159 lines
5.1 KiB
C++
159 lines
5.1 KiB
C++
////////////////////////////////////////////////////////////////////////////////
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///
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/// Sample rate transposer. Changes sample rate by using linear interpolation
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/// together with anti-alias filtering (first order interpolation with anti-
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/// alias filtering should be quite adequate for this application).
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///
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/// Use either of the derived classes of 'RateTransposerInteger' or
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/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
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/// algorithm implementation.
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///
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/// Author : Copyright (c) Olli Parviainen
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/// Author e-mail : oparviai @ iki.fi
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/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
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///
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////////////////////////////////////////////////////////////////////////////////
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//
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// Last changed : $Date$
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// File revision : $Revision$
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//
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// $Id$
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//
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////////////////////////////////////////////////////////////////////////////////
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//
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// License :
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//
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// SoundTouch audio processing library
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// Copyright (c) Olli Parviainen
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//
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// This library is free software; you can redistribute it and/or
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// modify it under the terms of the GNU Lesser General Public
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// License as published by the Free Software Foundation; either
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// version 2.1 of the License, or (at your option) any later version.
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//
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// This library is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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// Lesser General Public License for more details.
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//
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// You should have received a copy of the GNU Lesser General Public
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// License along with this library; if not, write to the Free Software
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// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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//
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////////////////////////////////////////////////////////////////////////////////
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#ifndef RateTransposer_H
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#define RateTransposer_H
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#include "AAFilter.h"
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#include "FIFOSamplePipe.h"
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#include "FIFOSampleBuffer.h"
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#include "STTypes.h"
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namespace soundtouch
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{
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/// A common linear samplerate transposer class.
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///
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/// Note: Use function "RateTransposer::newInstance()" to create a new class
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/// instance instead of the "new" operator; that function automatically
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/// chooses a correct implementation depending on if integer or floating
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/// arithmetics are to be used.
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class RateTransposer : public FIFOProcessor
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{
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protected:
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/// Anti-alias filter object
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AAFilter *pAAFilter;
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float fRate;
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uint uChannels;
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/// Buffer for collecting samples to feed the anti-alias filter between
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/// two batches
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FIFOSampleBuffer storeBuffer;
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/// Buffer for keeping samples between transposing & anti-alias filter
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FIFOSampleBuffer tempBuffer;
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/// Output sample buffer
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FIFOSampleBuffer outputBuffer;
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BOOL bUseAAFilter;
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void init();
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virtual void resetRegisters() = 0;
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virtual uint transposeStereo(SAMPLETYPE *dest,
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const SAMPLETYPE *src,
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uint numSamples) = 0;
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virtual uint transposeMono(SAMPLETYPE *dest,
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const SAMPLETYPE *src,
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uint numSamples) = 0;
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uint transpose(SAMPLETYPE *dest,
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const SAMPLETYPE *src,
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uint numSamples);
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void flushStoreBuffer();
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void downsample(const SAMPLETYPE *src,
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uint numSamples);
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void upsample(const SAMPLETYPE *src,
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uint numSamples);
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/// Transposes sample rate by applying anti-alias filter to prevent folding.
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/// Returns amount of samples returned in the "dest" buffer.
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/// The maximum amount of samples that can be returned at a time is set by
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/// the 'set_returnBuffer_size' function.
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void processSamples(const SAMPLETYPE *src,
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uint numSamples);
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RateTransposer();
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public:
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virtual ~RateTransposer();
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/// Use this function instead of "new" operator to create a new instance of this class.
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/// This function automatically chooses a correct implementation, depending on if
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/// integer ot floating point arithmetics are to be used.
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static RateTransposer *newInstance();
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/// Returns the output buffer object
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FIFOSamplePipe *getOutput() { return &outputBuffer; };
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/// Returns the store buffer object
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FIFOSamplePipe *getStore() { return &storeBuffer; };
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/// Return anti-alias filter object
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AAFilter *getAAFilter() const;
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/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
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void enableAAFilter(BOOL newMode);
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/// Returns nonzero if anti-alias filter is enabled.
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BOOL isAAFilterEnabled() const;
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/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
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/// rate, larger faster rates.
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virtual void setRate(float newRate);
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/// Sets the number of channels, 1 = mono, 2 = stereo
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void setChannels(uint channels);
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/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
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/// the input of the object.
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void putSamples(const SAMPLETYPE *samples, uint numSamples);
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/// Clears all the samples in the object
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void clear();
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/// Returns nonzero if there aren't any samples available for outputting.
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uint isEmpty();
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};
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}
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#endif
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