3rdparty/SoundTouch/RateTransposer.h

////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application).
///
/// Use either of the derived classes of 'RateTransposerInteger' or
/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
/// algorithm implementation.
///
/// Author        : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed  : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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 RateTransposer_H
#define RateTransposer_H

#include <stddef.h>
#include "AAFilter.h"
#include "FIFOSamplePipe.h"
#include "FIFOSampleBuffer.h"

#include "STTypes.h"

namespace soundtouch
{

/// A common linear samplerate transposer class.
///
/// Note: Use function "RateTransposer::newInstance()" to create a new class
/// instance instead of the "new" operator; that function automatically
/// chooses a correct implementation depending on if integer or floating
/// arithmetics are to be used.
class RateTransposer : public FIFOProcessor
{
protected:
    /// Anti-alias filter object
    AAFilter *pAAFilter;

    float fRate;

    int numChannels;

    /// Buffer for collecting samples to feed the anti-alias filter between
    /// two batches
    FIFOSampleBuffer storeBuffer;

    /// Buffer for keeping samples between transposing & anti-alias filter
    FIFOSampleBuffer tempBuffer;

    /// Output sample buffer
    FIFOSampleBuffer outputBuffer;

    BOOL bUseAAFilter;

    virtual void resetRegisters() = 0;

    virtual uint transposeStereo(SAMPLETYPE *dest,
                         const SAMPLETYPE *src,
                         uint numSamples) = 0;
    virtual uint transposeMono(SAMPLETYPE *dest,
                       const SAMPLETYPE *src,
                       uint numSamples) = 0;
    inline uint transpose(SAMPLETYPE *dest,
                   const SAMPLETYPE *src,
                   uint numSamples);

    void downsample(const SAMPLETYPE *src,
                    uint numSamples);
    void upsample(const SAMPLETYPE *src,
                 uint numSamples);

    /// Transposes sample rate by applying anti-alias filter to prevent folding.
    /// Returns amount of samples returned in the "dest" buffer.
    /// The maximum amount of samples that can be returned at a time is set by
    /// the 'set_returnBuffer_size' function.
    void processSamples(const SAMPLETYPE *src,
                        uint numSamples);


public:
    RateTransposer();
    virtual ~RateTransposer();

    /// Operator 'new' is overloaded so that it automatically creates a suitable instance
    /// depending on if we're to use integer or floating point arithmetics.
    static void *operator new(size_t s);

    /// Use this function instead of "new" operator to create a new instance of this class.
    /// This function automatically chooses a correct implementation, depending on if
    /// integer ot floating point arithmetics are to be used.
    static RateTransposer *newInstance();

    /// Returns the output buffer object
    FIFOSamplePipe *getOutput() { return &outputBuffer; };

    /// Returns the store buffer object
    FIFOSamplePipe *getStore() { return &storeBuffer; };

    /// Return anti-alias filter object
    AAFilter *getAAFilter();

    /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
    void enableAAFilter(BOOL newMode);

    /// Returns nonzero if anti-alias filter is enabled.
    BOOL isAAFilterEnabled() const;

    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower
    /// rate, larger faster rates.
    virtual void setRate(float newRate);

    /// Sets the number of channels, 1 = mono, 2 = stereo
    void setChannels(int channels);

    /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
    /// the input of the object.
    void putSamples(const SAMPLETYPE *samples, uint numSamples);

    /// Clears all the samples in the object
    void clear();

    /// Returns nonzero if there aren't any samples available for outputting.
    int isEmpty() const;
};

