1 files changed, 0 insertions, 163 deletions
diff --git a/lib/soundtouch/RateTransposer.h b/lib/soundtouch/RateTransposer.h
deleted file mode 100644
index 52b7441b89..0000000000
--- a/lib/soundtouch/RateTransposer.h
+++ /dev/null
@@ -1,163 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////
-/// 
-/// 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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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:
-    double rate;
-    int numChannels;
-
-    TransposerBase();
-    virtual ~TransposerBase();
-
-    virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
-    virtual void setRate(double 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();
-
-    /// Returns the output buffer object
-    FIFOSamplePipe *getOutput() { return &outputBuffer; };
-
-    /// 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(double 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;
-
-    /// Return approximate initial input-output latency
-    int getLatency() const;
-};
-
-}
-
-#endif