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- ////////////////////////////////////////////////////////////////////////////////
- ///
- /// 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;
- SBOOL 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(SBOOL newMode);
- /// Returns nonzero if anti-alias filter is enabled.
- SBOOL 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
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