 platforms.
#else
uint res = 0;
if (_dwDisabledISA == 0xffffffff) return 0;
asm volatile(
"\n\txor %%esi, %%esi" // clear %%esi = result register
// check if 'cpuid' instructions is available by toggling eflags bit 21
"\n\tpushf" // save eflags to stack
"\n\tpop %%eax" // load eax from stack (with eflags)
"\n\tmovl %%eax, %%ecx" // save the original eflags values to ecx
"\n\txor $0x00200000, %%eax" // toggle bit 21
"\n\tpush %%eax" // store toggled eflags to stack
"\n\tpopf" // load eflags from stack
"\n\tpushf" // save updated eflags to stack
"\n\tpop %%eax" // load from stack
"\n\txor %%edx, %%edx" // clear edx for defaulting no mmx
"\n\tcmp %%ecx, %%eax" // compare to original eflags values
"\n\tjz end" // jumps to 'end' if cpuid not present
// cpuid instruction available, test for presence of mmx instructions
"\n\tmovl $1, %%eax"
"\n\tcpuid"
// movl $0x00800000, %edx // force enable MMX
"\n\ttest $0x00800000, %%edx"
"\n\tjz end" // branch if MMX not available
"\n\tor $0x01, %%esi" // otherwise add MMX support bit
"\n\ttest $0x02000000, %%edx"
"\n\tjz test3DNow" // branch if SSE not available
"\n\tor $0x08, %%esi" // otherwise add SSE support bit
"\n\ttest3DNow:"
// test for precense of AMD extensions
"\n\tmov $0x80000000, %%eax"
"\n\tcpuid"
"\n\tcmp $0x80000000, %%eax"
"\n\tjbe end" // branch if no AMD extensions detected
// test for precense of 3DNow! extension
"\n\tmov $0x80000001, %%eax"
"\n\tcpuid"
"\n\ttest $0x80000000, %%edx"
"\n\tjz end" // branch if 3DNow! not detected
"\n\tor $0x02, %%esi" // otherwise add 3DNow support bit
"\n\tend:"
"\n\tmov %%esi, %0"
: "=r" (res)
: /* no inputs */
: "%edx", "%eax", "%ecx", "%esi" );
return res & ~_dwDisabledISA;
#endif
}
////////////////////////////////////////////////////////////////////////////////
///
/// gcc version of the x86 CPU detect routine.
///
/// This file is to be compiled on any platform with the GNU C compiler.
/// Compiler. Please see 'cpu_detect_x86_win.cpp' for the x86 Windows version
/// of this file.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:42 $
// File revision : $Revision: 1.3 $
//
// $Id: cpu_detect_x86_gcc.cpp,v 1.3 2006-09-18 07:31:42 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <string>
#include "cpu_detect.h"
#ifndef __GNUC__
#error wrong platform - this source code file is for the GNU C compiler.
#endif
using namespace std;
#include <stdio.h>
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
#ifndef __i386__
return 0; // always disable extensions on non-x86 platforms.
#else
uint res = 0;
if (_dwDisabledISA == 0xffffffff) return 0;
asm volatile(
"\n\txor %%esi, %%esi" // clear %%esi = result register
// check if 'cpuid' instructions is available by toggling eflags bit 21
"\n\tpushf" // save eflags to stack
"\n\tpop %%eax" // load eax from stack (with eflags)
"\n\tmovl %%eax, %%ecx" // save the original eflags values to ecx
"\n\txor $0x00200000, %%eax" // toggle bit 21
"\n\tpush %%eax" // store toggled eflags to stack
"\n\tpopf" // load eflags from stack
"\n\tpushf" // save updated eflags to stack
"\n\tpop %%eax" // load from stack
"\n\txor %%edx, %%edx" // clear edx for defaulting no mmx
"\n\tcmp %%ecx, %%eax" // compare to original eflags values
"\n\tjz end" // jumps to 'end' if cpuid not present
// cpuid instruction available, test for presence of mmx instructions
"\n\tmovl $1, %%eax"
"\n\tcpuid"
// movl $0x00800000, %edx // force enable MMX
"\n\ttest $0x00800000, %%edx"
"\n\tjz end" // branch if MMX not available
"\n\tor $0x01, %%esi" // otherwise add MMX support bit
"\n\ttest $0x02000000, %%edx"
"\n\tjz test3DNow" // branch if SSE not available
"\n\tor $0x08, %%esi" // otherwise add SSE support bit
"\n\ttest3DNow:"
// test for precense of AMD extensions
"\n\tmov $0x80000000, %%eax"
"\n\tcpuid"
"\n\tcmp $0x80000000, %%eax"
"\n\tjbe end" // branch if no AMD extensions detected
// test for precense of 3DNow! extension
"\n\tmov $0x80000001, %%eax"
"\n\tcpuid"
"\n\ttest $0x80000000, %%edx"
"\n\tjz end" // branch if 3DNow! not detected
"\n\tor $0x02, %%esi" // otherwise add 3DNow support bit
"\n\tend:"
"\n\tmov %%esi, %0"
: "=r" (res)
: /* no inputs */
: "%edx", "%eax", "%ecx", "%esi" );
return res & ~_dwDisabledISA;
#endif
}

586
lib-src/soundtouch/source/example/SoundStretch/RunParameters.cpp
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@@ -1,293 +1,293 @@
////////////////////////////////////////////////////////////////////////////////
///
/// A class for parsing the 'soundstretch' application command line parameters
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:47 $
// File revision : $Revision: 1.2 $
//
// $Id: RunParameters.cpp,v 1.2 2006-09-18 07:31:47 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <string>
#include <stdlib.h>
#include "RunParameters.h"
using namespace std;
// Program usage instructions
static const char licenseText[] =
" LICENSE:\n"
" ========\n"
" \n"
" SoundTouch sound processing library\n"
" Copyright (c) Olli Parviainen\n"
" \n"
" This library is free software; you can redistribute it and/or\n"
" modify it under the terms of the GNU Lesser General Public\n"
" License as published by the Free Software Foundation; either\n"
" version 2.1 of the License, or (at your option) any later version.\n"
" \n"
" This library is distributed in the hope that it will be useful,\n"
" but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
" MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU\n"
" Lesser General Public License for more details.\n"
" \n"
" You should have received a copy of the GNU Lesser General Public\n"
" License along with this library; if not, write to the Free Software\n"
" Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA\n"
" \n"
"This application is distributed with full source codes; however, if you\n"
"didn't receive them, please visit the author's homepage (see the link above).";
static const char whatText[] =
"This application processes WAV audio files by modifying the sound tempo,\n"
"pitch and playback rate properties independently from each other.\n"
"\n";
static const char usage[] =
"Usage :\n"
" soundstretch infile.wav outfile.wav [switches]\n\n"
"Available switches are:\n"
" -tempo=n : Change sound tempo by n percents (n=-95..+5000 %)\n"
" -pitch=n : Change sound pitch by n semitones (n=-60..+60 semitones)\n"
" -rate=n : Change sound rate by n percents (n=-95..+5000 %)\n"
" -bpm=n : Detect the BPM rate of sound and adjust tempo to meet 'n' BPMs.\n"
" If '=n' is omitted, just detects the BPM rate.\n"
" -quick : Use quicker tempo change algorithm (gain speed, lose quality)\n"
" -naa : Don't use anti-alias filtering (gain speed, lose quality)\n"
" -license : Display the program license text (LGPL)\n";
// Converts a char into lower case
static int _toLowerCase(int c)
{
if (c >= 'A' && c <= 'Z')
{
c += 'a' - 'A';
}
return c;
}
// Constructor
RunParameters::RunParameters(const int nParams, const char *paramStr[])
{
int i;
int nFirstParam;
if (nParams < 3)
{
// Too few parameters
if (nParams > 1 && paramStr[1][0] == '-' &&
_toLowerCase(paramStr[1][1]) == 'l')
{
// '-license' switch
throwLicense();
}
string msg = whatText;
msg += usage;
throw runtime_error(msg.c_str());
}
inFileName = NULL;
outFileName = NULL;
tempoDelta = 0;
pitchDelta = 0;
rateDelta = 0;
quick = 0;
noAntiAlias = 0;
goalBPM = 0;
detectBPM = FALSE;
// Get input & output file names
inFileName = (char*)paramStr[1];
outFileName = (char*)paramStr[2];
if (outFileName[0] == '-')
{
// no outputfile name was given but parameters
outFileName = NULL;
nFirstParam = 2;
}
else
{
nFirstParam = 3;
}
// parse switch parameters
for (i = nFirstParam; i < nParams; i ++)
{
parseSwitchParam(paramStr[i]);
}
checkLimits();
}
// Checks parameter limits
void RunParameters::checkLimits()
{
if (tempoDelta < -95.0f)
{
tempoDelta = -95.0f;
}
else if (tempoDelta > 5000.0f)
{
tempoDelta = 5000.0f;
}
if (pitchDelta < -60.0f)
{
pitchDelta = -60.0f;
}
else if (pitchDelta > 60.0f)
{
pitchDelta = 60.0f;
}
if (rateDelta < -95.0f)
{
rateDelta = -95.0f;
}
else if (rateDelta > 5000.0f)
{
rateDelta = 5000.0f;
}
}
// Unknown switch parameter -- throws an exception with an error message
void RunParameters::throwIllegalParamExp(const string &str) const
{
string msg = "ERROR : Illegal parameter \"";
msg += str;
msg += "\".\n\n";
msg += usage;
throw runtime_error(msg.c_str());
}
void RunParameters::throwLicense() const
{
throw runtime_error(licenseText);
}
float RunParameters::parseSwitchValue(const string &str) const
{
int pos;
pos = str.find_first_of('=');
if (pos < 0)
{
// '=' missing
throwIllegalParamExp(str);
}
// Read numerical parameter value after '='
return (float)atof(str.substr(pos + 1).c_str());
}
// Interprets a single switch parameter string of format "-switch=xx"
// Valid switches are "-tempo=xx", "-pitch=xx" and "-rate=xx". Stores
// switch values into 'params' structure.
