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godot/servers/audio/audio_mixer_sw.cpp

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/*************************************************************************/
/* audio_mixer_sw.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* http://www.godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2017 Juan Linietsky, Ariel Manzur. */
/* */
/* 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, */
/* distribute, sublicense, and/or sell copies of the Software, and to */
/* permit persons to whom the Software is furnished to do so, subject to */
/* 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, */
/* 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 */
/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#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
}
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