////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.10 $
//
// $Id: RateTransposer.h,v 1.10 2006/02/05 16:44:06 Olli 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 RateTransposer_H
#define RateTransposer_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;

	uint32_t uChannels;

	/// 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;

	void init();

	virtual void resetRegisters() = 0;

	virtual uint32_t transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples) = 0;
	virtual uint32_t transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples) = 0;
	uint32_t transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples);

	void flushStoreBuffer();

	void downsample(const SAMPLETYPE *src, uint32_t numSamples);
	void upsample(const SAMPLETYPE *src, uint32_t 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, uint32_t 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.
	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() const;

	/// 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(uint32_t channels);

	/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
	/// the input of the object.
	void putSamples(const SAMPLETYPE *samples, uint32_t numSamples);

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

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

}

#endif