/*************************************************************************/
/*  audio_mixer_sw.cpp                                                   */
/*************************************************************************/
/*                       This file is part of:                           */
/*                           GODOT ENGINE                                */
/*                      https://godotengine.org                          */
/*************************************************************************/
/* Copyright (c) 2007-2018 Juan Linietsky, Ariel Manzur.                 */
/* Copyright (c) 2014-2018 Godot Engine contributors (cf. AUTHORS.md)    */
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/* Permission is hereby granted, free of charge, to any person obtaining */
/* a copy of this software and associated documentation files (the       */
/* "Software"), to deal in the Software without restriction, including   */
/* without limitation the rights to use, copy, modify, merge, publish,   */
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/* permit persons to whom the Software is furnished to do so, subject to */
/* the following conditions:                                             */
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/* 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,       */
/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF    */
/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY  */
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/*************************************************************************/
#include "audio_mixer_sw.h"
#include "os/os.h"
#include "print_string.h"
//TODO implement FAST_AUDIO macro

#ifdef FAST_AUDIO
#define NO_REVERB
#endif

template <class Depth, bool is_stereo, bool is_ima_adpcm, bool use_filter, bool use_fx, AudioMixerSW::InterpolationType type, AudioMixerSW::MixChannels mix_mode>
void AudioMixerSW::do_resample(const Depth *p_src, int32_t *p_dst, ResamplerState *p_state) {

	// this function will be compiled branchless by any decent compiler

	int32_t final, final_r, next, next_r;
	int32_t *reverb_dst = p_state->reverb_buffer;
	while (p_state->amount--) {

		int32_t pos = p_state->pos >> MIX_FRAC_BITS;
		if (is_stereo && !is_ima_adpcm)
			pos <<= 1;

		if (is_ima_adpcm) {

			int sample_pos = pos + p_state->ima_adpcm[0].window_ofs;

			while (sample_pos > p_state->ima_adpcm[0].last_nibble) {

				static const int16_t _ima_adpcm_step_table[89] = {
					7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
					19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
					50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
					130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
					337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
					876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
					2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
					5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
					15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
				};

				static const int8_t _ima_adpcm_index_table[16] = {
					-1, -1, -1, -1, 2, 4, 6, 8,
					-1, -1, -1, -1, 2, 4, 6, 8
				};

				for (int i = 0; i < (is_stereo ? 2 : 1); i++) {

					int16_t nibble, diff, step;

					p_state->ima_adpcm[i].last_nibble++;
					const uint8_t *src_ptr = p_state->ima_adpcm[i].ptr;

					uint8_t nbb = src_ptr[(p_state->ima_adpcm[i].last_nibble >> 1) * (is_stereo ? 2 : 1) + i];
					nibble = (p_state->ima_adpcm[i].last_nibble & 1) ? (nbb >> 4) : (nbb & 0xF);
					step = _ima_adpcm_step_table[p_state->ima_adpcm[i].step_index];

					p_state->ima_adpcm[i].step_index += _ima_adpcm_index_table[nibble];
					if (p_state->ima_adpcm[i].step_index < 0)
						p_state->ima_adpcm[i].step_index = 0;
					if (p_state->ima_adpcm[i].step_index > 88)
						p_state->ima_adpcm[i].step_index = 88;

					diff = step >> 3;
					if (nibble & 1)
						diff += step >> 2;
					if (nibble & 2)
						diff += step >> 1;
					if (nibble & 4)
						diff += step;
					if (nibble & 8)
						diff = -diff;

					p_state->ima_adpcm[i].predictor += diff;
					if (p_state->ima_adpcm[i].predictor < -0x8000)
						p_state->ima_adpcm[i].predictor = -0x8000;
					else if (p_state->ima_adpcm[i].predictor > 0x7FFF)
						p_state->ima_adpcm[i].predictor = 0x7FFF;

					/* store loop if there */
					if (p_state->ima_adpcm[i].last_nibble == p_state->ima_adpcm[i].loop_pos) {

						p_state->ima_adpcm[i].loop_step_index = p_state->ima_adpcm[i].step_index;
						p_state->ima_adpcm[i].loop_predictor = p_state->ima_adpcm[i].predictor;
					}

					//printf("%i - %i - pred %i\n",int(p_state->ima_adpcm[i].last_nibble),int(nibble),int(p_state->ima_adpcm[i].predictor));
				}
			}

			final = p_state->ima_adpcm[0].predictor;
			if (is_stereo) {
				final_r = p_state->ima_adpcm[1].predictor;
			}