}

#endif
1	////////////////////////////////////////////////////////////////////////////////
2	///
3	/// Sample rate transposer. Changes sample rate by using linear interpolation
4	/// together with anti-alias filtering (first order interpolation with anti-
5	/// alias filtering should be quite adequate for this application).
6	///
7	/// Use either of the derived classes of 'RateTransposerInteger' or
8	/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
9	/// algorithm implementation.
10	///
11	/// Author : Copyright (c) Olli Parviainen
12	/// Author e-mail : oparviai 'at' iki.fi
13	/// SoundTouch WWW: http://www.surina.net/soundtouch
14	///
15	////////////////////////////////////////////////////////////////////////////////
16	//
17	// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
18	// File revision : $Revision: 4 $
19	//
20	// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $
21	//
22	////////////////////////////////////////////////////////////////////////////////
23	//
24	// License :
25	//
26	// SoundTouch audio processing library
27	// Copyright (c) Olli Parviainen
28	//
29	// This library is free software; you can redistribute it and/or
30	// modify it under the terms of the GNU Lesser General Public
31	// License as published by the Free Software Foundation; either
32	// version 2.1 of the License, or (at your option) any later version.
33	//
34	// This library is distributed in the hope that it will be useful,
35	// but WITHOUT ANY WARRANTY; without even the implied warranty of
36	// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
37	// Lesser General Public License for more details.
38	//
39	// You should have received a copy of the GNU Lesser General Public
40	// License along with this library; if not, write to the Free Software
41	// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
42	//
43	////////////////////////////////////////////////////////////////////////////////
44
45	#ifndef RateTransposer_H
46	#define RateTransposer_H
47
48	#include <stddef.h>
49	#include "AAFilter.h"
50	#include "FIFOSamplePipe.h"
51	#include "FIFOSampleBuffer.h"
52
53	#include "STTypes.h"
54
55	namespace soundtouch
56	{
57
58	/// A common linear samplerate transposer class.
59	///
60	/// Note: Use function "RateTransposer::newInstance()" to create a new class
61	/// instance instead of the "new" operator; that function automatically
62	/// chooses a correct implementation depending on if integer or floating
63	/// arithmetics are to be used.
64	class RateTransposer : public FIFOProcessor
65	{
66	protected:
67	/// Anti-alias filter object
68	AAFilter *pAAFilter;
69
70	float fRate;
71
72	int numChannels;
73
74	/// Buffer for collecting samples to feed the anti-alias filter between
75	/// two batches
76	FIFOSampleBuffer storeBuffer;
77
78	/// Buffer for keeping samples between transposing & anti-alias filter
79	FIFOSampleBuffer tempBuffer;
80
81	/// Output sample buffer
82	FIFOSampleBuffer outputBuffer;
83
84	BOOL bUseAAFilter;
85
86	virtual void resetRegisters() = 0;
87
88	virtual uint transposeStereo(SAMPLETYPE *dest,
89	const SAMPLETYPE *src,
90	uint numSamples) = 0;
91	virtual uint transposeMono(SAMPLETYPE *dest,
92	const SAMPLETYPE *src,
93	uint numSamples) = 0;
94	inline uint transpose(SAMPLETYPE *dest,
95	const SAMPLETYPE *src,
96	uint numSamples);
97
98	void downsample(const SAMPLETYPE *src,
99	uint numSamples);
100	void upsample(const SAMPLETYPE *src,
101	uint numSamples);
102
103	/// Transposes sample rate by applying anti-alias filter to prevent folding.
104	/// Returns amount of samples returned in the "dest" buffer.
105	/// The maximum amount of samples that can be returned at a time is set by
106	/// the 'set_returnBuffer_size' function.
107	void processSamples(const SAMPLETYPE *src,
108	uint numSamples);
109
110
111	public:
112	RateTransposer();
113	virtual ~RateTransposer();
114
115	/// Operator 'new' is overloaded so that it automatically creates a suitable instance
116	/// depending on if we're to use integer or floating point arithmetics.
117	static void *operator new(size_t s);
118
119	/// Use this function instead of "new" operator to create a new instance of this class.
120	/// This function automatically chooses a correct implementation, depending on if
121	/// integer ot floating point arithmetics are to be used.
122	static RateTransposer *newInstance();
123
124	/// Returns the output buffer object
125	FIFOSamplePipe *getOutput() { return &outputBuffer; };
126
127	/// Returns the store buffer object
128	FIFOSamplePipe *getStore() { return &storeBuffer; };
129
130	/// Return anti-alias filter object
131	AAFilter *getAAFilter();
132
133	/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
134	void enableAAFilter(BOOL newMode);
135
136	/// Returns nonzero if anti-alias filter is enabled.
137	BOOL isAAFilterEnabled() const;
138
139	/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
140	/// rate, larger faster rates.
141	virtual void setRate(float newRate);
142
143	/// Sets the number of channels, 1 = mono, 2 = stereo
144	void setChannels(int channels);
145
146	/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
147	/// the input of the object.
148	void putSamples(const SAMPLETYPE *samples, uint numSamples);
149
150	/// Clears all the samples in the object
151	void clear();
152
153	/// Returns nonzero if there aren't any samples available for outputting.
154	int isEmpty() const;
155	};
156
157	}
158
159	#endif