void RunParameters::parseSwitchParam(const string &str)
{
int upS;
if (str[0] != '-')
{
// leading hyphen missing => not a valid parameter
throwIllegalParamExp(str);
}
// Take the first character of switch name & change to lower case
upS = _toLowerCase(str[1]);
// interpret the switch name & operate accordingly
switch (upS)
{
case 't' :
// switch '-tempo=xx'
tempoDelta = parseSwitchValue(str);
break;
case 'p' :
// switch '-pitch=xx'
pitchDelta = parseSwitchValue(str);
break;
case 'r' :
// switch '-rate=xx'
rateDelta = parseSwitchValue(str);
break;
case 'b' :
// switch '-bpm=xx'
detectBPM = TRUE;
try
{
goalBPM = parseSwitchValue(str);
}
catch (runtime_error)
{
// illegal or missing bpm value => just calculate bpm
goalBPM = 0;
}
break;
case 'q' :
// switch '-quick'
quick = 1;
break;
case 'n' :
// switch '-naa'
noAntiAlias = 1;
break;
case 'l' :
// switch '-license'
throwLicense();
break;
default:
// unknown switch
throwIllegalParamExp(str);
}
}
////////////////////////////////////////////////////////////////////////////////
///
/// A class for parsing the 'soundstretch' application command line parameters
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:47 $
// File revision : $Revision: 1.2 $
//
// $Id: RunParameters.cpp,v 1.2 2006-09-18 07:31:47 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <string>
#include <stdlib.h>
#include "RunParameters.h"
using namespace std;
// Program usage instructions
static const char licenseText[] =
" LICENSE:\n"
" ========\n"
" \n"
" SoundTouch sound processing library\n"
" Copyright (c) Olli Parviainen\n"
" \n"
" This library is free software; you can redistribute it and/or\n"
" modify it under the terms of the GNU Lesser General Public\n"
" License as published by the Free Software Foundation; either\n"
" version 2.1 of the License, or (at your option) any later version.\n"
" \n"
" This library is distributed in the hope that it will be useful,\n"
" but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
" MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU\n"
" Lesser General Public License for more details.\n"
" \n"
" You should have received a copy of the GNU Lesser General Public\n"
" License along with this library; if not, write to the Free Software\n"
" Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA\n"
" \n"
"This application is distributed with full source codes; however, if you\n"
"didn't receive them, please visit the author's homepage (see the link above).";
static const char whatText[] =
"This application processes WAV audio files by modifying the sound tempo,\n"
"pitch and playback rate properties independently from each other.\n"
"\n";
static const char usage[] =
"Usage :\n"
" soundstretch infile.wav outfile.wav [switches]\n\n"
"Available switches are:\n"
" -tempo=n : Change sound tempo by n percents (n=-95..+5000 %)\n"
" -pitch=n : Change sound pitch by n semitones (n=-60..+60 semitones)\n"
" -rate=n : Change sound rate by n percents (n=-95..+5000 %)\n"
" -bpm=n : Detect the BPM rate of sound and adjust tempo to meet 'n' BPMs.\n"
" If '=n' is omitted, just detects the BPM rate.\n"
" -quick : Use quicker tempo change algorithm (gain speed, lose quality)\n"
" -naa : Don't use anti-alias filtering (gain speed, lose quality)\n"
" -license : Display the program license text (LGPL)\n";
// Converts a char into lower case
static int _toLowerCase(int c)
{
if (c >= 'A' && c <= 'Z')
{
c += 'a' - 'A';
}
return c;
}
// Constructor
RunParameters::RunParameters(const int nParams, const char *paramStr[])
{
int i;
int nFirstParam;
if (nParams < 3)
{
// Too few parameters
if (nParams > 1 && paramStr[1][0] == '-' &&
_toLowerCase(paramStr[1][1]) == 'l')
{
// '-license' switch
throwLicense();
}
string msg = whatText;
msg += usage;
throw runtime_error(msg.c_str());
}
inFileName = NULL;
outFileName = NULL;
tempoDelta = 0;
pitchDelta = 0;
rateDelta = 0;
quick = 0;
noAntiAlias = 0;
goalBPM = 0;
detectBPM = FALSE;
// Get input & output file names
inFileName = (char*)paramStr[1];
outFileName = (char*)paramStr[2];
if (outFileName[0] == '-')
{
// no outputfile name was given but parameters
outFileName = NULL;
nFirstParam = 2;
}
else
{
nFirstParam = 3;
}
// parse switch parameters
for (i = nFirstParam; i < nParams; i ++)
{
parseSwitchParam(paramStr[i]);
}
checkLimits();
}
// Checks parameter limits
void RunParameters::checkLimits()
{
if (tempoDelta < -95.0f)
{
tempoDelta = -95.0f;
}
else if (tempoDelta > 5000.0f)
{
tempoDelta = 5000.0f;
}
if (pitchDelta < -60.0f)
{
pitchDelta = -60.0f;
}
else if (pitchDelta > 60.0f)
{
pitchDelta = 60.0f;
}
if (rateDelta < -95.0f)
{
rateDelta = -95.0f;
}
else if (rateDelta > 5000.0f)
{
rateDelta = 5000.0f;
}
}
// Unknown switch parameter -- throws an exception with an error message
void RunParameters::throwIllegalParamExp(const string &str) const
{
string msg = "ERROR : Illegal parameter \"";
msg += str;
msg += "\".\n\n";
msg += usage;
throw runtime_error(msg.c_str());
}
void RunParameters::throwLicense() const
{
throw runtime_error(licenseText);
}
float RunParameters::parseSwitchValue(const string &str) const
{
int pos;
pos = str.find_first_of('=');
if (pos < 0)
{
// '=' missing
throwIllegalParamExp(str);
}
// Read numerical parameter value after '='
return (float)atof(str.substr(pos + 1).c_str());
}
// Interprets a single switch parameter string of format "-switch=xx"
// Valid switches are "-tempo=xx", "-pitch=xx" and "-rate=xx". Stores
// switch values into 'params' structure.