		} else {
			final = p_src[pos];
			if (is_stereo)
				final_r = p_src[pos + 1];

			if (sizeof(Depth) == 1) { /* conditions will not exist anymore when compiled! */
				final <<= 8;
				if (is_stereo)
					final_r <<= 8;
			}

			if (type == INTERPOLATION_LINEAR) {

				if (is_stereo) {

					next = p_src[pos + 2];
					next_r = p_src[pos + 3];
				} else {
					next = p_src[pos + 1];
				}

				if (sizeof(Depth) == 1) {
					next <<= 8;
					if (is_stereo)
						next_r <<= 8;
				}

				int32_t frac = int32_t(p_state->pos & MIX_FRAC_MASK);

				final = final + ((next - final) * frac >> MIX_FRAC_BITS);
				if (is_stereo)
					final_r = final_r + ((next_r - final_r) * frac >> MIX_FRAC_BITS);
			}
		}

		if (use_filter) {

			Channel::Mix::Filter *f = p_state->filter_l;
			float finalf = final;
			float pre = finalf;
			finalf = ((finalf * p_state->coefs.b0) + (f->hb[0] * p_state->coefs.b1) + (f->hb[1] * p_state->coefs.b2) + (f->ha[0] * p_state->coefs.a1) + (f->ha[1] * p_state->coefs.a2));

			f->ha[1] = f->ha[0];
			f->hb[1] = f->hb[0];
			f->hb[0] = pre;
			f->ha[0] = finalf;

			final = Math::fast_ftoi(finalf);

			if (is_stereo) {

				f = p_state->filter_r;
				finalf = final_r;
				pre = finalf;
				finalf = ((finalf * p_state->coefs.b0) + (f->hb[0] * p_state->coefs.b1) + (f->hb[1] * p_state->coefs.b2) + (f->ha[0] * p_state->coefs.a1) + (f->ha[1] * p_state->coefs.a2));
				f->ha[1] = f->ha[0];
				f->hb[1] = f->hb[0];
				f->hb[0] = pre;
				f->ha[0] = finalf;

				final_r = Math::fast_ftoi(finalf);
			}

			p_state->coefs.b0 += p_state->coefs_inc.b0;
			p_state->coefs.b1 += p_state->coefs_inc.b1;
			p_state->coefs.b2 += p_state->coefs_inc.b2;
			p_state->coefs.a1 += p_state->coefs_inc.a1;
			p_state->coefs.a2 += p_state->coefs_inc.a2;
		}

		if (!is_stereo) {
			final_r = final; //copy to right channel if stereo
		}

		//convert back to 24 bits and mix to buffers

		if (mix_mode == MIX_STEREO) {
			*p_dst++ += (final * (p_state->vol[0] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
			*p_dst++ += (final_r * (p_state->vol[1] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;

			p_state->vol[0] += p_state->vol_inc[0];
			p_state->vol[1] += p_state->vol_inc[1];

			if (use_fx) {
				*reverb_dst++ += (final * (p_state->reverb_vol[0] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
				*reverb_dst++ += (final_r * (p_state->reverb_vol[1] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
				p_state->reverb_vol[0] += p_state->reverb_vol_inc[0];
				p_state->reverb_vol[1] += p_state->reverb_vol_inc[1];
			}

		} else if (mix_mode == MIX_QUAD) {

			*p_dst++ += (final * (p_state->vol[0] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
			*p_dst++ += (final_r * (p_state->vol[1] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;

			*p_dst++ += (final * (p_state->vol[2] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
			*p_dst++ += (final_r * (p_state->vol[3] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;

			p_state->vol[0] += p_state->vol_inc[0];
			p_state->vol[1] += p_state->vol_inc[1];
			p_state->vol[2] += p_state->vol_inc[2];
			p_state->vol[3] += p_state->vol_inc[3];

			if (use_fx) {
				*reverb_dst++ += (final * (p_state->reverb_vol[0] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
				*reverb_dst++ += (final_r * (p_state->reverb_vol[1] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
				*reverb_dst++ += (final * (p_state->reverb_vol[2] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
				*reverb_dst++ += (final_r * (p_state->reverb_vol[3] >> MIX_VOLRAMP_FRAC_BITS)) >> MIX_VOL_MOVE_TO_24;
				p_state->reverb_vol[0] += p_state->reverb_vol_inc[0];
				p_state->reverb_vol[1] += p_state->reverb_vol_inc[1];
				p_state->reverb_vol[2] += p_state->reverb_vol_inc[2];
				p_state->reverb_vol[3] += p_state->reverb_vol_inc[3];
			}
		}

		p_state->pos += p_state->increment;
	}
}

void AudioMixerSW::mix_channel(Channel &c) {

	if (!sample_manager->is_sample(c.sample)) {
		// sample is gone!
		c.active = false;
		return;
	}

	/* some 64-bit fixed point precaches */

	int64_t loop_begin_fp = ((int64_t)sample_manager->sample_get_loop_begin(c.sample) << MIX_FRAC_BITS);
	int64_t loop_end_fp = ((int64_t)sample_manager->sample_get_loop_end(c.sample) << MIX_FRAC_BITS);
	int64_t length_fp = ((int64_t)sample_manager->sample_get_length(c.sample) << MIX_FRAC_BITS);
	int64_t begin_limit = (sample_manager->sample_get_loop_format(c.sample) != AS::SAMPLE_LOOP_NONE) ? loop_begin_fp : 0;
	int64_t end_limit = (sample_manager->sample_get_loop_format(c.sample) != AS::SAMPLE_LOOP_NONE) ? loop_end_fp : length_fp;
	bool is_stereo = sample_manager->sample_is_stereo(c.sample);

	int32_t todo = mix_chunk_size;
	//	int mixed=0;
	bool use_filter = false;

