godot/scene/resources/audio_stream_resampled.cpp
Rémi Verschelde 5dbf1809c6 A Whole New World (clang-format edition)
I can show you the code
Pretty, with proper whitespace
Tell me, coder, now when did
You last write readable code?

I can open your eyes
Make you see your bad indent
Force you to respect the style
The core devs agreed upon

A whole new world
A new fantastic code format
A de facto standard
With some sugar
Enforced with clang-format

A whole new world
A dazzling style we all dreamed of
And when we read it through
It's crystal clear
That now we're in a whole new world of code
2017-03-05 16:44:50 +01:00

388 lines
9.0 KiB
C++

/*************************************************************************/
/* audio_stream_resampled.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* http://www.godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2017 Juan Linietsky, Ariel Manzur. */
/* */
/* Permission is hereby granted, free of charge, to any person obtaining */
/* a copy of this software and associated documentation files (the */
/* "Software"), to deal in the Software without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of the Software, and to */
/* permit persons to whom the Software is furnished to do so, subject to */
/* the following conditions: */
/* */
/* The above copyright notice and this permission notice shall be */
/* included in all copies or substantial portions of the Software. */
/* */
/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#include "audio_stream_resampled.h"
#include "global_config.h"
#if 0
int AudioStreamResampled::get_channel_count() const {
if (!rb)
return 0;
return channels;
}
template<int C>
uint32_t AudioStreamResampled::_resample(int32_t *p_dest,int p_todo,int32_t p_increment) {
uint32_t read=offset&MIX_FRAC_MASK;
for (int i=0;i<p_todo;i++) {
offset = (offset + p_increment)&(((1<<(rb_bits+MIX_FRAC_BITS))-1));
read+=p_increment;
uint32_t pos = offset >> MIX_FRAC_BITS;
uint32_t frac = offset & MIX_FRAC_MASK;
#ifndef FAST_AUDIO
ERR_FAIL_COND_V(pos>=rb_len,0);
#endif
uint32_t pos_next = (pos+1)&rb_mask;
//printf("rb pos %i\n",pos);
// since this is a template with a known compile time value (C), conditionals go away when compiling.
if (C==1) {
int32_t v0 = rb[pos];
int32_t v0n=rb[pos_next];
#ifndef FAST_AUDIO
v0+=(v0n-v0)*(int32_t)frac >> MIX_FRAC_BITS;
#endif
v0<<=16;
p_dest[i]=v0;
}
if (C==2) {
int32_t v0 = rb[(pos<<1)+0];
int32_t v1 = rb[(pos<<1)+1];
int32_t v0n=rb[(pos_next<<1)+0];
int32_t v1n=rb[(pos_next<<1)+1];
#ifndef FAST_AUDIO
v0+=(v0n-v0)*(int32_t)frac >> MIX_FRAC_BITS;
v1+=(v1n-v1)*(int32_t)frac >> MIX_FRAC_BITS;
#endif
v0<<=16;
v1<<=16;
p_dest[(i<<1)+0]=v0;
p_dest[(i<<1)+1]=v1;
}
if (C==4) {
int32_t v0 = rb[(pos<<2)+0];
int32_t v1 = rb[(pos<<2)+1];
int32_t v2 = rb[(pos<<2)+2];
int32_t v3 = rb[(pos<<2)+3];
int32_t v0n = rb[(pos_next<<2)+0];
int32_t v1n=rb[(pos_next<<2)+1];
int32_t v2n=rb[(pos_next<<2)+2];
int32_t v3n=rb[(pos_next<<2)+3];
#ifndef FAST_AUDIO
v0+=(v0n-v0)*(int32_t)frac >> MIX_FRAC_BITS;
v1+=(v1n-v1)*(int32_t)frac >> MIX_FRAC_BITS;
v2+=(v2n-v2)*(int32_t)frac >> MIX_FRAC_BITS;
v3+=(v3n-v3)*(int32_t)frac >> MIX_FRAC_BITS;
#endif
v0<<=16;
v1<<=16;
v2<<=16;
v3<<=16;
p_dest[(i<<2)+0]=v0;
p_dest[(i<<2)+1]=v1;
p_dest[(i<<2)+2]=v2;
p_dest[(i<<2)+3]=v3;
}
if (C==6) {
int32_t v0 = rb[(pos*6)+0];
int32_t v1 = rb[(pos*6)+1];
int32_t v2 = rb[(pos*6)+2];
int32_t v3 = rb[(pos*6)+3];
int32_t v4 = rb[(pos*6)+4];
int32_t v5 = rb[(pos*6)+5];
int32_t v0n = rb[(pos_next*6)+0];
int32_t v1n=rb[(pos_next*6)+1];
int32_t v2n=rb[(pos_next*6)+2];
int32_t v3n=rb[(pos_next*6)+3];
