1226 lines
33 KiB
C++
1226 lines
33 KiB
C++
/*************************************************************************/
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/* audio_mixer_sw.cpp */
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/*************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* http://www.godotengine.org */
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/*************************************************************************/
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/* Copyright (c) 2007-2015 Juan Linietsky, Ariel Manzur. */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/*************************************************************************/
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#include "audio_mixer_sw.h"
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#include "print_string.h"
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#include "os/os.h"
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//TODO implement FAST_AUDIO macro
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#ifdef FAST_AUDIO
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#define NO_REVERB
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#endif
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template<class Depth,bool is_stereo,bool is_ima_adpcm,bool use_filter,bool use_fx,AudioMixerSW::InterpolationType type,AudioMixerSW::MixChannels mix_mode>
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void AudioMixerSW::do_resample(const Depth* p_src, int32_t *p_dst, ResamplerState *p_state) {
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// this function will be compiled branchless by any decent compiler
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int32_t final,final_r,next,next_r;
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int32_t *reverb_dst = p_state->reverb_buffer;
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while (p_state->amount--) {
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int32_t pos=p_state->pos >> MIX_FRAC_BITS;
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if (is_stereo && !is_ima_adpcm)
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pos<<=1;
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if (is_ima_adpcm) {
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int sample_pos = pos + p_state->ima_adpcm[0].window_ofs;
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while(sample_pos>p_state->ima_adpcm[0].last_nibble) {
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static const int16_t _ima_adpcm_step_table[89] = {
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7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
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19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
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50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
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130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
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876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
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2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
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5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
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};
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static const int8_t _ima_adpcm_index_table[16] = {
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-1, -1, -1, -1, 2, 4, 6, 8,
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-1, -1, -1, -1, 2, 4, 6, 8
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};
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for(int i=0;i<(is_stereo?2:1);i++) {
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int16_t nibble,signed_nibble,diff,step;
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p_state->ima_adpcm[i].last_nibble++;
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const uint8_t *src_ptr=p_state->ima_adpcm[i].ptr;
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uint8_t nbb = src_ptr[ (p_state->ima_adpcm[i].last_nibble>>1) * (is_stereo?2:1) + i ];
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nibble = (p_state->ima_adpcm[i].last_nibble&1)?(nbb>>4):(nbb&0xF);
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step=_ima_adpcm_step_table[p_state->ima_adpcm[i].step_index];
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p_state->ima_adpcm[i].step_index += _ima_adpcm_index_table[nibble];
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if (p_state->ima_adpcm[i].step_index<0)
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p_state->ima_adpcm[i].step_index=0;
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if (p_state->ima_adpcm[i].step_index>88)
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p_state->ima_adpcm[i].step_index=88;
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/*
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signed_nibble = (nibble&7) * ((nibble&8)?-1:1);
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diff = (2 * signed_nibble + 1) * step / 4; */
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diff = step >> 3 ;
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if (nibble & 1)
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diff += step >> 2 ;
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if (nibble & 2)
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diff += step >> 1 ;
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if (nibble & 4)
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diff += step ;
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if (nibble & 8)
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diff = -diff ;
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p_state->ima_adpcm[i].predictor+=diff;
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if (p_state->ima_adpcm[i].predictor<-0x8000)
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p_state->ima_adpcm[i].predictor=-0x8000;
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else if (p_state->ima_adpcm[i].predictor>0x7FFF)
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p_state->ima_adpcm[i].predictor=0x7FFF;
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/* store loop if there */
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if (p_state->ima_adpcm[i].last_nibble==p_state->ima_adpcm[i].loop_pos) {
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p_state->ima_adpcm[i].loop_step_index = p_state->ima_adpcm[i].step_index;
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p_state->ima_adpcm[i].loop_predictor = p_state->ima_adpcm[i].predictor;
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}
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//printf("%i - %i - pred %i\n",int(p_state->ima_adpcm[i].last_nibble),int(nibble),int(p_state->ima_adpcm[i].predictor));
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}
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}
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final=p_state->ima_adpcm[0].predictor;
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if (is_stereo) {
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final_r=p_state->ima_adpcm[1].predictor;
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}
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} else {
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final=p_src[pos];
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if (is_stereo)
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final_r=p_src[pos+1];
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if (sizeof(Depth)==1) { /* conditions will not exist anymore when compiled! */
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final<<=8;
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if (is_stereo)
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final_r<<=8;
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}
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if (type==INTERPOLATION_LINEAR) {
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if (is_stereo) {
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next=p_src[pos+2];
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next_r=p_src[pos+3];
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} else {
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next=p_src[pos+1];
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}
