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Diffstat (limited to 'plugins/dsp_soundtouch/soundtouch/source/SoundTouch/RateTransposer.h')
| -rw-r--r-- | plugins/dsp_soundtouch/soundtouch/source/SoundTouch/RateTransposer.h | 159 |
1 files changed, 159 insertions, 0 deletions
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
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