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Diffstat (limited to 'media/libsoundtouch/src/RateTransposer.h')
| -rw-r--r-- | media/libsoundtouch/src/RateTransposer.h | 179 |
1 files changed, 179 insertions, 0 deletions
diff --git a/media/libsoundtouch/src/RateTransposer.h b/media/libsoundtouch/src/RateTransposer.h new file mode 100644 index 000000000..fad6d6571 --- /dev/null +++ b/media/libsoundtouch/src/RateTransposer.h @@ -0,0 +1,179 @@ +//////////////////////////////////////////////////////////////////////////////// +/// +/// 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: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $ +// File revision : $Revision: 4 $ +// +// $Id: RateTransposer.h 195 2014-04-06 15:57:21Z 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 +{ + +/// Abstract base class for transposer implementations (linear, advanced vs integer, float etc) +class TransposerBase +{ +public: + enum ALGORITHM { + LINEAR = 0, + CUBIC, + SHANNON + }; + +protected: + virtual void resetRegisters() = 0; + + virtual int transposeMono(SAMPLETYPE *dest, + const SAMPLETYPE *src, + int &srcSamples) = 0; + virtual int transposeStereo(SAMPLETYPE *dest, + const SAMPLETYPE *src, + int &srcSamples) = 0; + virtual int transposeMulti(SAMPLETYPE *dest, + const SAMPLETYPE *src, + int &srcSamples) = 0; + + static ALGORITHM algorithm; + +public: + float rate; + int numChannels; + + TransposerBase(); + virtual ~TransposerBase(); + + virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src); + virtual void setRate(float newRate); + virtual void setChannels(int channels); + + // static factory function + static TransposerBase *newInstance(); + + // static function to set interpolation algorithm + static void setAlgorithm(ALGORITHM a); +}; + + +/// A common linear samplerate transposer class. +/// +class RateTransposer : public FIFOProcessor +{ +protected: + /// Anti-alias filter object + AAFilter *pAAFilter; + TransposerBase *pTransposer; + + /// Buffer for collecting samples to feed the anti-alias filter between + /// two batches + FIFOSampleBuffer inputBuffer; + + /// Buffer for keeping samples between transposing & anti-alias filter + FIFOSampleBuffer midBuffer; + + /// Output sample buffer + FIFOSampleBuffer outputBuffer; + + bool bUseAAFilter; + + + /// 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 |