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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /media/libsoundtouch/src/RateTransposer.h
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
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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