void RunParameters::parseSwitchParam(const string &str)
{
int upS;
if (str[0] != '-')
{
// leading hyphen missing => not a valid parameter
throwIllegalParamExp(str);
}
// Take the first character of switch name & change to lower case
upS = _toLowerCase(str[1]);
// interpret the switch name & operate accordingly
switch (upS)
{
case 't' :
// switch '-tempo=xx'
tempoDelta = parseSwitchValue(str);
break;
case 'p' :
// switch '-pitch=xx'
pitchDelta = parseSwitchValue(str);
break;
case 'r' :
// switch '-rate=xx'
rateDelta = parseSwitchValue(str);
break;
case 'b' :
// switch '-bpm=xx'
detectBPM = TRUE;
try
{
goalBPM = parseSwitchValue(str);
}
catch (runtime_error)
{
// illegal or missing bpm value => just calculate bpm
goalBPM = 0;
}
break;
case 'q' :
// switch '-quick'
quick = 1;
break;
case 'n' :
// switch '-naa'
noAntiAlias = 1;
break;
case 'l' :
// switch '-license'
throwLicense();
break;
default:
// unknown switch
throwIllegalParamExp(str);
}
}

142
lib-src/soundtouch/source/example/SoundStretch/RunParameters.h
View File

@@ -1,71 +1,71 @@
////////////////////////////////////////////////////////////////////////////////
///
/// A class for parsing the 'soundstretch' application command line parameters
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:47 $
// File revision : $Revision: 1.2 $
//
// $Id: RunParameters.h,v 1.2 2006-09-18 07:31:47 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef RUNPARAMETERS_H
#define RUNPARAMETERS_H
#include "STTypes.h"
#include <string>
using namespace std;
/// Parses command line parameters into program parameters
class RunParameters
{
private:
void throwIllegalParamExp(const string &str) const;
void throwLicense() const;
void parseSwitchParam(const string &str);
void checkLimits();
float parseSwitchValue(const string &str) const;
public:
char *inFileName;
char *outFileName;
float tempoDelta;
float pitchDelta;
float rateDelta;
int quick;
int noAntiAlias;
float goalBPM;
BOOL detectBPM;
RunParameters(const int nParams, const char *paramStr[]);
};
#endif
////////////////////////////////////////////////////////////////////////////////
///
/// A class for parsing the 'soundstretch' application command line parameters
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:47 $
// File revision : $Revision: 1.2 $
//
// $Id: RunParameters.h,v 1.2 2006-09-18 07:31:47 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef RUNPARAMETERS_H
#define RUNPARAMETERS_H
#include "STTypes.h"
#include <string>
using namespace std;
/// Parses command line parameters into program parameters
class RunParameters
{
private:
void throwIllegalParamExp(const string &str) const;
void throwLicense() const;
void parseSwitchParam(const string &str);
void checkLimits();
float parseSwitchValue(const string &str) const;
public:
char *inFileName;
char *outFileName;
float tempoDelta;
float pitchDelta;
float rateDelta;
int quick;
int noAntiAlias;
float goalBPM;
BOOL detectBPM;
RunParameters(const int nParams, const char *paramStr[]);
};
#endif

1424
lib-src/soundtouch/source/example/SoundStretch/WavFile.cpp
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506
lib-src/soundtouch/source/example/SoundStretch/WavFile.h
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@@ -1,253 +1,253 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define BIG_ENDIAN during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain, but
/// the reason for 'yet another' one is that those generic WAV reader libraries are
/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e.
/// something that's not already larger than rest of the SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: WavFile.h,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef WAVFILE_H
#define WAVFILE_H
#include <stdio.h>
#ifndef uint
typedef unsigned int uint;
#endif
/// WAV audio file 'riff' section header
typedef struct
{
char riff_char[4];
int package_len;
char wave[4];
} WavRiff;
/// WAV audio file 'format' section header
typedef struct
{
char fmt[4];
int format_len;
short fixed;
short channel_number;
int sample_rate;
int byte_rate;
short byte_per_sample;
short bits_per_sample;
} WavFormat;
/// WAV audio file 'data' section header
typedef struct
{
char data_field[4];
uint data_len;
} WavData;
/// WAV audio file header
typedef struct
{
WavRiff riff;
WavFormat format;
WavData data;
} WavHeader;
/// Class for reading WAV audio files.
class WavInFile
{
private:
/// File pointer.
FILE *fptr;
/// Counter of how many bytes of sample data have been read from the file.
uint dataRead;
/// WAV header information
WavHeader header;
/// Read WAV file headers.
/// \return zero if all ok, nonzero if file format is invalid.
int readWavHeaders();
/// Checks WAV file header tags.
/// \return zero if all ok, nonzero if file format is invalid.
int checkCharTags();
/// Reads a single WAV file header block.
/// \return zero if all ok, nonzero if file format is invalid.
int readHeaderBlock();
/// Reads WAV file 'riff' block
int readRIFFBlock();
public:
/// Constructor: Opens the given WAV file. If the file can't be opened,
/// throws 'runtime_error' exception.
WavInFile(const char *filename);
/// Destructor: Closes the file.
~WavInFile();
/// Close the file. Notice that file is automatically closed also when the
/// class instance is deleted.
void close();
/// Rewind to beginning of the file
void rewind();
/// Get sample rate.
uint getSampleRate() const;
/// Get number of bits per sample, i.e. 8 or 16.
uint getNumBits() const;
/// Get sample data size in bytes. Ahem, this should return same information as
/// 'getBytesPerSample'...
uint getDataSizeInBytes() const;
/// Get total number of samples in file.
uint getNumSamples() const;
/// Get number of bytes per audio sample (e.g. 16bit stereo = 4 bytes/sample)
uint getBytesPerSample() const;
/// Get number of audio channels in the file (1=mono, 2=stereo)
uint getNumChannels() const;
/// Get the audio file length in milliseconds
uint getLengthMS() const;
/// Reads audio samples from the WAV file. This routine works only for 8 bit samples.
/// Reads given number of elements from the file or if end-of-file reached, as many
/// elements as are left in the file.
///
/// \return Number of 8-bit integers read from the file.
int read(char *buffer, int maxElems);
/// Reads audio samples from the WAV file to 16 bit integer format. Reads given number
/// of elements from the file or if end-of-file reached, as many elements as are
/// left in the file.
///
/// \return Number of 16-bit integers read from the file.
int read(short *buffer, ///< Pointer to buffer where to read data.
int maxElems ///< Size of 'buffer' array (number of array elements).
);
/// Reads audio samples from the WAV file to floating point format, converting
/// sample values to range [-1,1[. Reads given number of elements from the file
/// or if end-of-file reached, as many elements as are left in the file.
///
/// \return Number of elements read from the file.
int read(float *buffer, ///< Pointer to buffer where to read data.
int maxElems ///< Size of 'buffer' array (number of array elements).
);
/// Check end-of-file.
///
/// \return Nonzero if end-of-file reached.
int eof() const;
};
/// Class for writing WAV audio files.
class WavOutFile
{
private:
/// Pointer to the WAV file
FILE *fptr;
/// WAV file header data.
WavHeader header;
/// Counter of how many bytes have been written to the file so far.
int bytesWritten;
/// Fills in WAV file header information.
void fillInHeader(const uint sampleRate, const uint bits, const uint channels);
/// Finishes the WAV file header by supplementing information of amount of
/// data written to file etc
void finishHeader();
/// Writes the WAV file header.
void writeHeader();
public:
/// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
/// if file creation fails.
WavOutFile(const char *fileName, ///< Filename
int sampleRate, ///< Sample rate (e.g. 44100 etc)
int bits, ///< Bits per sample (8 or 16 bits)
int channels ///< Number of channels (1=mono, 2=stereo)
);
/// Destructor: Finalizes & closes the WAV file.
