d8223ffa75
That year should bring the long-awaited OpenGL ES 3.0 compatible renderer
with state-of-the-art rendering techniques tuned to work as low as middle
end handheld devices - without compromising with the possibilities given
for higher end desktop games of course. Great times ahead for the Godot
community and the gamers that will play our games!
(cherry picked from commit c7bc44d5ad)
389 lines
9 KiB
C++
389 lines
9 KiB
C++
/*************************************************************************/
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/* audio_stream_resampled.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) 2007-2017 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_stream_resampled.h"
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#include "globals.h"
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#if 0
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int AudioStreamResampled::get_channel_count() const {
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if (!rb)
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return 0;
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return channels;
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}
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template<int C>
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uint32_t AudioStreamResampled::_resample(int32_t *p_dest,int p_todo,int32_t p_increment) {
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uint32_t read=offset&MIX_FRAC_MASK;
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for (int i=0;i<p_todo;i++) {
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offset = (offset + p_increment)&(((1<<(rb_bits+MIX_FRAC_BITS))-1));
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read+=p_increment;
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uint32_t pos = offset >> MIX_FRAC_BITS;
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uint32_t frac = offset & MIX_FRAC_MASK;
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#ifndef FAST_AUDIO
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ERR_FAIL_COND_V(pos>=rb_len,0);
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#endif
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uint32_t pos_next = (pos+1)&rb_mask;
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//printf("rb pos %i\n",pos);
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// since this is a template with a known compile time value (C), conditionals go away when compiling.
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if (C==1) {
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int32_t v0 = rb[pos];
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int32_t v0n=rb[pos_next];
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#ifndef FAST_AUDIO
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v0+=(v0n-v0)*(int32_t)frac >> MIX_FRAC_BITS;
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#endif
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v0<<=16;
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p_dest[i]=v0;
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}
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if (C==2) {
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int32_t v0 = rb[(pos<<1)+0];
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int32_t v1 = rb[(pos<<1)+1];
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int32_t v0n=rb[(pos_next<<1)+0];
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int32_t v1n=rb[(pos_next<<1)+1];
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#ifndef FAST_AUDIO
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v0+=(v0n-v0)*(int32_t)frac >> MIX_FRAC_BITS;
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v1+=(v1n-v1)*(int32_t)frac >> MIX_FRAC_BITS;
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#endif
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v0<<=16;
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v1<<=16;
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p_dest[(i<<1)+0]=v0;
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p_dest[(i<<1)+1]=v1;
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}
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if (C==4) {
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int32_t v0 = rb[(pos<<2)+0];
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int32_t v1 = rb[(pos<<2)+1];
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int32_t v2 = rb[(pos<<2)+2];
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int32_t v3 = rb[(pos<<2)+3];
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int32_t v0n = rb[(pos_next<<2)+0];
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int32_t v1n=rb[(pos_next<<2)+1];
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int32_t v2n=rb[(pos_next<<2)+2];
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int32_t v3n=rb[(pos_next<<2)+3];
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#ifndef FAST_AUDIO
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v0+=(v0n-v0)*(int32_t)frac >> MIX_FRAC_BITS;
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v1+=(v1n-v1)*(int32_t)frac >> MIX_FRAC_BITS;
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v2+=(v2n-v2)*(int32_t)frac >> MIX_FRAC_BITS;
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v3+=(v3n-v3)*(int32_t)frac >> MIX_FRAC_BITS;
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#endif
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v0<<=16;
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v1<<=16;
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v2<<=16;
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v3<<=16;
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p_dest[(i<<2)+0]=v0;
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p_dest[(i<<2)+1]=v1;
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p_dest[(i<<2)+2]=v2;
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p_dest[(i<<2)+3]=v3;
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}
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if (C==6) {
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int32_t v0 = rb[(pos*6)+0];
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int32_t v1 = rb[(pos*6)+1];
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int32_t v2 = rb[(pos*6)+2];
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int32_t v3 = rb[(pos*6)+3];
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int32_t v4 = rb[(pos*6)+4];
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int32_t v5 = rb[(pos*6)+5];
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int32_t v0n = rb[(pos_next*6)+0];
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int32_t v1n=rb[(pos_next*6)+1];
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int32_t v2n=rb[(pos_next*6)+2];
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int32_t v3n=rb[(pos_next*6)+3];
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int32_t v4n=rb[(pos_next*6)+4];
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int32_t v5n=rb[(pos_next*6)+5];
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#ifndef FAST_AUDIO
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v0+=(v0n-v0)*(int32_t)frac >> MIX_FRAC_BITS;
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v1+=(v1n-v1)*(int32_t)frac >> MIX_FRAC_BITS;
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v2+=(v2n-v2)*(int32_t)frac >> MIX_FRAC_BITS;
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v3+=(v3n-v3)*(int32_t)frac >> MIX_FRAC_BITS;
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v4+=(v4n-v4)*(int32_t)frac >> MIX_FRAC_BITS;
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v5+=(v5n-v5)*(int32_t)frac >> MIX_FRAC_BITS;
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#endif
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v0<<=16;
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v1<<=16;
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v2<<=16;
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v3<<=16;
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v4<<=16;
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v5<<=16;
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p_dest[(i*6)+0]=v0;
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p_dest[(i*6)+1]=v1;
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p_dest[(i*6)+2]=v2;
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p_dest[(i*6)+3]=v3;
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p_dest[(i*6)+4]=v4;
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p_dest[(i*6)+5]=v5;
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}
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}
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return read>>MIX_FRAC_BITS;//rb_read_pos=offset>>MIX_FRAC_BITS;
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}
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bool AudioStreamResampled::mix(int32_t *p_dest, int p_frames) {
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if (!rb || !_can_mix())
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return false;
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int write_pos_cache=rb_write_pos;
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int32_t increment=(mix_rate*MIX_FRAC_LEN)/get_mix_rate();
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int rb_todo;
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if (write_pos_cache==rb_read_pos) {
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return false; //out of buffer
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} else if (rb_read_pos<write_pos_cache) {
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rb_todo=write_pos_cache-rb_read_pos; //-1?