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if (sizeof(Depth)==1) {
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next<<=8;
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if (is_stereo)
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next_r<<=8;
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}
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int32_t frac=int32_t(p_state->pos&MIX_FRAC_MASK);
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final=final+((next-final)*frac >> MIX_FRAC_BITS);
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if (is_stereo)
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final_r=final_r+((next_r-final_r)*frac >> MIX_FRAC_BITS);
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}
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}
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if (use_filter) {
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Channel::Mix::Filter *f = p_state->filter_l;
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float finalf=final;
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float pre = finalf;
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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)
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);
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f->ha[1]=f->ha[0];
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f->hb[1]=f->hb[0];
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f->hb[0]=pre;
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f->ha[0]=finalf;
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final=Math::fast_ftoi(finalf);
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if (is_stereo) {
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f = p_state->filter_r;
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finalf=final_r;
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pre = finalf;
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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)
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);
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f->ha[1]=f->ha[0];
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f->hb[1]=f->hb[0];
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f->hb[0]=pre;
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f->ha[0]=finalf;
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final_r=Math::fast_ftoi(finalf);
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}
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p_state->coefs.b0+=p_state->coefs_inc.b0;
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p_state->coefs.b1+=p_state->coefs_inc.b1;
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p_state->coefs.b2+=p_state->coefs_inc.b2;
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p_state->coefs.a1+=p_state->coefs_inc.a1;
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p_state->coefs.a2+=p_state->coefs_inc.a2;
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}
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if (!is_stereo) {
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final_r=final; //copy to right channel if stereo
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}
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//convert back to 24 bits and mix to buffers
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if (mix_mode==MIX_STEREO) {
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*p_dst++ +=(final*(p_state->vol[0]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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*p_dst++ +=(final_r*(p_state->vol[1]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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p_state->vol[0]+=p_state->vol_inc[0];
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p_state->vol[1]+=p_state->vol_inc[1];
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if (use_fx) {
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*reverb_dst++ +=(final*(p_state->reverb_vol[0]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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*reverb_dst++ +=(final_r*(p_state->reverb_vol[1]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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p_state->reverb_vol[0]+=p_state->reverb_vol_inc[0];
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p_state->reverb_vol[1]+=p_state->reverb_vol_inc[1];
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}
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} else if (mix_mode==MIX_QUAD) {
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*p_dst++ +=(final*(p_state->vol[0]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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*p_dst++ +=(final_r*(p_state->vol[1]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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*p_dst++ +=(final*(p_state->vol[2]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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*p_dst++ +=(final_r*(p_state->vol[3]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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p_state->vol[0]+=p_state->vol_inc[0];
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p_state->vol[1]+=p_state->vol_inc[1];
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p_state->vol[2]+=p_state->vol_inc[2];
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p_state->vol[3]+=p_state->vol_inc[3];
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if (use_fx) {
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*reverb_dst++ +=(final*(p_state->reverb_vol[0]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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*reverb_dst++ +=(final_r*(p_state->reverb_vol[1]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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*reverb_dst++ +=(final*(p_state->reverb_vol[2]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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*reverb_dst++ +=(final_r*(p_state->reverb_vol[3]>>MIX_VOLRAMP_FRAC_BITS))>>MIX_VOL_MOVE_TO_24;
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p_state->reverb_vol[0]+=p_state->reverb_vol_inc[0];
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p_state->reverb_vol[1]+=p_state->reverb_vol_inc[1];
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p_state->reverb_vol[2]+=p_state->reverb_vol_inc[2];
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p_state->reverb_vol[3]+=p_state->reverb_vol_inc[3];
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}
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}
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p_state->pos+=p_state->increment;
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}
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}
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void AudioMixerSW::mix_channel(Channel& c) {
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if (!sample_manager->is_sample(c.sample)) {
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// sample is gone!
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c.active=false;
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return;
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}
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/* some 64-bit fixed point precaches */
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int64_t loop_begin_fp=((int64_t)sample_manager->sample_get_loop_begin(c.sample) << MIX_FRAC_BITS);
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int64_t loop_end_fp=((int64_t)sample_manager->sample_get_loop_end(c.sample) << MIX_FRAC_BITS);
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int64_t length_fp=((int64_t)sample_manager->sample_get_length(c.sample) << MIX_FRAC_BITS);
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int64_t begin_limit=(sample_manager->sample_get_loop_format(c.sample)!=AS::SAMPLE_LOOP_NONE)?loop_begin_fp:0;
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int64_t end_limit=(sample_manager->sample_get_loop_format(c.sample)!=AS::SAMPLE_LOOP_NONE)?loop_end_fp:length_fp;
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bool is_stereo=sample_manager->sample_is_stereo(c.sample);
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int32_t todo=mix_chunk_size;
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// int mixed=0;