~WavOutFile();
/// Write data to WAV file. This function works only with 8bit samples.
/// Throws a 'runtime_error' exception if writing to file fails.
void write(const char *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Write data to WAV file. Throws a 'runtime_error' exception if writing to
/// file fails.
void write(const short *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Write data to WAV file in floating point format, saturating sample values to range
/// [-1..+1[. Throws a 'runtime_error' exception if writing to file fails.
void write(const float *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Finalize & close the WAV file. Automatically supplements the WAV file header
/// information according to written data etc.
///
/// Notice that file is automatically closed also when the class instance is deleted.
void close();
};
#endif
////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define BIG_ENDIAN during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain, but
/// the reason for 'yet another' one is that those generic WAV reader libraries are
/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e.
/// something that's not already larger than rest of the SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: WavFile.h,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef WAVFILE_H
#define WAVFILE_H
#include <stdio.h>
#ifndef uint
typedef unsigned int uint;
#endif
/// WAV audio file 'riff' section header
typedef struct
{
char riff_char[4];
int package_len;
char wave[4];
} WavRiff;
/// WAV audio file 'format' section header
typedef struct
{
char fmt[4];
int format_len;
short fixed;
short channel_number;
int sample_rate;
int byte_rate;
short byte_per_sample;
short bits_per_sample;
} WavFormat;
/// WAV audio file 'data' section header
typedef struct
{
char data_field[4];
uint data_len;
} WavData;
/// WAV audio file header
typedef struct
{
WavRiff riff;
WavFormat format;
WavData data;
} WavHeader;
/// Class for reading WAV audio files.
class WavInFile
{
private:
/// File pointer.
FILE *fptr;
/// Counter of how many bytes of sample data have been read from the file.
uint dataRead;
/// WAV header information
WavHeader header;
/// Read WAV file headers.
/// \return zero if all ok, nonzero if file format is invalid.
int readWavHeaders();
/// Checks WAV file header tags.
/// \return zero if all ok, nonzero if file format is invalid.
int checkCharTags();
/// Reads a single WAV file header block.
/// \return zero if all ok, nonzero if file format is invalid.
int readHeaderBlock();
/// Reads WAV file 'riff' block
int readRIFFBlock();
public:
/// Constructor: Opens the given WAV file. If the file can't be opened,
/// throws 'runtime_error' exception.
WavInFile(const char *filename);
/// Destructor: Closes the file.
~WavInFile();
/// Close the file. Notice that file is automatically closed also when the
/// class instance is deleted.
void close();
/// Rewind to beginning of the file
void rewind();
/// Get sample rate.
uint getSampleRate() const;
/// Get number of bits per sample, i.e. 8 or 16.
uint getNumBits() const;
/// Get sample data size in bytes. Ahem, this should return same information as
/// 'getBytesPerSample'...
uint getDataSizeInBytes() const;
/// Get total number of samples in file.
uint getNumSamples() const;
/// Get number of bytes per audio sample (e.g. 16bit stereo = 4 bytes/sample)
uint getBytesPerSample() const;
/// Get number of audio channels in the file (1=mono, 2=stereo)
uint getNumChannels() const;
/// Get the audio file length in milliseconds
uint getLengthMS() const;
/// Reads audio samples from the WAV file. This routine works only for 8 bit samples.
/// Reads given number of elements from the file or if end-of-file reached, as many
/// elements as are left in the file.
///
/// \return Number of 8-bit integers read from the file.
int read(char *buffer, int maxElems);
/// Reads audio samples from the WAV file to 16 bit integer format. Reads given number
/// of elements from the file or if end-of-file reached, as many elements as are
/// left in the file.
///
/// \return Number of 16-bit integers read from the file.
int read(short *buffer, ///< Pointer to buffer where to read data.
int maxElems ///< Size of 'buffer' array (number of array elements).
);
/// Reads audio samples from the WAV file to floating point format, converting
/// sample values to range [-1,1[. Reads given number of elements from the file
/// or if end-of-file reached, as many elements as are left in the file.
///
/// \return Number of elements read from the file.
int read(float *buffer, ///< Pointer to buffer where to read data.
int maxElems ///< Size of 'buffer' array (number of array elements).
);
/// Check end-of-file.
///
/// \return Nonzero if end-of-file reached.
int eof() const;
};
/// Class for writing WAV audio files.
class WavOutFile
{
private:
/// Pointer to the WAV file
FILE *fptr;
/// WAV file header data.
WavHeader header;
/// Counter of how many bytes have been written to the file so far.
int bytesWritten;
/// Fills in WAV file header information.
void fillInHeader(const uint sampleRate, const uint bits, const uint channels);
/// Finishes the WAV file header by supplementing information of amount of
/// data written to file etc
void finishHeader();
/// Writes the WAV file header.
void writeHeader();
public:
/// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
/// if file creation fails.
WavOutFile(const char *fileName, ///< Filename
int sampleRate, ///< Sample rate (e.g. 44100 etc)
int bits, ///< Bits per sample (8 or 16 bits)
int channels ///< Number of channels (1=mono, 2=stereo)
);
/// Destructor: Finalizes & closes the WAV file.
~WavOutFile();
/// Write data to WAV file. This function works only with 8bit samples.
/// Throws a 'runtime_error' exception if writing to file fails.
void write(const char *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Write data to WAV file. Throws a 'runtime_error' exception if writing to
/// file fails.
void write(const short *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Write data to WAV file in floating point format, saturating sample values to range
/// [-1..+1[. Throws a 'runtime_error' exception if writing to file fails.
void write(const float *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Finalize & close the WAV file. Automatically supplements the WAV file header
/// information according to written data etc.
///
/// Notice that file is automatically closed also when the class instance is deleted.
void close();
};
#endif

576
lib-src/soundtouch/source/example/SoundStretch/main.cpp
View File

@@ -1,288 +1,288 @@
////////////////////////////////////////////////////////////////////////////////
///
/// SoundStretch main routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.3 $
//
// $Id: main.cpp,v 1.3 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <stdio.h>
#include "RunParameters.h"
#include "WavFile.h"
#include "SoundTouch.h"
#include "BPMDetect.h"
using namespace soundtouch;
using namespace std;
// Processing chunk size
#define BUFF_SIZE 2048
static const char _helloText[] =
"\n"
" SoundStretch v%s - Written by Olli Parviainen 2001 - 2006\n"
"==================================================================\n"
"author e-mail: <oparviai@iki.fi> - WWW: http://www.surina.net/soundtouch\n"
"\n"
"This program is subject to (L)GPL license. Run \"soundstretch -license\" for\n"
"more information.\n"
"\n";
static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParameters *params)
{
int bits, samplerate, channels;
// open input file...
