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50 lines
1.4 KiB
C
50 lines
1.4 KiB
C
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// Polyphase decimation filter.
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//
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// Convert an oversampled audio stream to non-oversampled. Uses a
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// windowed sinc FIR filter w/ Blackman window to control aliasing.
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// Christian Floisand's 'blog explains it very well.
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//
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// This version has a very simple main processing loop (the decimate
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// method) which vectorizes easily.
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//
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// Refs:
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// https://christianfloisand.wordpress.com/2012/12/05/audio-resampling-part-1/
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// https://christianfloisand.wordpress.com/2013/01/28/audio-resampling-part-2/
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// http://www.dspguide.com/ch16.htm
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// http://en.wikipedia.org/wiki/Window_function#Blackman_windows
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#ifndef INCLUDE_M17MODDECIMATOR_H
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#define INCLUDE_M17MODDECIMATOR_H
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#include <cstdint>
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class M17ModDecimator {
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public:
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M17ModDecimator();
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~M17ModDecimator();
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void initialize(
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double decimatedSampleRate,
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double passFrequency,
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unsigned oversampleRatio
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);
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double oversampleRate() const { return mOversampleRate; }
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int oversampleRatio() const { return mRatio; }
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void decimate(int16_t *in, int16_t *out, unsigned int outCount);
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// N.B., input must have (ratio * outCount) samples.
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private:
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double mDecimatedSampleRate;
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double mOversampleRate;
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int mRatio; // oversample ratio
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float *mKernel;
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unsigned int mKernelSize;
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float *mShift; // shift register
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unsigned int mCursor;
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};
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#endif // INCLUDE_M17MODDECIMATOR_H
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