fa7c141813
git-svn-id: svn://localhost/trunk/ardour2@13 d708f5d6-7413-0410-9779-e7cbd77b26cf
159 lines
5.1 KiB
C++
159 lines
5.1 KiB
C++
////////////////////////////////////////////////////////////////////////////////
|
|
///
|
|
/// 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 @ iki.fi
|
|
/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
|
|
///
|
|
////////////////////////////////////////////////////////////////////////////////
|
|
//
|
|
// Last changed : $Date$
|
|
// File revision : $Revision$
|
|
//
|
|
// $Id$
|
|
//
|
|
////////////////////////////////////////////////////////////////////////////////
|
|
//
|
|
// 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;
|
|
|
|
uint 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 uint transposeStereo(SAMPLETYPE *dest,
|
|
const SAMPLETYPE *src,
|
|
uint numSamples) = 0;
|
|
virtual uint transposeMono(SAMPLETYPE *dest,
|
|
const SAMPLETYPE *src,
|
|
uint numSamples) = 0;
|
|
uint transpose(SAMPLETYPE *dest,
|
|
const SAMPLETYPE *src,
|
|
uint numSamples);
|
|
|
|
void flushStoreBuffer();
|
|
|
|
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);
|
|
|
|
RateTransposer();
|
|
|
|
public:
|
|
virtual ~RateTransposer();
|
|
|
|
/// 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(uint 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.
|
|
uint isEmpty();
|
|
};
|
|
|
|
}
|
|
|
|
#endif
|