*inFile = new WavInFile(params->inFileName);
// ... open output file with same sound parameters
bits = (*inFile)->getNumBits();
samplerate = (*inFile)->getSampleRate();
channels = (*inFile)->getNumChannels();
if (params->outFileName)
{
*outFile = new WavOutFile(params->outFileName, samplerate, bits, channels);
}
else
{
*outFile = NULL;
}
}
// Sets the 'SoundTouch' object up according to input file sound format &
// command line parameters
static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunParameters *params)
{
int sampleRate;
int channels;
sampleRate = inFile->getSampleRate();
channels = inFile->getNumChannels();
pSoundTouch->setSampleRate(sampleRate);
pSoundTouch->setChannels(channels);
pSoundTouch->setTempoChange(params->tempoDelta);
pSoundTouch->setPitchSemiTones(params->pitchDelta);
pSoundTouch->setRateChange(params->rateDelta);
pSoundTouch->setSetting(SETTING_USE_QUICKSEEK, params->quick);
pSoundTouch->setSetting(SETTING_USE_AA_FILTER, !params->noAntiAlias);
// print processing information
if (params->outFileName)
{
#ifdef INTEGER_SAMPLES
printf("Uses 16bit integer sample type in processing.\n\n");
#else
#ifndef FLOAT_SAMPLES
#error "Sampletype not defined"
#endif
printf("Uses 32bit floating point sample type in processing.\n\n");
#endif
// print processing information only if outFileName given i.e. some processing will happen
printf("Processing the file with the following changes:\n");
printf(" tempo change = %+g %%\n", params->tempoDelta);
printf(" pitch change = %+g semitones\n", params->pitchDelta);
printf(" rate change = %+g %%\n\n", params->rateDelta);
printf("Working...");
}
else
{
// outFileName not given
printf("Warning: output file name missing, won't output anything.\n\n");
}
fflush(stdout);
}
// Processes the sound
static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outFile)
{
int nSamples;
int nChannels;
int buffSizeSamples;
SAMPLETYPE sampleBuffer[BUFF_SIZE];
if ((inFile == NULL) || (outFile == NULL)) return; // nothing to do.
nChannels = inFile->getNumChannels();
buffSizeSamples = BUFF_SIZE / nChannels;
// Process samples read from the input file
while (inFile->eof() == 0)
{
int num;
// Read a chunk of samples from the input file
num = inFile->read(sampleBuffer, BUFF_SIZE);
nSamples = num / inFile->getNumChannels();
// Feed the samples into SoundTouch processor
pSoundTouch->putSamples(sampleBuffer, nSamples);
// Read ready samples from SoundTouch processor & write them output file.
// NOTES:
// - 'receiveSamples' doesn't necessarily return any samples at all
// during some rounds!
// - On the other hand, during some round 'receiveSamples' may have more
// ready samples than would fit into 'sampleBuffer', and for this reason
// the 'receiveSamples' call is iterated for as many times as it
// outputs samples.
do
{
nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
outFile->write(sampleBuffer, nSamples * nChannels);
} while (nSamples != 0);
}
// Now the input file is processed, yet 'flush' few last samples that are
// hiding in the SoundTouch's internal processing pipeline.
pSoundTouch->flush();
do
{
nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
outFile->write(sampleBuffer, nSamples * nChannels);
} while (nSamples != 0);
}
// Detect BPM rate of inFile and adjust tempo setting accordingly if necessary
static void detectBPM(WavInFile *inFile, RunParameters *params)
{
float bpmValue;
int nChannels;
BPMDetect bpm(inFile->getNumChannels(), inFile->getSampleRate());
SAMPLETYPE sampleBuffer[BUFF_SIZE];
// detect bpm rate
printf("Detecting BPM rate...");
fflush(stdout);
nChannels = inFile->getNumChannels();
// Process the 'inFile' in small blocks, repeat until whole file has
// been processed
while (inFile->eof() == 0)
{
int num, samples;
// Read sample data from input file
num = inFile->read(sampleBuffer, BUFF_SIZE);
// Enter the new samples to the bpm analyzer class
samples = num / nChannels;
bpm.inputSamples(sampleBuffer, samples);
}
// Now the whole song data has been analyzed. Read the resulting bpm.
bpmValue = bpm.getBpm();
printf("Done!\n");
// rewind the file after bpm detection
inFile->rewind();
if (bpmValue > 0)
{
printf("Detected BPM rate %.1f\n\n", bpmValue);
}
else
{
printf("Couldn't detect BPM rate.\n\n");
return;
}
if (params->goalBPM > 0)
{
// adjust tempo to given bpm
params->tempoDelta = (params->goalBPM / bpmValue - 1.0f) * 100.0f;
printf("The file will be converted to %.1f BPM\n\n", params->goalBPM);
}
}
int main(const int nParams, const char *paramStr[])
{
WavInFile *inFile;
WavOutFile *outFile;
RunParameters *params;
SoundTouch SoundTouch;
printf(_helloText, SoundTouch::getVersionString());
try
{
// Parse command line parameters
params = new RunParameters(nParams, paramStr);
// Open input & output files
openFiles(&inFile, &outFile, params);
if (params->detectBPM == TRUE)
{
// detect sound BPM (and adjust processing parameters
// accordingly if necessary)
detectBPM(inFile, params);
}
// Setup the 'SoundTouch' object for processing the sound
setup(&SoundTouch, inFile, params);
// Process the sound
process(&SoundTouch, inFile, outFile);
// Close WAV file handles & dispose of the objects
delete inFile;
delete outFile;
delete params;
printf("Done!\n");
}
catch (runtime_error &e)
{
// An exception occurred during processing, display an error message
printf("%s\n", e.what());
return -1;
}
return 0;
}
////////////////////////////////////////////////////////////////////////////////
///
/// SoundStretch main routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.3 $
//
// $Id: main.cpp,v 1.3 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <stdio.h>
#include "RunParameters.h"
#include "WavFile.h"
#include "SoundTouch.h"
#include "BPMDetect.h"
using namespace soundtouch;
using namespace std;
// Processing chunk size
#define BUFF_SIZE 2048
static const char _helloText[] =
"\n"
" SoundStretch v%s - Written by Olli Parviainen 2001 - 2006\n"
"==================================================================\n"
"author e-mail: <oparviai@iki.fi> - WWW: http://www.surina.net/soundtouch\n"
"\n"
"This program is subject to (L)GPL license. Run \"soundstretch -license\" for\n"
"more information.\n"
"\n";
static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParameters *params)
{
int bits, samplerate, channels;
// open input file...
*inFile = new WavInFile(params->inFileName);
// ... open output file with same sound parameters
bits = (*inFile)->getNumBits();
samplerate = (*inFile)->getSampleRate();
channels = (*inFile)->getNumChannels();
if (params->outFileName)
{
*outFile = new WavOutFile(params->outFileName, samplerate, bits, channels);
}
else
{
*outFile = NULL;
}
}
// Sets the 'SoundTouch' object up according to input file sound format &
// command line parameters
static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunParameters *params)
{
int sampleRate;
int channels;
sampleRate = inFile->getSampleRate();
channels = inFile->getNumChannels();
pSoundTouch->setSampleRate(sampleRate);
pSoundTouch->setChannels(channels);
pSoundTouch->setTempoChange(params->tempoDelta);
pSoundTouch->setPitchSemiTones(params->pitchDelta);
pSoundTouch->setRateChange(params->rateDelta);
pSoundTouch->setSetting(SETTING_USE_QUICKSEEK, params->quick);
pSoundTouch->setSetting(SETTING_USE_AA_FILTER, !params->noAntiAlias);
// print processing information
if (params->outFileName)
{
#ifdef INTEGER_SAMPLES
printf("Uses 16bit integer sample type in processing.\n\n");
#else
#ifndef FLOAT_SAMPLES
#error "Sampletype not defined"
#endif
printf("Uses 32bit floating point sample type in processing.\n\n");
#endif
// print processing information only if outFileName given i.e. some processing will happen
printf("Processing the file with the following changes:\n");
printf(" tempo change = %+g %%\n", params->tempoDelta);
printf(" pitch change = %+g semitones\n", params->pitchDelta);
printf(" rate change = %+g %%\n\n", params->rateDelta);
printf("Working...");
}
else
{
// outFileName not given
printf("Warning: output file name missing, won't output anything.\n\n");
}
fflush(stdout);
}
// Processes the sound
static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outFile)
{
int nSamples;
int nChannels;
int buffSizeSamples;
SAMPLETYPE sampleBuffer[BUFF_SIZE];
if ((inFile == NULL) || (outFile == NULL)) return; // nothing to do.
nChannels = inFile->getNumChannels();
buffSizeSamples = BUFF_SIZE / nChannels;
// Process samples read from the input file
while (inFile->eof() == 0)
{
int num;
// Read a chunk of samples from the input file
num = inFile->read(sampleBuffer, BUFF_SIZE);
nSamples = num / inFile->getNumChannels();
// Feed the samples into SoundTouch processor
pSoundTouch->putSamples(sampleBuffer, nSamples);
// Read ready samples from SoundTouch processor & write them output file.
