diff options
| author |  waker <wakeroid@gmail.com> | 2011-01-02 12:04:10 +0100 |
|---|---|---|
| committer | waker <wakeroid@gmail.com> | 2011-01-02 12:04:10 +0100 |
| commit | be0631a8bbf54616547a54121efa1d7c7cf606bb (patch) | |
| tree | cc12063a9c2c50642c842804ac43c6da9e630255 /plugins | |
| parent | e8ed8a04ba8a31ad7734524dba40b02aabcd67eb (diff) | |
added missing soundtouch files
Diffstat (limited to 'plugins')
6 files changed, 844 insertions, 0 deletions
diff --git a/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/AAFilter.h b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/AAFilter.h new file mode 100644 index 00000000..d5c8ce4c --- /dev/null +++ b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/AAFilter.h @@ -0,0 +1,91 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
+/// while maintaining the original pitch by using a time domain WSOLA-like method
+/// with several performance-increasing tweaks.
+///
+/// Anti-alias filter is used to prevent folding of high frequencies when
+/// transposing the sample rate with interpolation.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
+// File revision : $Revision: 4 $
+//
+// $Id: AAFilter.h 11 2008-02-10 16:26:55Z 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 AAFilter_H
+#define AAFilter_H
+
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class AAFilter
+{
+protected:
+ class FIRFilter *pFIR;
+
+ /// Low-pass filter cut-off frequency, negative = invalid
+ double cutoffFreq;
+
+ /// num of filter taps
+ uint length;
+
+ /// Calculate the FIR coefficients realizing the given cutoff-frequency
+ void calculateCoeffs();
+public:
+ AAFilter(uint length);
+
+ ~AAFilter();
+
+ /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
+ /// frequency (nyquist frequency = 0.5). The filter will cut off the
+ /// frequencies than that.
+ void setCutoffFreq(double newCutoffFreq);
+
+ /// Sets number of FIR filter taps, i.e. ~filter complexity
+ void setLength(uint newLength);
+
+ uint getLength() const;
+
+ /// Applies the filter to the given sequence of samples.
+ /// Note : The amount of outputted samples is by value of 'filter length'
+ /// smaller than the amount of input samples.
+ uint evaluate(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples,
+ uint numChannels) const;
+};
+
+}
+
+#endif
diff --git a/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/FIRFilter.h b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/FIRFilter.h new file mode 100644 index 00000000..5713f7bb --- /dev/null +++ b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/FIRFilter.h @@ -0,0 +1,164 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// General FIR digital filter routines with MMX optimization.
+///
+/// Note : MMX optimized functions reside in a separate, platform-specific file,
+/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
+///
+/// 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: FIRFilter.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 FIRFilter_H
+#define FIRFilter_H
+
+#include <stddef.h>
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class FIRFilter
+{
+protected:
+ // Number of FIR filter taps
+ uint length;
+ // Number of FIR filter taps divided by 8
+ uint lengthDiv8;
+
+ // Result divider factor in 2^k format
+ uint resultDivFactor;
+
+ // Result divider value.
+ SAMPLETYPE resultDivider;
+
+ // Memory for filter coefficients
+ SAMPLETYPE *filterCoeffs;
+
+ virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples) const;
+ virtual uint evaluateFilterMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples) const;
+
+public:
+ FIRFilter();
+ virtual ~FIRFilter();
+
+ /// Operator 'new' is overloaded so that it automatically creates a suitable instance
+ /// depending on if we've a MMX-capable CPU available or not.
+ static void * operator new(size_t s);
+
+ static FIRFilter *newInstance();
+
+ /// Applies the filter to the given sequence of samples.
+ /// Note : The amount of outputted samples is by value of 'filter_length'
+ /// smaller than the amount of input samples.
+ ///
+ /// \return Number of samples copied to 'dest'.
+ uint evaluate(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples,
+ uint numChannels) const;
+
+ uint getLength() const;
+
+ virtual void setCoefficients(const SAMPLETYPE *coeffs,
+ uint newLength,
+ uint uResultDivFactor);
+};
+
+
+// Optional subclasses that implement CPU-specific optimizations:
+
+#ifdef ALLOW_MMX
+
+/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
+ class FIRFilterMMX : public FIRFilter
+ {
+ protected:
+ short *filterCoeffsUnalign;
+ short *filterCoeffsAlign;
+
+ virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;
+ public:
+ FIRFilterMMX();
+ ~FIRFilterMMX();
+
+ virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
+ };
+
+#endif // ALLOW_MMX
+
+
+#ifdef ALLOW_3DNOW
+
+ /// Class that implements 3DNow! optimized functions exclusive for floating point samples type.