	ResamplerState rstate;

	/* compute voume ramps, increment, etc */

	for (int i = 0; i < mix_channels; i++) {
		c.mix.old_vol[i] = c.mix.vol[i];
		c.mix.old_reverb_vol[i] = c.mix.reverb_vol[i];
		c.mix.old_chorus_vol[i] = c.mix.chorus_vol[i];
	}

	float vol = c.vol * channel_nrg;

	float reverb_vol = c.reverb_send * channel_nrg;
	float chorus_vol = c.chorus_send * channel_nrg;

	if (mix_channels == 2) {
		//stereo pan
		float pan = c.pan * 0.5 + 0.5;
		float panv[2] = {
			(1.0 - pan) * (1 << MIX_VOL_FRAC_BITS),
			(pan) * (1 << MIX_VOL_FRAC_BITS)
		};

		for (int i = 0; i < 2; i++) {

			c.mix.vol[i] = Math::fast_ftoi(vol * panv[i]);
			c.mix.reverb_vol[i] = Math::fast_ftoi(reverb_vol * panv[i]);
			c.mix.chorus_vol[i] = Math::fast_ftoi(chorus_vol * panv[i]);
		}

	} else {
		//qudra pan
		float panx = c.pan * 0.5 + 0.5;
		float pany = c.depth * 0.5 + 0.5;
		// with this model every speaker plays at 0.25 energy at the center.. i'm not sure if it's correct but it seems to be balanced
		float panv[4] = {
			(1.0 - pany) * (1.0 - panx) * (1 << MIX_VOL_FRAC_BITS),
			(1.0 - pany) * (panx) * (1 << MIX_VOL_FRAC_BITS),
			(pany) * (1.0 - panx) * (1 << MIX_VOL_FRAC_BITS),
			(pany) * (panx) * (1 << MIX_VOL_FRAC_BITS)
		};

		for (int i = 0; i < 4; i++) {

			c.mix.vol[i] = Math::fast_ftoi(vol * panv[i]);
			c.mix.reverb_vol[i] = Math::fast_ftoi(reverb_vol * panv[i]);
			c.mix.chorus_vol[i] = Math::fast_ftoi(chorus_vol * panv[i]);
		}
	}

	if (c.first_mix) { // avoid ramp up

		for (int i = 0; i < mix_channels; i++) {
			c.mix.old_vol[i] = c.mix.vol[i];
			c.mix.old_reverb_vol[i] = c.mix.reverb_vol[i];
			c.mix.old_chorus_vol[i] = c.mix.chorus_vol[i];
		}

		c.first_mix = false;
	}

	Channel::Filter::Coefs filter_coefs;
	Channel::Filter::Coefs filter_inc;

	if (c.filter.type != AudioMixer::FILTER_NONE) {

		filter_coefs = c.filter.old_coefs;
		filter_inc.b0 = (c.filter.coefs.b0 - filter_coefs.b0) / (1 << mix_chunk_bits);
		filter_inc.b1 = (c.filter.coefs.b1 - filter_coefs.b1) / (1 << mix_chunk_bits);
		filter_inc.b2 = (c.filter.coefs.b2 - filter_coefs.b2) / (1 << mix_chunk_bits);
		filter_inc.a1 = (c.filter.coefs.a1 - filter_coefs.a1) / (1 << mix_chunk_bits);
		filter_inc.a2 = (c.filter.coefs.a2 - filter_coefs.a2) / (1 << mix_chunk_bits);
		use_filter = true;
	}

	if (c.mix.increment > 0)
		c.mix.increment = ((int64_t)c.speed << MIX_FRAC_BITS) / mix_rate;
	else
		c.mix.increment = -((int64_t)c.speed << MIX_FRAC_BITS) / mix_rate;

	//volume ramp

	for (int i = 0; i < mix_channels; i++) {
		rstate.vol_inc[i] = ((c.mix.vol[i] - c.mix.old_vol[i]) << MIX_VOLRAMP_FRAC_BITS) >> mix_chunk_bits;
		rstate.vol[i] = c.mix.old_vol[i] << MIX_VOLRAMP_FRAC_BITS;
		rstate.reverb_vol_inc[i] = ((c.mix.reverb_vol[i] - c.mix.old_reverb_vol[i]) << MIX_VOLRAMP_FRAC_BITS) >> mix_chunk_bits;
		rstate.reverb_vol[i] = c.mix.old_reverb_vol[i] << MIX_VOLRAMP_FRAC_BITS;
		rstate.chorus_vol_inc[i] = ((c.mix.chorus_vol[i] - c.mix.old_chorus_vol[i]) << MIX_VOLRAMP_FRAC_BITS) >> mix_chunk_bits;
		rstate.chorus_vol[i] = c.mix.old_chorus_vol[i] << MIX_VOLRAMP_FRAC_BITS;
	}