// NOTES:
// - 'receiveSamples' doesn't necessarily return any samples at all
// during some rounds!
// - On the other hand, during some round 'receiveSamples' may have more
// ready samples than would fit into 'sampleBuffer', and for this reason
// the 'receiveSamples' call is iterated for as many times as it
// outputs samples.
do
{
nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
outFile->write(sampleBuffer, nSamples * nChannels);
} while (nSamples != 0);
}
// Now the input file is processed, yet 'flush' few last samples that are
// hiding in the SoundTouch's internal processing pipeline.
pSoundTouch->flush();
do
{
nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
outFile->write(sampleBuffer, nSamples * nChannels);
} while (nSamples != 0);
}
// Detect BPM rate of inFile and adjust tempo setting accordingly if necessary
static void detectBPM(WavInFile *inFile, RunParameters *params)
{
float bpmValue;
int nChannels;
BPMDetect bpm(inFile->getNumChannels(), inFile->getSampleRate());
SAMPLETYPE sampleBuffer[BUFF_SIZE];
// detect bpm rate
printf("Detecting BPM rate...");
fflush(stdout);
nChannels = inFile->getNumChannels();
// Process the 'inFile' in small blocks, repeat until whole file has
// been processed
while (inFile->eof() == 0)
{
int num, samples;
// Read sample data from input file
num = inFile->read(sampleBuffer, BUFF_SIZE);
// Enter the new samples to the bpm analyzer class
samples = num / nChannels;
bpm.inputSamples(sampleBuffer, samples);
}
// Now the whole song data has been analyzed. Read the resulting bpm.
bpmValue = bpm.getBpm();
printf("Done!\n");
// rewind the file after bpm detection
inFile->rewind();
if (bpmValue > 0)
{
printf("Detected BPM rate %.1f\n\n", bpmValue);
}
else
{
printf("Couldn't detect BPM rate.\n\n");
return;
}
if (params->goalBPM > 0)
{
// adjust tempo to given bpm
params->tempoDelta = (params->goalBPM / bpmValue - 1.0f) * 100.0f;
printf("The file will be converted to %.1f BPM\n\n", params->goalBPM);
}
}
int main(const int nParams, const char *paramStr[])
{
WavInFile *inFile;
WavOutFile *outFile;
RunParameters *params;
SoundTouch SoundTouch;
printf(_helloText, SoundTouch::getVersionString());
try
{
// Parse command line parameters
params = new RunParameters(nParams, paramStr);
// Open input & output files
openFiles(&inFile, &outFile, params);
if (params->detectBPM == TRUE)
{
// detect sound BPM (and adjust processing parameters
// accordingly if necessary)
detectBPM(inFile, params);
}
// Setup the 'SoundTouch' object for processing the sound
setup(&SoundTouch, inFile, params);
// Process the sound
process(&SoundTouch, inFile, outFile);
// Close WAV file handles & dispose of the objects
delete inFile;
delete outFile;
delete params;
printf("Done!\n");
}
catch (runtime_error &e)
{
// An exception occurred during processing, display an error message
printf("%s\n", e.what());
return -1;
}
return 0;
}

622
lib-src/soundtouch/source/example/bpm/BPMDetect.cpp
View File

@@ -1,311 +1,311 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: BPMDetect.cpp,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include <string.h>
#include "FIFOSampleBuffer.h"
#include "PeakFinder.h"
#include "BPMDetect.h"
using namespace soundtouch;
#define INPUT_BLOCK_SAMPLES 2048
#define DECIMATED_BLOCK_SAMPLES 256
typedef unsigned short ushort;
/// decay constant for calculating RMS volume sliding average approximation
/// (time constant is about 10 sec)
const float avgdecay = 0.99986f;
/// Normalization coefficient for calculating RMS sliding average approximation.
const float avgnorm = (1 - avgdecay);
BPMDetect::BPMDetect(int numChannels, int sampleRate)
{
xcorr = NULL;
buffer = new FIFOSampleBuffer();
decimateSum = 0;
decimateCount = 0;
decimateBy = 0;
this->sampleRate = sampleRate;
this->channels = numChannels;
envelopeAccu = 0;
// Initialize RMS volume accumulator to RMS level of 3000 (out of 32768) that's
// a typical RMS signal level value for song data. This value is then adapted
// to the actual level during processing.
#ifdef INTEGER_SAMPLES
// integer samples
RMSVolumeAccu = (3000 * 3000) / avgnorm;
#else
// float samples, scaled to range [-1..+1[
RMSVolumeAccu = (0.092f * 0.092f) / avgnorm;
#endif
init(numChannels, sampleRate);
}
BPMDetect::~BPMDetect()
{
delete[] xcorr;
delete buffer;
}
/// low-pass filter & decimate to about 500 Hz. return number of outputted samples.
///
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
/// the amount of data needed to be processed as calculating autocorrelation
/// function is a very-very heavy operation.
///
/// Anti-alias filtering is done simply by averaging the samples. This is really a
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
/// narrow band)
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
{
int count, outcount;
LONG_SAMPLETYPE out;
assert(decimateBy != 0);
outcount = 0;
for (count = 0; count < numsamples; count ++)
{
decimateSum += src[count];
decimateCount ++;
if (decimateCount >= decimateBy)
{
// Store every Nth sample only
out = (LONG_SAMPLETYPE)(decimateSum / decimateBy);
decimateSum = 0;
decimateCount = 0;
#ifdef INTEGER_SAMPLES
// check ranges for sure (shouldn't actually be necessary)
if (out > 32767)
{
out = 32767;
}
else if (out < -32768)
{
out = -32768;
}
#endif // INTEGER_SAMPLES
dest[outcount] = (SAMPLETYPE)out;
outcount ++;
}
}
return outcount;
}
// Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples)
{
int offs;
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
pBuffer = buffer->ptrBegin();
for (offs = windowStart; offs < windowLen; offs ++)
{
LONG_SAMPLETYPE sum;
int i;
sum = 0;
for (i = 0; i < process_samples; i ++)
{
sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
}
// xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable coefficients
// if it's desired that the system adapts automatically to
// various bpms, e.g. in processing continouos music stream.
// The 'xcorr_decay' should be a value that's smaller than but
// close to one, and should also depend on 'process_samples' value.
xcorr[offs] += (float)sum;
}
}
// Calculates envelope of the sample data
void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
{
const float decay = 0.7f; // decay constant for smoothing the envelope
const float norm = (1 - decay);
int i;
LONG_SAMPLETYPE out;
float val;
for (i = 0; i < numsamples; i ++)
{
// calc average RMS volume
RMSVolumeAccu *= avgdecay;
val = (float)fabs((float)samples[i]);
RMSVolumeAccu += val * val;
// cut amplitudes that are below 2 times average RMS volume
// (we're interested in peak values, not the silent moments)
val -= 2 * (float)sqrt(RMSVolumeAccu * avgnorm);
val = (val > 0) ? val : 0;
// smooth amplitude envelope
envelopeAccu *= decay;
envelopeAccu += val;
out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
#ifdef INTEGER_SAMPLES
// cut peaks (shouldn't be necessary though)
if (out > 32767) out = 32767;
#endif // INTEGER_SAMPLES
samples[i] = (SAMPLETYPE)out;
}
}
void BPMDetect::inputSamples(SAMPLETYPE *samples, int numSamples)
{
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
// convert from stereo to mono if necessary
if (channels == 2)
{
int i;
for (i = 0; i < numSamples; i ++)
{
samples[i] = (samples[i * 2] + samples[i * 2 + 1]) / 2;
}
}
// decimate
numSamples = decimate(decimated, samples, numSamples);
// envelope new samples and add them to buffer
calcEnvelope(decimated, numSamples);
buffer->putSamples(decimated, numSamples);
// when the buffer has enought samples for processing...
if ((int)buffer->numSamples() > windowLen)
{
int processLength;
// how many samples are processed
processLength = buffer->numSamples() - windowLen;
// ... calculate autocorrelations for oldest samples...
updateXCorr(processLength);
// ... and remove them from the buffer
buffer->receiveSamples(processLength);
}
}
void BPMDetect::init(int numChannels, int sampleRate)
{
this->sampleRate = sampleRate;
// choose decimation factor so that result is approx. 500 Hz
decimateBy = sampleRate / 500;
assert(decimateBy > 0);
assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
// Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
assert(windowLen > windowStart);
// allocate new working objects
xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float));
// we do processing in mono mode
buffer->setChannels(1);
buffer->clear();
}
float BPMDetect::getBpm()
{
float peakPos;
PeakFinder peakFinder;
// find peak position
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
assert(decimateBy != 0);
if (peakPos < 1e-6) return 0.0; // detection failed.
