Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 164 insertions, 0 deletions

diff --git a/media/libsoundtouch/src/RateTransposer.h b/media/libsoundtouch/src/RateTransposer.h
new file mode 100644
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+++ b/media/libsoundtouch/src/RateTransposer.h
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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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 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);
+    virtual int getLatency() const = 0;
+
+    virtual void resetRegisters() = 0;
+
+    // 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