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} else {
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rb_todo=(rb_len-rb_read_pos)+write_pos_cache; //-1?
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}
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int todo = MIN( ((int64_t(rb_todo)<<MIX_FRAC_BITS)/increment)+1, p_frames );
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#if 0
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if (int(mix_rate)==get_mix_rate()) {
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if (channels==6) {
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for(int i=0;i<p_frames;i++) {
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int from = ((rb_read_pos+i)&rb_mask)*6;
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int to = i*6;
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p_dest[from+0]=int32_t(rb[to+0])<<16;
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p_dest[from+1]=int32_t(rb[to+1])<<16;
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p_dest[from+2]=int32_t(rb[to+2])<<16;
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p_dest[from+3]=int32_t(rb[to+3])<<16;
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p_dest[from+4]=int32_t(rb[to+4])<<16;
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p_dest[from+5]=int32_t(rb[to+5])<<16;
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}
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} else {
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int len=p_frames*channels;
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int from=rb_read_pos*channels;
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int mask=0;
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switch(channels) {
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case 1: mask=rb_len-1; break;
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case 2: mask=(rb_len*2)-1; break;
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case 4: mask=(rb_len*4)-1; break;
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}
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for(int i=0;i<len;i++) {
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p_dest[i]=int32_t(rb[(from+i)&mask])<<16;
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}
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}
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rb_read_pos = (rb_read_pos+p_frames)&rb_mask;
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} else
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#endif
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{
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uint32_t read=0;
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switch(channels) {
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case 1: read=_resample<1>(p_dest,todo,increment); break;
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case 2: read=_resample<2>(p_dest,todo,increment); break;
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case 4: read=_resample<4>(p_dest,todo,increment); break;
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case 6: read=_resample<6>(p_dest,todo,increment); break;
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}
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#if 1
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//end of stream, fadeout
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int remaining = p_frames-todo;
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if (remaining && todo>0) {
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//print_line("fadeout");
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for(int c=0;c<channels;c++) {
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for(int i=0;i<todo;i++) {
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int32_t samp = p_dest[i*channels+c]>>8;
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uint32_t mul = (todo-i) * 256 /todo;
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//print_line("mul: "+itos(i)+" "+itos(mul));
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p_dest[i*channels+c]=samp*mul;
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}
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}
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}
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#else
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int remaining = p_frames-todo;
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if (remaining && todo>0) {
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for(int c=0;c<channels;c++) {
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int32_t from = p_dest[(todo-1)*channels+c]>>8;
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for(int i=0;i<remaining;i++) {
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uint32_t mul = (remaining-i) * 256 /remaining;
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p_dest[(todo+i)*channels+c]=from*mul;
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}
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}
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}
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#endif
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//zero out what remains there to avoid glitches
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for(int i=todo*channels;i<int(p_frames)*channels;i++) {
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p_dest[i]=0;
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}
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if (read>rb_todo)
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read=rb_todo;
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rb_read_pos = (rb_read_pos+read)&rb_mask;
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}
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return true;
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}
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Error AudioStreamResampled::_setup(int p_channels,int p_mix_rate,int p_minbuff_needed) {
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ERR_FAIL_COND_V(p_channels!=1 && p_channels!=2 && p_channels!=4 && p_channels!=6,ERR_INVALID_PARAMETER);
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float buffering_sec = int(GLOBAL_DEF("audio/stream_buffering_ms",500))/1000.0;
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int desired_rb_bits =nearest_shift(MAX(buffering_sec*p_mix_rate,p_minbuff_needed));
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bool recreate=!rb;
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if (rb && (uint32_t(desired_rb_bits)!=rb_bits || channels!=uint32_t(p_channels))) {
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//recreate
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memdelete_arr(rb);
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memdelete_arr(read_buf);
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recreate=true;
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}
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if (recreate) {
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channels=p_channels;
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rb_bits=desired_rb_bits;
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rb_len=(1<<rb_bits);
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rb_mask=rb_len-1;
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rb = memnew_arr( int16_t, rb_len * p_channels );
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read_buf = memnew_arr( int16_t, rb_len * p_channels );
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}
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mix_rate=p_mix_rate;
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offset=0;
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rb_read_pos=0;
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rb_write_pos=0;
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//avoid maybe strange noises upon load
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for (int i=0;i<(rb_len*channels);i++) {
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rb[i]=0;
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read_buf[i]=0;
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}
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return OK;
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}
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void AudioStreamResampled::_clear() {
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if (!rb)
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return;
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AudioServer::get_singleton()->lock();
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//should be stopped at this point but just in case
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if (rb) {
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memdelete_arr(rb);
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memdelete_arr(read_buf);
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}
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rb=NULL;
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offset=0;
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rb_read_pos=0;
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rb_write_pos=0;
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read_buf=NULL;
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AudioServer::get_singleton()->unlock();
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}
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AudioStreamResampled::AudioStreamResampled() {
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rb=NULL;
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offset=0;
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read_buf=NULL;
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rb_read_pos=0;
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rb_write_pos=0;
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rb_bits=0;
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rb_len=0;
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rb_mask=0;
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read_buff_len=0;
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channels=0;
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mix_rate=0;
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}
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AudioStreamResampled::~AudioStreamResampled() {
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if (rb) {
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memdelete_arr(rb);
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memdelete_arr(read_buf);
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}
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}
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#endif
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