mirror of
https://github.com/godotengine/godot.git
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31ce3c5fd0
-fix some menus -fixed bug in out transition curves -detect and remove file:/// in collada -remove multiscript for now -remove dependencies on mouse in OS, moved to Input -avoid fscache from screwing up (fix might make it slower, but it works) -funcref was missing, it's there now
318 lines
7.9 KiB
C++
318 lines
7.9 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) 2007-2014 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_stream_resampled.h"
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#include "globals.h"
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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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void AudioStreamResampled::_resample(int32_t *p_dest,int p_todo,int32_t p_increment) {
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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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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(pos>=rb_len);
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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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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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switch(channels) {
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case 1: _resample<1>(p_dest,todo,increment); break;
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case 2: _resample<2>(p_dest,todo,increment); break;
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case 4: _resample<4>(p_dest,todo,increment); break;
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case 6: _resample<6>(p_dest,todo,increment); break;
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}
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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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}
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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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