	//looping

	AS::SampleLoopFormat loop_format = sample_manager->sample_get_loop_format(c.sample);
	AS::SampleFormat format = sample_manager->sample_get_format(c.sample);

	bool use_fx = false;

	if (fx_enabled) {

		for (int i = 0; i < mix_channels; i++) {
			if (c.mix.old_reverb_vol[i] || c.mix.reverb_vol[i] || c.mix.old_chorus_vol[i] || c.mix.chorus_vol[i]) {
				use_fx = true;
				break;
			}
		}
	}

	/* audio data */

	const void *data = sample_manager->sample_get_data_ptr(c.sample);
	int32_t *dst_buff = mix_buffer;

#ifndef NO_REVERB
	rstate.reverb_buffer = reverb_state[c.reverb_room].buffer;
#endif

	/* @TODO validar loops al registrar? */

	rstate.coefs = filter_coefs;
	rstate.coefs_inc = filter_inc;
	rstate.filter_l = &c.mix.filter_l;
	rstate.filter_r = &c.mix.filter_r;

	if (format == AS::SAMPLE_FORMAT_IMA_ADPCM) {

		rstate.ima_adpcm = c.mix.ima_adpcm;
		if (loop_format != AS::SAMPLE_LOOP_NONE) {
			c.mix.ima_adpcm[0].loop_pos = loop_begin_fp >> MIX_FRAC_BITS;
			c.mix.ima_adpcm[1].loop_pos = loop_begin_fp >> MIX_FRAC_BITS;
			loop_format = AS::SAMPLE_LOOP_FORWARD;
		}
	}

	while (todo > 0) {

		int64_t limit = 0;
		int32_t target = 0, aux = 0;

		/** LOOP CHECKING **/

		if (c.mix.increment < 0) {
			/* going backwards */

			if (loop_format != AS::SAMPLE_LOOP_NONE && c.mix.offset < loop_begin_fp) {
				/* loopstart reached */
				if (loop_format == AS::SAMPLE_LOOP_PING_PONG) {
					/* bounce ping pong */
					c.mix.offset = loop_begin_fp + (loop_begin_fp - c.mix.offset);
					c.mix.increment = -c.mix.increment;
				} else {
					/* go to loop-end */
					c.mix.offset = loop_end_fp - (loop_begin_fp - c.mix.offset);
				}
			} else {
				/* check for sample not reaching begining */
				if (c.mix.offset < 0) {

					c.active = false;
					break;
				}
			}
		} else {
			/* going forward */
			if (loop_format != AS::SAMPLE_LOOP_NONE && c.mix.offset >= loop_end_fp) {
				/* loopend reached */

				if (loop_format == AS::SAMPLE_LOOP_PING_PONG) {
					/* bounce ping pong */
					c.mix.offset = loop_end_fp - (c.mix.offset - loop_end_fp);
					c.mix.increment = -c.mix.increment;
				} else {
					/* go to loop-begin */

					if (format == AS::SAMPLE_FORMAT_IMA_ADPCM) {
						for (int i = 0; i < 2; i++) {
							c.mix.ima_adpcm[i].step_index = c.mix.ima_adpcm[i].loop_step_index;
							c.mix.ima_adpcm[i].predictor = c.mix.ima_adpcm[i].loop_predictor;
							c.mix.ima_adpcm[i].last_nibble = loop_begin_fp >> MIX_FRAC_BITS;
						}
						c.mix.offset = loop_begin_fp;
					} else {
						c.mix.offset = loop_begin_fp + (c.mix.offset - loop_end_fp);
					}
				}
			} else {
				/* no loop, check for end of sample */
				if (c.mix.offset >= length_fp) {

					c.active = false;
					break;
				}
			}
		}

		/** MIXCOUNT COMPUTING **/

		/* next possible limit (looppoints or sample begin/end */
		limit = (c.mix.increment < 0) ? begin_limit : end_limit;

		/* compute what is shorter, the todo or the limit? */
		aux = (limit - c.mix.offset) / c.mix.increment + 1;
		target = (aux < todo) ? aux : todo; /* mix target is the shorter buffer */

		/* check just in case */
		if (target <= 0) {
			c.active = false;
			break;
		}

		todo -= target;

		int32_t offset = c.mix.offset & mix_chunk_mask; /* strip integer */
		c.mix.offset -= offset;

		rstate.increment = c.mix.increment;
		rstate.amount = target;
		rstate.pos = offset;

		/* Macros to call the resample function for all possibilities, creating a dedicated-non branchy function call for each thanks to template magic*/

#define CALL_RESAMPLE_FUNC(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode) \
	do_resample<m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode>(           \
			src_ptr,                                                                                 \
			dst_buff, &rstate);

