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b324ff7ea5
-IMA-ADPCM support for samples, this means that sound effects can be compressed and use 4 timess less RAM. -New 3D import workflow based on Wavefront OBJ. Import single objects as mesh resources instead of full scenes. Many people prefers to work this way. Just like the rest of the imported resources, these are updated in realtime if modified externally. -Mesh resources now support naming surfaces. This helps reimporting to identify which user-created materials must be kept. -Several fixes and improvements to SurfaceTool. -Anti Aliasing added to WorldEnvironment effects (using FXAA) -2D Physics bodies (RigidBody, KinematicBody, etc), Raycasts, Tilemap, etc support collision layers. This makes easy to group which objects collide against which. -2D Trigger shapes can now also trigger collision reporting in other 2D bodies (it used to be in Area2D before) -Viewport render target textures can now be filtered. -Few fixes in GDscript make it easier to work with static functions and class members. -Several and many bugfixes.
1194 lines
32 KiB
C++
1194 lines
32 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) 2007-2014 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)
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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->window_ofs;
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while(sample_pos>p_state->ima_adpcm->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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int16_t nibble,signed_nibble,diff,step;
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p_state->ima_adpcm->last_nibble++;
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const uint8_t *src_ptr=p_state->ima_adpcm->ptr;
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nibble = (p_state->ima_adpcm->last_nibble&1)?
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(src_ptr[p_state->ima_adpcm->last_nibble>>1]>>4):(src_ptr[p_state->ima_adpcm->last_nibble>>1]&0xF);
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step=_ima_adpcm_step_table[p_state->ima_adpcm->step_index];
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p_state->ima_adpcm->step_index += _ima_adpcm_index_table[nibble];
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if (p_state->ima_adpcm->step_index<0)
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p_state->ima_adpcm->step_index=0;
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if (p_state->ima_adpcm->step_index>88)
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p_state->ima_adpcm->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->predictor+=diff;
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if (p_state->ima_adpcm->predictor<-0x8000)
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p_state->ima_adpcm->predictor=-0x8000;
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else if (p_state->ima_adpcm->predictor>0x7FFF)
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p_state->ima_adpcm->predictor=0x7FFF;
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/* store loop if there */
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if (p_state->ima_adpcm->last_nibble==p_state->ima_adpcm->loop_pos) {
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p_state->ima_adpcm->loop_step_index = p_state->ima_adpcm->step_index;
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p_state->ima_adpcm->loop_predictor = p_state->ima_adpcm->predictor;
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}
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}
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final=p_state->ima_adpcm->predictor;
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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=fx_enabled && (c.mix.old_reverb_vol || c.mix.reverb_vol || c.mix.old_chorus_vol || c.mix.chorus_vol );
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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;
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if (loop_format!=AS::SAMPLE_LOOP_NONE) {
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c.mix.ima_adpcm.loop_pos=loop_begin_fp>>MIX_FRAC_BITS;
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loop_format=AS::SAMPLE_LOOP_FORWARD;
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}
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}
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|
|
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) {
|
|
c.mix.ima_adpcm.step_index=c.mix.ima_adpcm.loop_step_index;
|
|
c.mix.ima_adpcm.predictor=c.mix.ima_adpcm.loop_predictor;
|
|
c.mix.ima_adpcm.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) {
|
|
c.mix.ima_adpcm.window_ofs=c.mix.offset>>MIX_FRAC_BITS;
|
|
c.mix.ima_adpcm.ptr=(const uint8_t*)data;
|
|
int8_t *src_ptr = &((int8_t*)data)[(c.mix.offset >> MIX_FRAC_BITS)<<(is_stereo?1:0) ];
|
|
CALL_RESAMPLE_MODE(int8_t,false,true,use_filter,use_fx,interpolation_type,mix_channels);
|
|
|
|
}
|
|
|
|
c.mix.offset+=rstate.pos;
|
|
dst_buff+=target*2;
|
|
|
|
}
|
|
|
|
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.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) {
|
|
|
|
c.mix.ima_adpcm.step_index=0;
|
|
c.mix.ima_adpcm.predictor=0;
|
|
c.mix.ima_adpcm.loop_step_index=0;
|
|
c.mix.ima_adpcm.loop_predictor=0;
|
|
c.mix.ima_adpcm.last_nibble=-1;
|
|
c.mix.ima_adpcm.loop_pos=0x7FFFFFFF;
|
|
c.mix.ima_adpcm.window_ofs=0;
|
|
c.mix.ima_adpcm.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
|
|
|
|
|
|
}
|