// calculate BPM
return 60.0f * (((float)sampleRate / (float)decimateBy) / peakPos);
}
////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: BPMDetect.cpp,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include <string.h>
#include "FIFOSampleBuffer.h"
#include "PeakFinder.h"
#include "BPMDetect.h"
using namespace soundtouch;
#define INPUT_BLOCK_SAMPLES 2048
#define DECIMATED_BLOCK_SAMPLES 256
typedef unsigned short ushort;
/// decay constant for calculating RMS volume sliding average approximation
/// (time constant is about 10 sec)
const float avgdecay = 0.99986f;
/// Normalization coefficient for calculating RMS sliding average approximation.
const float avgnorm = (1 - avgdecay);
BPMDetect::BPMDetect(int numChannels, int sampleRate)
{
xcorr = NULL;
buffer = new FIFOSampleBuffer();
decimateSum = 0;
decimateCount = 0;
decimateBy = 0;
this->sampleRate = sampleRate;
this->channels = numChannels;
envelopeAccu = 0;
// Initialize RMS volume accumulator to RMS level of 3000 (out of 32768) that's
// a typical RMS signal level value for song data. This value is then adapted
// to the actual level during processing.
#ifdef INTEGER_SAMPLES
// integer samples
RMSVolumeAccu = (3000 * 3000) / avgnorm;
#else
// float samples, scaled to range [-1..+1[
RMSVolumeAccu = (0.092f * 0.092f) / avgnorm;
#endif
init(numChannels, sampleRate);
}
BPMDetect::~BPMDetect()
{
delete[] xcorr;
delete buffer;
}
/// low-pass filter & decimate to about 500 Hz. return number of outputted samples.
///
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
/// the amount of data needed to be processed as calculating autocorrelation
/// function is a very-very heavy operation.
///
/// Anti-alias filtering is done simply by averaging the samples. This is really a
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
/// narrow band)
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
{
int count, outcount;
LONG_SAMPLETYPE out;
assert(decimateBy != 0);
outcount = 0;
for (count = 0; count < numsamples; count ++)
{
decimateSum += src[count];
decimateCount ++;
if (decimateCount >= decimateBy)
{
// Store every Nth sample only
out = (LONG_SAMPLETYPE)(decimateSum / decimateBy);
decimateSum = 0;
decimateCount = 0;
#ifdef INTEGER_SAMPLES
// check ranges for sure (shouldn't actually be necessary)
if (out > 32767)
{
out = 32767;
}
else if (out < -32768)
{
out = -32768;
}
#endif // INTEGER_SAMPLES
dest[outcount] = (SAMPLETYPE)out;
outcount ++;
}
}
return outcount;
}
// Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples)
{
int offs;
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
pBuffer = buffer->ptrBegin();
for (offs = windowStart; offs < windowLen; offs ++)
{
LONG_SAMPLETYPE sum;
int i;
sum = 0;
for (i = 0; i < process_samples; i ++)
{
sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
}
// xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable coefficients
// if it's desired that the system adapts automatically to
// various bpms, e.g. in processing continouos music stream.
// The 'xcorr_decay' should be a value that's smaller than but
// close to one, and should also depend on 'process_samples' value.
xcorr[offs] += (float)sum;
}
}
// Calculates envelope of the sample data
void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
{
const float decay = 0.7f; // decay constant for smoothing the envelope
const float norm = (1 - decay);
int i;
LONG_SAMPLETYPE out;
float val;
for (i = 0; i < numsamples; i ++)
{
// calc average RMS volume
RMSVolumeAccu *= avgdecay;
val = (float)fabs((float)samples[i]);
RMSVolumeAccu += val * val;
// cut amplitudes that are below 2 times average RMS volume
// (we're interested in peak values, not the silent moments)
val -= 2 * (float)sqrt(RMSVolumeAccu * avgnorm);
val = (val > 0) ? val : 0;
// smooth amplitude envelope
envelopeAccu *= decay;
envelopeAccu += val;
out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
#ifdef INTEGER_SAMPLES
// cut peaks (shouldn't be necessary though)
if (out > 32767) out = 32767;
#endif // INTEGER_SAMPLES
samples[i] = (SAMPLETYPE)out;
}
}
void BPMDetect::inputSamples(SAMPLETYPE *samples, int numSamples)
{
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
// convert from stereo to mono if necessary
if (channels == 2)
{
int i;
for (i = 0; i < numSamples; i ++)
{
samples[i] = (samples[i * 2] + samples[i * 2 + 1]) / 2;
}
}
// decimate
numSamples = decimate(decimated, samples, numSamples);
// envelope new samples and add them to buffer
calcEnvelope(decimated, numSamples);
buffer->putSamples(decimated, numSamples);
// when the buffer has enought samples for processing...
if ((int)buffer->numSamples() > windowLen)
{
int processLength;
// how many samples are processed
processLength = buffer->numSamples() - windowLen;
// ... calculate autocorrelations for oldest samples...
updateXCorr(processLength);
// ... and remove them from the buffer
buffer->receiveSamples(processLength);
}
}
void BPMDetect::init(int numChannels, int sampleRate)
{
this->sampleRate = sampleRate;
// choose decimation factor so that result is approx. 500 Hz
decimateBy = sampleRate / 500;
assert(decimateBy > 0);
assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
// Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
assert(windowLen > windowStart);
// allocate new working objects
xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float));
// we do processing in mono mode
buffer->setChannels(1);
buffer->clear();
}
float BPMDetect::getBpm()
{
float peakPos;
PeakFinder peakFinder;
// find peak position
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
assert(decimateBy != 0);
if (peakPos < 1e-6) return 0.0; // detection failed.
// calculate BPM
return 60.0f * (((float)sampleRate / (float)decimateBy) / peakPos);
}

382
lib-src/soundtouch/source/example/bpm/PeakFinder.cpp
View File

@@ -1,191 +1,191 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Peak detection routine.