#define CALL_RESAMPLE_INTERP(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode)            \
	if (m_interp == INTERPOLATION_RAW) {                                                                          \
		CALL_RESAMPLE_FUNC(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, INTERPOLATION_RAW, m_mode);    \
	} else if (m_interp == INTERPOLATION_LINEAR) {                                                                \
		CALL_RESAMPLE_FUNC(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, INTERPOLATION_LINEAR, m_mode); \
	} else if (m_interp == INTERPOLATION_CUBIC) {                                                                 \
		CALL_RESAMPLE_FUNC(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, INTERPOLATION_CUBIC, m_mode);  \
	}

#define CALL_RESAMPLE_FX(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode)   \
	if (m_use_fx) {                                                                                  \
		CALL_RESAMPLE_INTERP(m_depth, m_stereo, m_ima_adpcm, m_use_filter, true, m_interp, m_mode);  \
	} else {                                                                                         \
		CALL_RESAMPLE_INTERP(m_depth, m_stereo, m_ima_adpcm, m_use_filter, false, m_interp, m_mode); \
	}

#define CALL_RESAMPLE_FILTER(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode) \
	if (m_use_filter) {                                                                                \
		CALL_RESAMPLE_FX(m_depth, m_stereo, m_ima_adpcm, true, m_use_fx, m_interp, m_mode);            \
	} else {                                                                                           \
		CALL_RESAMPLE_FX(m_depth, m_stereo, m_ima_adpcm, false, m_use_fx, m_interp, m_mode);           \
	}

#define CALL_RESAMPLE_STEREO(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode) \
	if (m_stereo) {                                                                                    \
		CALL_RESAMPLE_FILTER(m_depth, true, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode);    \
	} else {                                                                                           \
		CALL_RESAMPLE_FILTER(m_depth, false, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode);   \
	}

#define CALL_RESAMPLE_MODE(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, m_mode)        \
	if (m_mode == MIX_STEREO) {                                                                             \
		CALL_RESAMPLE_STEREO(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, MIX_STEREO); \
	} else {                                                                                                \
		CALL_RESAMPLE_STEREO(m_depth, m_stereo, m_ima_adpcm, m_use_filter, m_use_fx, m_interp, MIX_QUAD);   \
	}

		if (format == AS::SAMPLE_FORMAT_PCM8) {

			int8_t *src_ptr = &((int8_t *)data)[(c.mix.offset >> MIX_FRAC_BITS) << (is_stereo ? 1 : 0)];
			CALL_RESAMPLE_MODE(int8_t, is_stereo, false, use_filter, use_fx, interpolation_type, mix_channels);

		} else if (format == AS::SAMPLE_FORMAT_PCM16) {
			int16_t *src_ptr = &((int16_t *)data)[(c.mix.offset >> MIX_FRAC_BITS) << (is_stereo ? 1 : 0)];
			CALL_RESAMPLE_MODE(int16_t, is_stereo, false, use_filter, use_fx, interpolation_type, mix_channels);

		} else if (format == AS::SAMPLE_FORMAT_IMA_ADPCM) {
			for (int i = 0; i < 2; i++) {
				c.mix.ima_adpcm[i].window_ofs = c.mix.offset >> MIX_FRAC_BITS;
				c.mix.ima_adpcm[i].ptr = (const uint8_t *)data;
			}
			int8_t *src_ptr = NULL;
			CALL_RESAMPLE_MODE(int8_t, is_stereo, true, use_filter, use_fx, interpolation_type, mix_channels);
		}

		c.mix.offset += rstate.pos;
		dst_buff += target * mix_channels;
		rstate.reverb_buffer += target * mix_channels;
	}

	c.filter.old_coefs = c.filter.coefs;
}

void AudioMixerSW::mix_chunk() {

	ERR_FAIL_COND(mix_chunk_left);

	inside_mix = true;

	// emit tick in usecs
	for (int i = 0; i < mix_chunk_size * mix_channels; i++) {

		mix_buffer[i] = 0;
	}
#ifndef NO_REVERB
	for (int i = 0; i < max_reverbs; i++)
		reverb_state[i].used_in_chunk = false;
#endif

	audio_mixer_chunk_call(mix_chunk_size);

	int ac = 0;
	for (int i = 0; i < MAX_CHANNELS; i++) {

		if (!channels[i].active)
			continue;
		ac++;

		/* process volume */
		Channel &c = channels[i];
#ifndef NO_REVERB
		bool has_reverb = c.reverb_send > CMP_EPSILON && fx_enabled;
		if (has_reverb || c.had_prev_reverb) {

			if (!reverb_state[c.reverb_room].used_in_chunk) {
				//zero the room
				int32_t *buff = reverb_state[c.reverb_room].buffer;
				int len = mix_chunk_size * mix_channels;
				for (int j = 0; j < len; j++) {

					buff[j] = 0; // buffer in use, clear it for appending
				}
				reverb_state[c.reverb_room].used_in_chunk = true;
			}
		}
#else
		bool has_reverb = false;
#endif
		bool has_chorus = c.chorus_send > CMP_EPSILON && fx_enabled;

		mix_channel(c);

