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* LinuxSampler - modular, streaming capable sampler * |
* LinuxSampler - modular, streaming capable sampler * |
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* * |
* * |
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* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck * |
* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck * |
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* Copyright (C) 2005 - 2007 Christian Schoenebeck * |
* Copyright (C) 2005 - 2016 Christian Schoenebeck * |
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* * |
* * |
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* This program is free software; you can redistribute it and/or modify * |
* This program is free software; you can redistribute it and/or modify * |
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* it under the terms of the GNU General Public License as published by * |
* it under the terms of the GNU General Public License as published by * |
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#include <math.h> |
#include <math.h> |
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#include <stdint.h> |
#include <stdint.h> |
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#include "global.h" |
#include "global_private.h" |
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/// Needed for calculating frequency ratio used to pitch a sample |
/// Needed for calculating frequency ratio used to pitch a sample |
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#define TWELVEHUNDREDTH_ROOT_OF_TWO 1.000577789506555 |
#define TWELVEHUNDREDTH_ROOT_OF_TWO 1.000577789506555 |
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} |
} |
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/** |
/** |
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* Slower version of CentsToFreqRatio, for big values. |
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* |
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* @param cents - pitch value in cents (+1200 cents means +1 octave) |
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* @returns frequency ratio (e.g. +2.0 for +1 octave) |
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*/ |
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static double CentsToFreqRatioUnlimited(double Cents) { |
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int octaves = int(Cents / 1200); |
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double x = CentsToFreqRatio(Cents - octaves * 1200); |
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return octaves < 0 ? x / (1 << -octaves) : x * (1 << octaves); |
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} |
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/** |
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* Inverse function to CentsToFreqRatio(). This function is a bit |
* Inverse function to CentsToFreqRatio(). This function is a bit |
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* slow, so it should not be called too frequently. |
* slow, so it should not be called too frequently. |
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*/ |
*/ |
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return log(FreqRatio) / log(TWELVEHUNDREDTH_ROOT_OF_TWO); |
return log(FreqRatio) / log(TWELVEHUNDREDTH_ROOT_OF_TWO); |
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} |
} |
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/** |
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* Calculates the line ratio value representation (linear scale) |
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* of the @a decibel value provided (exponential scale). |
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* |
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* The context of audio acoustic sound pressure levels is assumed, and |
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* hence the field version of the dB unit is used here (which uses a |
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* linear factor of 20). This function is a bit slow, so it should |
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* not be called too frequently. |
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* |
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* @param decibel - sound pressure level in dB |
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* @returns linear ratio of the supplied dB value |
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*/ |
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static float DecibelToLinRatio(float decibel) { |
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return powf(10.f, decibel / 20.f); |
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} |
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/** |
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* Calculates the relatively summed average of a set of values. |
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* |
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* @param current - the current avaerage value of all previously summed values |
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* @param sample - new value to be applied as summed average to the existing values |
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* @param n - amount of sample values applied so far |
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* @returns new average value of all summed values (including the new @a sample) |
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*/ |
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inline static float RelativeSummedAvg(float current, float sample, int n) { |
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return current + (sample - current) / float(n); |
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} |
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private: |
private: |
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static float* pCentsToFreqTable; |
static float* pCentsToFreqTable; |
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