pandemonium_engine/servers/audio/audio_rb_resampler.h

#ifndef AUDIO_RB_RESAMPLER_H
#define AUDIO_RB_RESAMPLER_H
/*************************************************************************/
/*  audio_rb_resampler.h                                                 */
/*************************************************************************/
/*                       This file is part of:                           */
/*                           GODOT ENGINE                                */
/*                      https://godotengine.org                          */
/*************************************************************************/
/* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur.                 */
/* Copyright (c) 2014-2022 Godot Engine contributors (cf. AUTHORS.md).   */
/*                                                                       */
/* Permission is hereby granted, free of charge, to any person obtaining */
/* a copy of this software and associated documentation files (the       */
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/* the following conditions:                                             */
/*                                                                       */
/* The above copyright notice and this permission notice shall be        */
/* included in all copies or substantial portions of the Software.       */
/*                                                                       */
/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,       */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/*************************************************************************/

#include "core/os/memory.h"
#include "core/safe_refcount.h"
#include "core/typedefs.h"
#include "servers/audio_server.h"

struct AudioRBResampler {
	uint32_t rb_bits;
	uint32_t rb_len;
	uint32_t rb_mask;
	uint32_t read_buff_len;
	uint32_t channels;
	uint32_t src_mix_rate;
	uint32_t target_mix_rate;

	SafeNumeric<int> rb_read_pos;
	SafeNumeric<int> rb_write_pos;

	int32_t offset; //contains the fractional remainder of the resampler
	enum {
		MIX_FRAC_BITS = 13,
		MIX_FRAC_LEN = (1 << MIX_FRAC_BITS),
		MIX_FRAC_MASK = MIX_FRAC_LEN - 1,
	};

	float *read_buf;
	float *rb;

	template <int C>
	uint32_t _resample(AudioFrame *p_dest, int p_todo, int32_t p_increment);

public:
	_FORCE_INLINE_ void flush() {
		rb_read_pos.set(0);
		rb_write_pos.set(0);
		offset = 0;
	}

	_FORCE_INLINE_ bool is_ready() const {
		return rb != nullptr;
	}

	_FORCE_INLINE_ int get_total() const {
		return rb_len - 1;
	}

	_FORCE_INLINE_ int get_writer_space() const {
		int space, r, w;

		r = rb_read_pos.get();
		w = rb_write_pos.get();

		if (r == w) {
			space = rb_len - 1;
		} else if (w < r) {
			space = r - w - 1;
		} else {
			space = (rb_len - r) + w - 1;
		}

		return space;
	}

	_FORCE_INLINE_ int get_reader_space() const {
		int space, r, w;

		r = rb_read_pos.get();
		w = rb_write_pos.get();

		if (r == w) {
			space = 0;
		} else if (w < r) {
			space = rb_len - r + w;
		} else {
			space = w - r;
		}

		return space;
	}

	_FORCE_INLINE_ bool has_data() const {
		return rb && rb_read_pos.get() != rb_write_pos.get();
	}

	_FORCE_INLINE_ float *get_write_buffer() { return read_buf; }
	_FORCE_INLINE_ void write(uint32_t p_frames) {
		ERR_FAIL_COND(p_frames >= rb_len);

		int wp = rb_write_pos.get();

		switch (channels) {
			case 1: {
				for (uint32_t i = 0; i < p_frames; i++) {
					rb[wp] = read_buf[i];
					wp = (wp + 1) & rb_mask;
				}
			} break;
			case 2: {
				for (uint32_t i = 0; i < p_frames; i++) {
					rb[(wp << 1) + 0] = read_buf[(i << 1) + 0];
					rb[(wp << 1) + 1] = read_buf[(i << 1) + 1];
					wp = (wp + 1) & rb_mask;
				}
			} break;
			case 4: {
				for (uint32_t i = 0; i < p_frames; i++) {
					rb[(wp << 2) + 0] = read_buf[(i << 2) + 0];
					rb[(wp << 2) + 1] = read_buf[(i << 2) + 1];
					rb[(wp << 2) + 2] = read_buf[(i << 2) + 2];
					rb[(wp << 2) + 3] = read_buf[(i << 2) + 3];
					wp = (wp + 1) & rb_mask;
				}
			} break;
			case 6: {
				for (uint32_t i = 0; i < p_frames; i++) {
					rb[(wp * 6) + 0] = read_buf[(i * 6) + 0];
					rb[(wp * 6) + 1] = read_buf[(i * 6) + 1];
					rb[(wp * 6) + 2] = read_buf[(i * 6) + 2];
					rb[(wp * 6) + 3] = read_buf[(i * 6) + 3];
					rb[(wp * 6) + 4] = read_buf[(i * 6) + 4];
					rb[(wp * 6) + 5] = read_buf[(i * 6) + 5];
					wp = (wp + 1) & rb_mask;
				}
			} break;
		}

		rb_write_pos.set(wp);
	}

	int get_channel_count() const;

	Error setup(int p_channels, int p_src_mix_rate, int p_target_mix_rate, int p_buffer_msec, int p_minbuff_needed = -1);
	void clear();
	bool mix(AudioFrame *p_dest, int p_frames);
	int get_num_of_ready_frames();

	AudioRBResampler();
	~AudioRBResampler();
};

#endif // AUDIO_RB_RESAMPLER_H