		c.had_prev_reverb = has_reverb;
		c.had_prev_chorus = has_chorus;
	}

//process reverb
#ifndef NO_REVERB
	if (fx_enabled) {

		for (int i = 0; i < max_reverbs; i++) {

			if (!reverb_state[i].enabled && !reverb_state[i].used_in_chunk)
				continue; //this reverb is not in use

			int32_t *src = NULL;

			if (reverb_state[i].used_in_chunk)
				src = reverb_state[i].buffer;
			else
				src = zero_buffer;

			bool in_use = false;

			int passes = mix_channels / 2;

			for (int j = 0; j < passes; j++) {

				if (reverb_state[i].reverb[j].process((int *)&src[j * 2], (int *)&mix_buffer[j * 2], mix_chunk_size, passes))
					in_use = true;
			}

			if (in_use) {
				reverb_state[i].enabled = true;
				reverb_state[i].frames_idle = 0;
				//copy data over

			} else {
				reverb_state[i].frames_idle += mix_chunk_size;
				if (false) { // go idle because too many frames passed
					//disable this reverb, as nothing important happened on it
					reverb_state[i].enabled = false;
					reverb_state[i].frames_idle = 0;
				}
			}
		}
	}
#endif
	mix_chunk_left = mix_chunk_size;
	inside_mix = false;
}

int AudioMixerSW::mix(int32_t *p_buffer, int p_frames) {

	int todo = p_frames;
	int mixes = 0;

	while (todo) {

		if (!mix_chunk_left) {

			if (step_callback)
				step_callback(step_udata);
			mix_chunk();
			mixes++;
		}

		int to_mix = MIN(mix_chunk_left, todo);
		int from = mix_chunk_size - mix_chunk_left;

		for (int i = 0; i < to_mix * 2; i++) {

			(*p_buffer++) = mix_buffer[from * 2 + i];
		}

		mix_chunk_left -= to_mix;
		todo -= to_mix;
	}

	return mixes;
}

uint64_t AudioMixerSW::get_step_usecs() const {

	double mct = (1 << mix_chunk_bits) / double(mix_rate);
	return mct * 1000000.0;
}

int AudioMixerSW::_get_channel(ChannelID p_channel) const {

	if (p_channel < 0) {
		return -1;
	}

	int idx = p_channel % MAX_CHANNELS;
	int check = p_channel / MAX_CHANNELS;
	ERR_FAIL_INDEX_V(idx, MAX_CHANNELS, -1);
	if (channels[idx].check != check) {
		return -1;
	}
	if (!channels[idx].active) {
		return -1;
	}

	return idx;
}

AudioMixer::ChannelID AudioMixerSW::channel_alloc(RID p_sample) {

	ERR_FAIL_COND_V(!sample_manager->is_sample(p_sample), INVALID_CHANNEL);

	int index = -1;
	for (int i = 0; i < MAX_CHANNELS; i++) {

		if (!channels[i].active) {
			index = i;
			break;
		}
	}

	if (index == -1)
		return INVALID_CHANNEL;

	Channel &c = channels[index];

	// init variables
	c.sample = p_sample;
	c.vol = 1;
	c.pan = 0;
	c.depth = 0;
	c.height = 0;
	c.chorus_send = 0;
	c.reverb_send = 0;
	c.reverb_room = REVERB_HALL;
	c.positional = false;
	c.filter.type = FILTER_NONE;
	c.speed = sample_manager->sample_get_mix_rate(p_sample);
	c.active = true;
	c.check = channel_id_count++;
	c.first_mix = true;

	// init mix variables

	c.mix.offset = 0;
	c.mix.increment = 1;
	//zero everything when this errors
	for (int i = 0; i < 4; i++) {
		c.mix.vol[i] = 0;
		c.mix.reverb_vol[i] = 0;
		c.mix.chorus_vol[i] = 0;

		c.mix.old_vol[i] = 0;
		c.mix.old_reverb_vol[i] = 0;
		c.mix.old_chorus_vol[i] = 0;
	}

	c.had_prev_chorus = false;
	c.had_prev_reverb = false;
	c.had_prev_vol = false;

	if (sample_manager->sample_get_format(c.sample) == AudioServer::SAMPLE_FORMAT_IMA_ADPCM) {

		for (int i = 0; i < 2; i++) {
			c.mix.ima_adpcm[i].step_index = 0;
			c.mix.ima_adpcm[i].predictor = 0;
			c.mix.ima_adpcm[i].loop_step_index = 0;
			c.mix.ima_adpcm[i].loop_predictor = 0;
			c.mix.ima_adpcm[i].last_nibble = -1;
			c.mix.ima_adpcm[i].loop_pos = 0x7FFFFFFF;
			c.mix.ima_adpcm[i].window_ofs = 0;
			c.mix.ima_adpcm[i].ptr = NULL;
		}
	}