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: PeakFinder.cpp,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include "PeakFinder.h"
PeakFinder::PeakFinder()
{
}
// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
// to direction defined by 'direction' until next 'hump' after minimum value will
// begin
int PeakFinder::findGround(const float *data, int peakpos, int direction) const
{
float refvalue;
int lowpos;
int pos;
int climb_count;
float delta;
climb_count = 0;
refvalue = data[peakpos];
lowpos = peakpos;
pos = peakpos;
while ((pos > minPos) && (pos < maxPos))
{
int prevpos;
prevpos = pos;
pos += direction;
// calculate derivate
delta = data[pos] - data[prevpos];
if (delta <= 0)
{
// going downhill, ok
if (climb_count)
{
climb_count --; // decrease climb count
}
// check if new minimum found
if (data[pos] < refvalue)
{
// new minimum found
lowpos = pos;
refvalue = data[pos];
}
}
else
{
// going uphill, increase climbing counter
climb_count ++;
if (climb_count > 5) break; // we've been climbing too long => it's next uphill => quit
}
}
return lowpos;
}
// Find offset where the value crosses the given level, when starting from 'peakpos' and
// proceeds to direction defined in 'direction'
int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
{
float peaklevel;
int pos;
peaklevel = data[peakpos];
assert(peaklevel >= level);
pos = peakpos;
while ((pos >= minPos) && (pos < maxPos))
{
if (data[pos + direction] < level) return pos; // crossing found
pos += direction;
}
return -1; // not found
}
// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
float PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
{
int i;
float sum;
float wsum;
sum = 0;
wsum = 0;
for (i = firstPos; i <= lastPos; i ++)
{
sum += (float)i * data[i];
wsum += data[i];
}
return sum / wsum;
}
float PeakFinder::detectPeak(const float *data, int minPos, int maxPos)
{
#define max(x, y) (((x) > (y)) ? (x) : (y))
int i;
int peakpos; // position of peak level
float peakLevel; // peak level
int crosspos1, crosspos2; // position where the peak 'hump' crosses cutting level
float cutLevel; // cutting value
float groundLevel; // ground level of the peak
int gp1, gp2; // bottom positions of the peak 'hump'
this->minPos = minPos;
this->maxPos = maxPos;
// find absolute peak
peakpos = minPos;
peakLevel = data[minPos];
for (i = minPos + 1; i < maxPos; i ++)
{
if (data[i] > peakLevel)
{
peakLevel = data[i];
peakpos = i;
}
}
// find ground positions.
gp1 = findGround(data, peakpos, -1);
gp2 = findGround(data, peakpos, 1);
groundLevel = max(data[gp1], data[gp2]);
if (groundLevel < 1e-6) return 0; // ground level too small => detection failed
if ((peakLevel / groundLevel) < 1.3) return 0; // peak less than 30% of the ground level => no good peak detected
// calculate 70%-level of the peak
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
// find mid-level crossings
crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
if ((crosspos1 < 0) || (crosspos2 < 0)) return 0; // no crossing, no peak..
// calculate mass center of the peak surroundings
return calcMassCenter(data, crosspos1, crosspos2);
}
////////////////////////////////////////////////////////////////////////////////
///
/// Peak detection routine.
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: PeakFinder.cpp,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include "PeakFinder.h"
PeakFinder::PeakFinder()
{
}
// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
// to direction defined by 'direction' until next 'hump' after minimum value will
// begin
int PeakFinder::findGround(const float *data, int peakpos, int direction) const
{
float refvalue;
int lowpos;
int pos;
int climb_count;
float delta;
climb_count = 0;
refvalue = data[peakpos];
lowpos = peakpos;
pos = peakpos;
while ((pos > minPos) && (pos < maxPos))
{
int prevpos;
prevpos = pos;
pos += direction;
// calculate derivate
delta = data[pos] - data[prevpos];
if (delta <= 0)
{
// going downhill, ok
if (climb_count)
{
climb_count --; // decrease climb count
}
// check if new minimum found
if (data[pos] < refvalue)
{
// new minimum found
lowpos = pos;
refvalue = data[pos];
}
}
else
{
// going uphill, increase climbing counter
climb_count ++;
if (climb_count > 5) break; // we've been climbing too long => it's next uphill => quit
}
}
return lowpos;
}
// Find offset where the value crosses the given level, when starting from 'peakpos' and
// proceeds to direction defined in 'direction'
int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
{
float peaklevel;
int pos;
peaklevel = data[peakpos];
assert(peaklevel >= level);
pos = peakpos;
while ((pos >= minPos) && (pos < maxPos))
{
if (data[pos + direction] < level) return pos; // crossing found
pos += direction;
}
return -1; // not found
}
// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
float PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
{
int i;
float sum;
float wsum;
sum = 0;
wsum = 0;
for (i = firstPos; i <= lastPos; i ++)
{
sum += (float)i * data[i];
wsum += data[i];
}
return sum / wsum;
}
float PeakFinder::detectPeak(const float *data, int minPos, int maxPos)
{
#define max(x, y) (((x) > (y)) ? (x) : (y))
int i;
int peakpos; // position of peak level
float peakLevel; // peak level
int crosspos1, crosspos2; // position where the peak 'hump' crosses cutting level
float cutLevel; // cutting value
float groundLevel; // ground level of the peak
int gp1, gp2; // bottom positions of the peak 'hump'
this->minPos = minPos;
this->maxPos = maxPos;
// find absolute peak
peakpos = minPos;
peakLevel = data[minPos];
for (i = minPos + 1; i < maxPos; i ++)
{
if (data[i] > peakLevel)
{
peakLevel = data[i];
peakpos = i;
}
}
// find ground positions.
gp1 = findGround(data, peakpos, -1);
gp2 = findGround(data, peakpos, 1);
groundLevel = max(data[gp1], data[gp2]);
if (groundLevel < 1e-6) return 0; // ground level too small => detection failed
if ((peakLevel / groundLevel) < 1.3) return 0; // peak less than 30% of the ground level => no good peak detected
// calculate 70%-level of the peak
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
// find mid-level crossings
crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
if ((crosspos1 < 0) || (crosspos2 < 0)) return 0; // no crossing, no peak..
// calculate mass center of the peak surroundings
return calcMassCenter(data, crosspos1, crosspos2);
}

170
lib-src/soundtouch/source/example/bpm/PeakFinder.h
View File

@@ -1,85 +1,85 @@
////////////////////////////////////////////////////////////////////////////////
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:49 $
// File revision : $Revision: 1.2 $
//
// $Id: PeakFinder.h,v 1.2 2006-09-18 07:31:49 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _PeakFinder_H_
#define _PeakFinder_H_
class PeakFinder
{
protected:
/// Min, max allowed peak positions within the data vector
int minPos, maxPos;
/// Calculates the mass center between given vector items.
float calcMassCenter(const float *data, ///< Data vector.
int firstPos, ///< Index of first vector item beloging to the peak.
int lastPos ///< Index of last vector item beloging to the peak.
) const;
/// Finds the data vector index where the monotoniously decreasing signal crosses the
/// given level.
int findCrossingLevel(const float *data, ///< Data vector.
float level, ///< Goal crossing level.
int peakpos, ///< Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
/// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right-
/// or left-hand side of the given peak position.
int findGround(const float *data, /// Data vector.
int peakpos, /// Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
public:
/// Constructor.
PeakFinder();
/// Detect exact peak position of the data vector by finding the largest peak 'hump'
/// and calculating the mass-center location of the peak hump.
///
/// \return The exact mass-center location of the largest peak hump.
float detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
/// to be at least 'maxPos' items long.
int minPos, ///< Min allowed peak location within the vector data.
int maxPos ///< Max allowed peak location within the vector data.
);
};
#endif // _PeakFinder_H_
////////////////////////////////////////////////////////////////////////////////
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 07:31:49 $
// File revision : $Revision: 1.2 $
//
// $Id: PeakFinder.h,v 1.2 2006-09-18 07:31:49 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _PeakFinder_H_
#define _PeakFinder_H_
class PeakFinder
{
protected:
/// Min, max allowed peak positions within the data vector
int minPos, maxPos;
/// Calculates the mass center between given vector items.
float calcMassCenter(const float *data, ///< Data vector.
int firstPos, ///< Index of first vector item beloging to the peak.
int lastPos ///< Index of last vector item beloging to the peak.
) const;
/// Finds the data vector index where the monotoniously decreasing signal crosses the
/// given level.
int findCrossingLevel(const float *data, ///< Data vector.
float level, ///< Goal crossing level.
int peakpos, ///< Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
/// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right-
/// or left-hand side of the given peak position.
int findGround(const float *data, /// Data vector.
int peakpos, /// Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
public:
/// Constructor.
PeakFinder();
/// Detect exact peak position of the data vector by finding the largest peak 'hump'
/// and calculating the mass-center location of the peak hump.
///
/// \return The exact mass-center location of the largest peak hump.
float detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
/// to be at least 'maxPos' items long.
int minPos, ///< Min allowed peak location within the vector data.
int maxPos ///< Max allowed peak location within the vector data.
);
};
#endif // _PeakFinder_H_