+ class FIRFilter3DNow : public FIRFilter
+ {
+ protected:
+ float *filterCoeffsUnalign;
+ float *filterCoeffsAlign;
+
+ virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
+ public:
+ FIRFilter3DNow();
+ ~FIRFilter3DNow();
+ virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
+ };
+
+#endif // ALLOW_3DNOW
+
+
+#ifdef ALLOW_SSE
+ /// Class that implements SSE optimized functions exclusive for floating point samples type.
+ class FIRFilterSSE : public FIRFilter
+ {
+ protected:
+ float *filterCoeffsUnalign;
+ float *filterCoeffsAlign;
+
+ virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
+ public:
+ FIRFilterSSE();
+ ~FIRFilterSSE();
+
+ virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
+ };
+
+#endif // ALLOW_SSE
+
+}
+
+#endif // FIRFilter_H
diff --git a/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/PeakFinder.h b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/PeakFinder.h new file mode 100644 index 00000000..e3640cc6 --- /dev/null +++ b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/PeakFinder.h @@ -0,0 +1,93 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// The routine detects highest value on an array of values and calculates the
+/// precise peak location as a mass-center of the 'hump' around the peak value.
+///
+/// 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: PeakFinder.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 _PeakFinder_H_
+#define _PeakFinder_H_
+
+namespace soundtouch
+{
+
+class PeakFinder
+{
+protected:
+ /// Min, max allowed peak positions within the data vector
+ int minPos, maxPos;
+
+ /// Calculates the mass center between given vector items.
+ double calcMassCenter(const float *data, ///< Data vector.
+ int firstPos, ///< Index of first vector item beloging to the peak.
+ int lastPos ///< Index of last vector item beloging to the peak.
+ ) const;
+
+ /// Finds the data vector index where the monotoniously decreasing signal crosses the
+ /// given level.
+ int findCrossingLevel(const float *data, ///< Data vector.
+ float level, ///< Goal crossing level.
+ int peakpos, ///< Peak position index within the data vector.
+ int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
+ ) const;
+
+ /// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right-
+ /// or left-hand side of the given peak position.
+ int findGround(const float *data, /// Data vector.
+ int peakpos, /// Peak position index within the data vector.
+ int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
+ ) const;
+
+ /// get exact center of peak near given position by calculating local mass of center
+ double getPeakCenter(const float *data, int peakpos) const;
+
+public:
+ /// Constructor.
+ PeakFinder();
+
+ /// Detect exact peak position of the data vector by finding the largest peak 'hump'
+ /// and calculating the mass-center location of the peak hump.
+ ///
+ /// \return The location of the largest base harmonic peak hump.
+ double detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
+ /// to be at least 'maxPos' items long.
+ int minPos, ///< Min allowed peak location within the vector data.
+ int maxPos ///< Max allowed peak location within the vector data.
+ );
+};
+
+}
+
+#endif // _PeakFinder_H_
diff --git a/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/RateTransposer.h b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/RateTransposer.h new file mode 100644 index 00000000..f035af2c --- /dev/null +++ b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/RateTransposer.h @@ -0,0 +1,159 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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;
+
+ BOOL 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(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
diff --git a/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/TDStretch.h b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/TDStretch.h new file mode 100644 index 00000000..00d1f3e3 --- /dev/null +++ b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/TDStretch.h @@ -0,0 +1,275 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
+/// while maintaining the original pitch by using a time domain WSOLA-like method
+/// with several performance-increasing tweaks.
+///
+/// Note : MMX/SSE optimized functions reside in separate, platform-specific files
+/// 'mmx_optimized.cpp' and 'sse_optimized.cpp'
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-05-17 14:35:13 +0300 (Sun, 17 May 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: TDStretch.h 71 2009-05-17 11:35:13Z 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 TDStretch_H
+#define TDStretch_H
+
+#include <stddef.h>
+#include "STTypes.h"
+#include "RateTransposer.h"
+#include "FIFOSamplePipe.h"
+
+namespace soundtouch
+{
+
+/// Default values for sound processing parameters:
+/// Notice that the default parameters are tuned for contemporary popular music
+/// processing. For speech processing applications these parameters suit better:
+/// #define DEFAULT_SEQUENCE_MS 40
+/// #define DEFAULT_SEEKWINDOW_MS 15
+/// #define DEFAULT_OVERLAP_MS 8
+///
+
+/// Default length of a single processing sequence, in milliseconds. This determines to how
+/// long sequences the original sound is chopped in the time-stretch algorithm.