	ChannelID ret_id = index + c.check * MAX_CHANNELS;

	return ret_id;
}

void AudioMixerSW::channel_set_volume(ChannelID p_channel, float p_gain) {

	if (p_gain > 3) // avoid gain going too high
		p_gain = 3;
	if (p_gain < 0)
		p_gain = 0;

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return;
	Channel &c = channels[chan];

	//Math::exp( p_db * 0.11512925464970228420089957273422 );
	c.vol = p_gain;
}

void AudioMixerSW::channel_set_pan(ChannelID p_channel, float p_pan, float p_depth, float p_height) {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return;
	Channel &c = channels[chan];

	c.pan = p_pan;
	c.depth = p_depth;
	c.height = p_height;
}
void AudioMixerSW::channel_set_filter(ChannelID p_channel, FilterType p_type, float p_cutoff, float p_resonance, float p_gain) {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return;

	Channel &c = channels[chan];

	if (c.filter.type == p_type && c.filter.cutoff == p_cutoff && c.filter.resonance == p_resonance && c.filter.gain == p_gain)
		return; //bye

	bool type_changed = p_type != c.filter.type;

	c.filter.type = p_type;
	c.filter.cutoff = p_cutoff;
	c.filter.resonance = p_resonance;
	c.filter.gain = p_gain;

	AudioFilterSW filter;
	switch (p_type) {
		case FILTER_NONE: {

			return; //do nothing else
		} break;
		case FILTER_LOWPASS: {
			filter.set_mode(AudioFilterSW::LOWPASS);
		} break;
		case FILTER_BANDPASS: {
			filter.set_mode(AudioFilterSW::BANDPASS);
		} break;
		case FILTER_HIPASS: {
			filter.set_mode(AudioFilterSW::HIGHPASS);
		} break;
		case FILTER_NOTCH: {
			filter.set_mode(AudioFilterSW::NOTCH);
		} break;
		case FILTER_PEAK: {
			filter.set_mode(AudioFilterSW::PEAK);
		} break;
		case FILTER_BANDLIMIT: {
			filter.set_mode(AudioFilterSW::BANDLIMIT);
		} break;
		case FILTER_LOW_SHELF: {
			filter.set_mode(AudioFilterSW::LOWSHELF);
		} break;
		case FILTER_HIGH_SHELF: {
			filter.set_mode(AudioFilterSW::HIGHSHELF);
		} break;
	}

	filter.set_cutoff(p_cutoff);
	filter.set_resonance(p_resonance);
	filter.set_gain(p_gain);
	filter.set_sampling_rate(mix_rate);
	filter.set_stages(1);

	AudioFilterSW::Coeffs coefs;
	filter.prepare_coefficients(&coefs);

	if (!type_changed)
		c.filter.old_coefs = c.filter.coefs;

	c.filter.coefs.b0 = coefs.b0;
	c.filter.coefs.b1 = coefs.b1;
	c.filter.coefs.b2 = coefs.b2;
	c.filter.coefs.a1 = coefs.a1;
	c.filter.coefs.a2 = coefs.a2;

	if (type_changed) {
		//type changed reset filter
		c.filter.old_coefs = c.filter.coefs;
		c.mix.filter_l.ha[0] = 0;
		c.mix.filter_l.ha[1] = 0;
		c.mix.filter_l.hb[0] = 0;
		c.mix.filter_l.hb[1] = 0;
		c.mix.filter_r.ha[0] = 0;
		c.mix.filter_r.ha[1] = 0;
		c.mix.filter_r.hb[0] = 0;
		c.mix.filter_r.hb[1] = 0;
	}
}
void AudioMixerSW::channel_set_chorus(ChannelID p_channel, float p_chorus) {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return;

	Channel &c = channels[chan];
	c.chorus_send = p_chorus;
}
void AudioMixerSW::channel_set_reverb(ChannelID p_channel, ReverbRoomType p_room_type, float p_reverb) {

	ERR_FAIL_INDEX(p_room_type, MAX_REVERBS);
	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return;

	Channel &c = channels[chan];
	c.reverb_room = p_room_type;
	c.reverb_send = p_reverb;
}

void AudioMixerSW::channel_set_mix_rate(ChannelID p_channel, int p_mix_rate) {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return;

	Channel &c = channels[chan];
	c.speed = p_mix_rate;
}
void AudioMixerSW::channel_set_positional(ChannelID p_channel, bool p_positional) {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return;

