godot/servers/audio/effects/audio_effect_pitch_shift.cpp

/**************************************************************************/
/*  audio_effect_pitch_shift.cpp                                          */
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#include "audio_effect_pitch_shift.h"

#include "core/math/math_funcs.h"
#include "servers/audio_server.h"

/* Thirdparty code, so disable clang-format with Godot style */
/* clang-format off */

/****************************************************************************
*
* NAME: smbPitchShift.cpp
* VERSION: 1.2
* HOME URL: https://blogs.zynaptiq.com/bernsee
* KNOWN BUGS: none
*
* SYNOPSIS: Routine for doing pitch shifting while maintaining
* duration using the Short Time Fourier Transform.
*
* DESCRIPTION: The routine takes a pitchShift factor value which is between 0.5
* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
* the pitch. numSampsToProcess tells the routine how many samples in indata[0...
* numSampsToProcess-1] should be pitch shifted and moved to outdata[0 ...
* numSampsToProcess-1]. The two buffers can be identical (ie. it can process the
* data in-place). fftFrameSize defines the FFT frame size used for the
* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
* MAX_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
* oversampling factor which also determines the overlap between adjacent STFT
* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
* recommended for best quality. sampleRate takes the sample rate for the signal
* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in
* indata[] should be in the range [-1.0, 1.0), which is also the output range
* for the data, make sure you scale the data accordingly (for 16bit signed integers
* you would have to divide (and multiply) by 32768).
*
* COPYRIGHT 1999-2015 Stephan M. Bernsee <s.bernsee [AT] zynaptiq [DOT] com>
*
* 						The Wide Open License (WOL)
*
* Permission to use, copy, modify, distribute and sell this software and its
* documentation for any purpose is hereby granted without fee, provided that
* the above copyright notice and this license appear in all source copies.
* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
* ANY KIND. See https://dspguru.com/wide-open-license/ for more information.
*
*****************************************************************************/

void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata,int stride) {}



void SMBPitchShift::smbFft(float *fftBuffer, long fftFrameSize, long sign)
/*
	FFT routine, (C)1996 S.M.Bernsee. Sign = -1 is FFT, 1 is iFFT (inverse)
	Fills fftBuffer[0...2*fftFrameSize-1] with the Fourier transform of the
	time domain data in fftBuffer[0...2*fftFrameSize-1]. The FFT array takes
	and returns the cosine and sine parts in an interleaved manner, ie.
	fftBuffer[0] = cosPart[0], fftBuffer[1] = sinPart[0], asf. fftFrameSize
	must be a power of 2. It expects a complex input signal (see footnote 2),
	ie. when working with 'common' audio signals our input signal has to be
	passed as {in[0],0.,in[1],0.,in[2],0.,...} asf. In that case, the transform
	of the frequencies of interest is in fftBuffer[0...fftFrameSize].
*/
{}


/* Godot code again */
/* clang-format on */

void AudioEffectPitchShiftInstance::process(const AudioFrame *p_src_frames, AudioFrame *p_dst_frames, int p_frame_count) {}

Ref<AudioEffectInstance> AudioEffectPitchShift::instantiate() {}

void AudioEffectPitchShift::set_pitch_scale(float p_pitch_scale) {}

float AudioEffectPitchShift::get_pitch_scale() const {}

void AudioEffectPitchShift::set_oversampling(int p_oversampling) {}

int AudioEffectPitchShift::get_oversampling() const {}

void AudioEffectPitchShift::set_fft_size(FFTSize p_fft_size) {}

AudioEffectPitchShift::FFTSize AudioEffectPitchShift::get_fft_size() const {}

void AudioEffectPitchShift::_bind_methods() {}