+///
+/// The larger this value is, the lesser sequences are used in processing. In principle
+/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo
+/// and vice versa.
+///
+/// Increasing this value reduces computational burden & vice versa.
+//#define DEFAULT_SEQUENCE_MS 40
+#define DEFAULT_SEQUENCE_MS USE_AUTO_SEQUENCE_LEN
+
+/// Giving this value for the sequence length sets automatic parameter value
+/// according to tempo setting (recommended)
+#define USE_AUTO_SEQUENCE_LEN 0
+
+/// Seeking window default length in milliseconds for algorithm that finds the best possible
+/// overlapping location. This determines from how wide window the algorithm may look for an
+/// optimal joining location when mixing the sound sequences back together.
+///
+/// The bigger this window setting is, the higher the possibility to find a better mixing
+/// position will become, but at the same time large values may cause a "drifting" artifact
+/// because consequent sequences will be taken at more uneven intervals.
+///
+/// If there's a disturbing artifact that sounds as if a constant frequency was drifting
+/// around, try reducing this setting.
+///
+/// Increasing this value increases computational burden & vice versa.
+//#define DEFAULT_SEEKWINDOW_MS 15
+#define DEFAULT_SEEKWINDOW_MS USE_AUTO_SEEKWINDOW_LEN
+
+/// Giving this value for the seek window length sets automatic parameter value
+/// according to tempo setting (recommended)
+#define USE_AUTO_SEEKWINDOW_LEN 0
+
+/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together,
+/// to form a continuous sound stream, this parameter defines over how long period the two
+/// consecutive sequences are let to overlap each other.
+///
+/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting
+/// by a large amount, you might wish to try a smaller value on this.
+///
+/// Increasing this value increases computational burden & vice versa.
+#define DEFAULT_OVERLAP_MS 8
+
+
+/// Class that does the time-stretch (tempo change) effect for the processed
+/// sound.
+class TDStretch : public FIFOProcessor
+{
+protected:
+ int channels;
+ int sampleReq;
+ float tempo;
+
+ SAMPLETYPE *pMidBuffer;
+ SAMPLETYPE *pRefMidBuffer;
+ SAMPLETYPE *pRefMidBufferUnaligned;
+ int overlapLength;
+ int seekLength;
+ int seekWindowLength;
+ int overlapDividerBits;
+ int slopingDivider;
+ float nominalSkip;
+ float skipFract;
+ FIFOSampleBuffer outputBuffer;
+ FIFOSampleBuffer inputBuffer;
+ BOOL bQuickSeek;
+// int outDebt;
+// BOOL bMidBufferDirty;
+
+ int sampleRate;
+ int sequenceMs;
+ int seekWindowMs;
+ int overlapMs;
+ BOOL bAutoSeqSetting;
+ BOOL bAutoSeekSetting;
+
+ void acceptNewOverlapLength(int newOverlapLength);
+
+ virtual void clearCrossCorrState();
+ void calculateOverlapLength(int overlapMs);
+
+ virtual LONG_SAMPLETYPE calcCrossCorrStereo(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+ virtual LONG_SAMPLETYPE calcCrossCorrMono(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+
+ virtual int seekBestOverlapPositionStereo(const SAMPLETYPE *refPos);
+ virtual int seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos);
+ virtual int seekBestOverlapPositionMono(const SAMPLETYPE *refPos);
+ virtual int seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos);
+ int seekBestOverlapPosition(const SAMPLETYPE *refPos);
+
+ virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+ virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+
+ void clearMidBuffer();
+ void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
+
+ void precalcCorrReferenceMono();
+ void precalcCorrReferenceStereo();
+
+ void calcSeqParameters();
+
+ /// Changes the tempo of the given sound samples.
+ /// Returns amount of samples returned in the "output" buffer.
+ /// The maximum amount of samples that can be returned at a time is set by
+ /// the 'set_returnBuffer_size' function.
+ void processSamples();
+
+public:
+ TDStretch();
+ virtual ~TDStretch();
+
+ /// Operator 'new' is overloaded so that it automatically creates a suitable instance
+ /// depending on if we've a MMX/SSE/etc-capable CPU available or not.