	Channel &c = channels[chan];
	c.positional = p_positional;
}

float AudioMixerSW::channel_get_volume(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	//Math::log( c.vol ) * 8.6858896380650365530225783783321;
	return c.vol;
}

float AudioMixerSW::channel_get_pan(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.pan;
}
float AudioMixerSW::channel_get_pan_depth(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.depth;
}
float AudioMixerSW::channel_get_pan_height(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.height;
}
AudioMixer::FilterType AudioMixerSW::channel_get_filter_type(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return FILTER_NONE;

	const Channel &c = channels[chan];
	return c.filter.type;
}
float AudioMixerSW::channel_get_filter_cutoff(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.filter.cutoff;
}
float AudioMixerSW::channel_get_filter_resonance(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.filter.resonance;
}

float AudioMixerSW::channel_get_filter_gain(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.filter.gain;
}

float AudioMixerSW::channel_get_chorus(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.chorus_send;
}
AudioMixer::ReverbRoomType AudioMixerSW::channel_get_reverb_type(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return REVERB_HALL;

	const Channel &c = channels[chan];
	return c.reverb_room;
}
float AudioMixerSW::channel_get_reverb(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.reverb_send;
}

int AudioMixerSW::channel_get_mix_rate(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return 0;

	const Channel &c = channels[chan];
	return c.speed;
}
bool AudioMixerSW::channel_is_positional(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return false;

	const Channel &c = channels[chan];
	return c.positional;
}

bool AudioMixerSW::channel_is_valid(ChannelID p_channel) const {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return false;
	return channels[chan].active;
}

void AudioMixerSW::channel_free(ChannelID p_channel) {

	int chan = _get_channel(p_channel);
	if (chan < 0 || chan >= MAX_CHANNELS)
		return;

	Channel &c = channels[chan];

	if (!c.active)
		return;

	bool has_vol = false;

	for (int i = 0; i < mix_channels; i++) {

		if (c.mix.vol[i])
			has_vol = true;
		if (c.mix.reverb_vol[i])
			has_vol = true;
		if (c.mix.chorus_vol[i])
			has_vol = true;
	}
	if (c.active && has_vol && inside_mix) {
		// drive voice to zero, and run a chunk, the VRAMP will fade it good
		c.vol = 0;
		c.reverb_send = 0;
		c.chorus_send = 0;
		mix_channel(c);
	}
	/* @TODO RAMP DOWN ON STOP */
	c.active = false;
}

AudioMixerSW::AudioMixerSW(SampleManagerSW *p_sample_manager, int p_desired_latency_ms, int p_mix_rate, MixChannels p_mix_channels, bool p_use_fx, InterpolationType p_interp, MixStepCallback p_step_callback, void *p_step_udata) {

	if (OS::get_singleton()->is_stdout_verbose()) {
		print_line("AudioServerSW Params: ");
		print_line(" -mix chans: " + itos(p_mix_channels));
		print_line(" -mix rate: " + itos(p_mix_rate));
		print_line(" -latency: " + itos(p_desired_latency_ms));
		print_line(" -fx: " + itos(p_use_fx));
		print_line(" -interp: " + itos(p_interp));
	}
	sample_manager = p_sample_manager;
	mix_channels = p_mix_channels;
	mix_rate = p_mix_rate;
	step_callback = p_step_callback;
	step_udata = p_step_udata;

	mix_chunk_bits = nearest_shift(p_desired_latency_ms * p_mix_rate / 1000);

	mix_chunk_size = (1 << mix_chunk_bits);
	mix_chunk_mask = mix_chunk_size - 1;
	mix_buffer = memnew_arr(int32_t, mix_chunk_size * mix_channels);
#ifndef NO_REVERB
	zero_buffer = memnew_arr(int32_t, mix_chunk_size * mix_channels);
	for (int i = 0; i < mix_chunk_size * mix_channels; i++)
		zero_buffer[i] = 0; //zero buffer is zero...

	max_reverbs = MAX_REVERBS;
	int reverberators = mix_channels / 2;

	reverb_state = memnew_arr(ReverbState, max_reverbs);
	for (int i = 0; i < max_reverbs; i++) {
		reverb_state[i].enabled = false;
		reverb_state[i].reverb = memnew_arr(ReverbSW, reverberators);
		reverb_state[i].buffer = memnew_arr(int32_t, mix_chunk_size * mix_channels);
		reverb_state[i].frames_idle = 0;
		for (int j = 0; j < reverberators; j++) {
			static ReverbSW::ReverbMode modes[MAX_REVERBS] = { ReverbSW::REVERB_MODE_STUDIO_SMALL, ReverbSW::REVERB_MODE_STUDIO_MEDIUM, ReverbSW::REVERB_MODE_STUDIO_LARGE, ReverbSW::REVERB_MODE_HALL };
			reverb_state[i].reverb[j].set_mix_rate(p_mix_rate);
			reverb_state[i].reverb[j].set_mode(modes[i]);
		}
	}
	fx_enabled = p_use_fx;
#else
	fx_enabled = false;
#endif
	mix_chunk_left = 0;

	interpolation_type = p_interp;
	channel_id_count = 1;
	inside_mix = false;
	channel_nrg = 1.0;
}

void AudioMixerSW::set_mixer_volume(float p_volume) {

	channel_nrg = p_volume;
}

AudioMixerSW::~AudioMixerSW() {

	memdelete_arr(mix_buffer);

#ifndef NO_REVERB
	memdelete_arr(zero_buffer);
	for (int i = 0; i < max_reverbs; i++) {
		memdelete_arr(reverb_state[i].reverb);
		memdelete_arr(reverb_state[i].buffer);
	}
	memdelete_arr(reverb_state);
#endif
}