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bool use_filter=false;
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ResamplerState rstate;
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/* compute voume ramps, increment, etc */
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for(int i=0;i<mix_channels;i++) {
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c.mix.old_vol[i]=c.mix.vol[i];
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c.mix.old_reverb_vol[i]=c.mix.reverb_vol[i];
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c.mix.old_chorus_vol[i]=c.mix.chorus_vol[i];
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}
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float vol = c.vol*channel_nrg;
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float reverb_vol = c.reverb_send*channel_nrg;
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float chorus_vol = c.chorus_send*channel_nrg;
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if (mix_channels==2) {
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//stereo pan
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float pan = c.pan * 0.5 + 0.5;
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float panv[2]={
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(1.0 - pan)*(1<<MIX_VOL_FRAC_BITS),
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(pan)*(1<<MIX_VOL_FRAC_BITS)
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};
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for(int i=0;i<2;i++) {
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c.mix.vol[i]=Math::fast_ftoi(vol*panv[i]);
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c.mix.reverb_vol[i]=Math::fast_ftoi(reverb_vol*panv[i]);
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c.mix.chorus_vol[i]=Math::fast_ftoi(chorus_vol*panv[i]);
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}
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} else {
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//qudra pan
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float panx = c.pan * 0.5 + 0.5;
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float pany = c.depth * 0.5 + 0.5;
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// 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
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float panv[4]={
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(1.0-pany)*(1.0-panx)*(1<<MIX_VOL_FRAC_BITS),
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(1.0-pany)*( panx)*(1<<MIX_VOL_FRAC_BITS),
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( pany)*(1.0-panx)*(1<<MIX_VOL_FRAC_BITS),
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( pany)*( panx)*(1<<MIX_VOL_FRAC_BITS)
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};
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for(int i=0;i<4;i++) {
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c.mix.vol[i]=Math::fast_ftoi(vol*panv[i]);
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c.mix.reverb_vol[i]=Math::fast_ftoi(reverb_vol*panv[i]);
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c.mix.chorus_vol[i]=Math::fast_ftoi(chorus_vol*panv[i]);
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}
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}
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if (c.first_mix) { // avoid ramp up
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for(int i=0;i<mix_channels;i++) {
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c.mix.old_vol[i]=c.mix.vol[i];
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c.mix.old_reverb_vol[i]=c.mix.reverb_vol[i];
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c.mix.old_chorus_vol[i]=c.mix.chorus_vol[i];
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}
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c.first_mix=false;
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}
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Channel::Filter::Coefs filter_coefs;
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Channel::Filter::Coefs filter_inc;
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if (c.filter.type!=AudioMixer::FILTER_NONE) {
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filter_coefs=c.filter.old_coefs;
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filter_inc.b0=(c.filter.coefs.b0-filter_coefs.b0)/(1<<mix_chunk_bits);
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filter_inc.b1=(c.filter.coefs.b1-filter_coefs.b1)/(1<<mix_chunk_bits);
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filter_inc.b2=(c.filter.coefs.b2-filter_coefs.b2)/(1<<mix_chunk_bits);
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filter_inc.a1=(c.filter.coefs.a1-filter_coefs.a1)/(1<<mix_chunk_bits);
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filter_inc.a2=(c.filter.coefs.a2-filter_coefs.a2)/(1<<mix_chunk_bits);
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use_filter=true;
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}
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if (c.mix.increment>0)
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c.mix.increment=((int64_t)c.speed<<MIX_FRAC_BITS)/mix_rate;
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else
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c.mix.increment=-((int64_t)c.speed<<MIX_FRAC_BITS)/mix_rate;
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//volume ramp
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for(int i=0;i<mix_channels;i++) {
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rstate.vol_inc[i]=((c.mix.vol[i]-c.mix.old_vol[i])<<MIX_VOLRAMP_FRAC_BITS)>>mix_chunk_bits;
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rstate.vol[i]=c.mix.old_vol[i]<<MIX_VOLRAMP_FRAC_BITS;
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rstate.reverb_vol_inc[i]=((c.mix.reverb_vol[i]-c.mix.old_reverb_vol[i])<<MIX_VOLRAMP_FRAC_BITS)>>mix_chunk_bits;
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rstate.reverb_vol[i]=c.mix.old_reverb_vol[i]<<MIX_VOLRAMP_FRAC_BITS;
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rstate.chorus_vol_inc[i]=((c.mix.chorus_vol[i]-c.mix.old_chorus_vol[i])<<MIX_VOLRAMP_FRAC_BITS)>>mix_chunk_bits;
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rstate.chorus_vol[i]=c.mix.old_chorus_vol[i]<<MIX_VOLRAMP_FRAC_BITS;
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}
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//looping
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AS::SampleLoopFormat loop_format=sample_manager->sample_get_loop_format(c.sample);
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AS::SampleFormat format=sample_manager->sample_get_format(c.sample);
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bool use_fx=false;
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if (fx_enabled) {
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for(int i=0;i<mix_channels;i++) {
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if (c.mix.old_reverb_vol[i] || c.mix.reverb_vol[i] || c.mix.old_chorus_vol[i] || c.mix.chorus_vol[i] ) {
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use_fx=true;
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break;
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}
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}
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}
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/* audio data */
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const void *data=sample_manager->sample_get_data_ptr(c.sample);
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int32_t *dst_buff=mix_buffer;
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#ifndef NO_REVERB
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rstate.reverb_buffer=reverb_state[c.reverb_room].buffer;
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#endif
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/* @TODO validar loops al registrar? */
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rstate.coefs=filter_coefs;
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rstate.coefs_inc=filter_inc;
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rstate.filter_l=&c.mix.filter_l;
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rstate.filter_r=&c.mix.filter_r;
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if (format==AS::SAMPLE_FORMAT_IMA_ADPCM) {
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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
|
|
|
|
|
|
}
|