+ 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 feature set depending on if the CPU
+ /// supports MMX/SSE/etc extensions.
+ static TDStretch *newInstance();
+
+ /// Returns the output buffer object
+ FIFOSamplePipe *getOutput() { return &outputBuffer; };
+
+ /// Returns the input buffer object
+ FIFOSamplePipe *getInput() { return &inputBuffer; };
+
+ /// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
+ /// tempo, larger faster tempo.
+ void setTempo(float newTempo);
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ virtual void clear();
+
+ /// Clears the input buffer
+ void clearInput();
+
+ /// Sets the number of channels, 1 = mono, 2 = stereo
+ void setChannels(int numChannels);
+
+ /// Enables/disables the quick position seeking algorithm. Zero to disable,
+ /// nonzero to enable
+ void enableQuickSeek(BOOL enable);
+
+ /// Returns nonzero if the quick seeking algorithm is enabled.
+ BOOL isQuickSeekEnabled() const;
+
+ /// Sets routine control parameters. These control are certain time constants
+ /// defining how the sound is stretched to the desired duration.
+ //
+ /// 'sampleRate' = sample rate of the sound
+ /// 'sequenceMS' = one processing sequence length in milliseconds
+ /// 'seekwindowMS' = seeking window length for scanning the best overlapping
+ /// position
+ /// 'overlapMS' = overlapping length
+ void setParameters(int sampleRate, ///< Samplerate of sound being processed (Hz)
+ int sequenceMS = -1, ///< Single processing sequence length (ms)
+ int seekwindowMS = -1, ///< Offset seeking window length (ms)
+ int overlapMS = -1 ///< Sequence overlapping length (ms)
+ );
+
+ /// Get routine control parameters, see setParameters() function.
+ /// Any of the parameters to this function can be NULL, in such case corresponding parameter
+ /// value isn't returned.
+ void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
+
+ /// Adds 'numsamples' pcs of samples from the 'samples' memory position into
+ /// the input of the object.
+ virtual void putSamples(
+ const SAMPLETYPE *samples, ///< Input sample data
+ uint numSamples ///< Number of samples in 'samples' so that one sample
+ ///< contains both channels if stereo
+ );
+};
+
+
+
+// Implementation-specific class declarations:
+
+#ifdef ALLOW_MMX
+ /// Class that implements MMX optimized routines for 16bit integer samples type.
+ class TDStretchMMX : public TDStretch
+ {
+ protected:
+ long calcCrossCorrStereo(const short *mixingPos, const short *compare) const;
+ virtual void overlapStereo(short *output, const short *input) const;
+ virtual void clearCrossCorrState();
+ };
+#endif /// ALLOW_MMX
+
+
+#ifdef ALLOW_3DNOW
+ /// Class that implements 3DNow! optimized routines for floating point samples type.
+ class TDStretch3DNow : public TDStretch
+ {
+ protected:
+ double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
+ };
+#endif /// ALLOW_3DNOW
+
+
+#ifdef ALLOW_SSE
+ /// Class that implements SSE optimized routines for floating point samples type.
+ class TDStretchSSE : public TDStretch
+ {
+ protected:
+ double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
+ };
+
+#endif /// ALLOW_SSE
+
+}
+#endif /// TDStretch_H
diff --git a/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/cpu_detect.h b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/cpu_detect.h new file mode 100644 index 00000000..025781da --- /dev/null +++ b/plugins/dsp_soundtouch/soundtouch/source/SoundTouch/cpu_detect.h @@ -0,0 +1,62 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// A header file for detecting the Intel MMX instructions set extension.
+///
+/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the
+/// routine implementations for x86 Windows, x86 gnu version and non-x86
+/// platforms, respectively.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
+// File revision : $Revision: 4 $
+//
+// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z 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 _CPU_DETECT_H_
+#define _CPU_DETECT_H_
+
+#include "STTypes.h"
+
+#define SUPPORT_MMX 0x0001
+#define SUPPORT_3DNOW 0x0002
+#define SUPPORT_ALTIVEC 0x0004
+#define SUPPORT_SSE 0x0008
+#define SUPPORT_SSE2 0x0010
+
+/// Checks which instruction set extensions are supported by the CPU.
+///
+/// \return A bitmask of supported extensions, see SUPPORT_... defines.
+uint detectCPUextensions(void);
+
+/// Disables given set of instruction extensions. See SUPPORT_... defines.
+void disableExtensions(uint wDisableMask);
+
+#endif // _CPU_DETECT_H_
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