int32_t v4n=rb[(pos_next*6)+4];
int32_t v5n=rb[(pos_next*6)+5];
#ifndef FAST_AUDIO
v0+=(v0n-v0)*(int32_t)frac >> MIX_FRAC_BITS;
v1+=(v1n-v1)*(int32_t)frac >> MIX_FRAC_BITS;
v2+=(v2n-v2)*(int32_t)frac >> MIX_FRAC_BITS;
v3+=(v3n-v3)*(int32_t)frac >> MIX_FRAC_BITS;
v4+=(v4n-v4)*(int32_t)frac >> MIX_FRAC_BITS;
v5+=(v5n-v5)*(int32_t)frac >> MIX_FRAC_BITS;
#endif
v0<<=16;
v1<<=16;
v2<<=16;
v3<<=16;
v4<<=16;
v5<<=16;
p_dest[(i*6)+0]=v0;
p_dest[(i*6)+1]=v1;
p_dest[(i*6)+2]=v2;
p_dest[(i*6)+3]=v3;
p_dest[(i*6)+4]=v4;
p_dest[(i*6)+5]=v5;
}
}
return read>>MIX_FRAC_BITS;//rb_read_pos=offset>>MIX_FRAC_BITS;
}
bool AudioStreamResampled::mix(int32_t *p_dest, int p_frames) {
if (!rb || !_can_mix())
return false;
int write_pos_cache=rb_write_pos;
int32_t increment=(mix_rate*MIX_FRAC_LEN)/get_mix_rate();
int rb_todo;
if (write_pos_cache==rb_read_pos) {
return false; //out of buffer
} else if (rb_read_pos<write_pos_cache) {
rb_todo=write_pos_cache-rb_read_pos; //-1?
} else {
rb_todo=(rb_len-rb_read_pos)+write_pos_cache; //-1?
}
int todo = MIN( ((int64_t(rb_todo)<<MIX_FRAC_BITS)/increment)+1, p_frames );
#if 0
if (int(mix_rate)==get_mix_rate()) {
if (channels==6) {
for(int i=0;i<p_frames;i++) {
int from = ((rb_read_pos+i)&rb_mask)*6;
int to = i*6;
p_dest[from+0]=int32_t(rb[to+0])<<16;
p_dest[from+1]=int32_t(rb[to+1])<<16;
p_dest[from+2]=int32_t(rb[to+2])<<16;
p_dest[from+3]=int32_t(rb[to+3])<<16;
p_dest[from+4]=int32_t(rb[to+4])<<16;
p_dest[from+5]=int32_t(rb[to+5])<<16;
}
} else {
int len=p_frames*channels;
int from=rb_read_pos*channels;
int mask=0;
switch(channels) {
case 1: mask=rb_len-1; break;
case 2: mask=(rb_len*2)-1; break;
case 4: mask=(rb_len*4)-1; break;
}
for(int i=0;i<len;i++) {
p_dest[i]=int32_t(rb[(from+i)&mask])<<16;
}
}
rb_read_pos = (rb_read_pos+p_frames)&rb_mask;
} else
#endif
{
uint32_t read=0;
switch(channels) {
case 1: read=_resample<1>(p_dest,todo,increment); break;
case 2: read=_resample<2>(p_dest,todo,increment); break;
case 4: read=_resample<4>(p_dest,todo,increment); break;
case 6: read=_resample<6>(p_dest,todo,increment); break;
}
#if 1
//end of stream, fadeout
int remaining = p_frames-todo;
if (remaining && todo>0) {
//print_line("fadeout");
for(int c=0;c<channels;c++) {
for(int i=0;i<todo;i++) {
int32_t samp = p_dest[i*channels+c]>>8;
uint32_t mul = (todo-i) * 256 /todo;
//print_line("mul: "+itos(i)+" "+itos(mul));
p_dest[i*channels+c]=samp*mul;
}
}
}
#else
int remaining = p_frames-todo;
if (remaining && todo>0) {
for(int c=0;c<channels;c++) {
int32_t from = p_dest[(todo-1)*channels+c]>>8;
for(int i=0;i<remaining;i++) {
uint32_t mul = (remaining-i) * 256 /remaining;
p_dest[(todo+i)*channels+c]=from*mul;
}
}
}
#endif
//zero out what remains there to avoid glitches
for(int i=todo*channels;i<int(p_frames)*channels;i++) {
p_dest[i]=0;
}
if (read>rb_todo)
read=rb_todo;
rb_read_pos = (rb_read_pos+read)&rb_mask;
}
return true;
}
Error AudioStreamResampled::_setup(int p_channels,int p_mix_rate,int p_minbuff_needed) {
ERR_FAIL_COND_V(p_channels!=1 && p_channels!=2 && p_channels!=4 && p_channels!=6,ERR_INVALID_PARAMETER);
float buffering_sec = int(GLOBAL_DEF("audio/stream_buffering_ms",500))/1000.0;
int desired_rb_bits =nearest_shift(MAX(buffering_sec*p_mix_rate,p_minbuff_needed));
bool recreate=!rb;
if (rb && (uint32_t(desired_rb_bits)!=rb_bits || channels!=uint32_t(p_channels))) {
//recreate
memdelete_arr(rb);
memdelete_arr(read_buf);
recreate=true;
}
if (recreate) {
channels=p_channels;
rb_bits=desired_rb_bits;
rb_len=(1<<rb_bits);
rb_mask=rb_len-1;
rb = memnew_arr( int16_t, rb_len * p_channels );
read_buf = memnew_arr( int16_t, rb_len * p_channels );
}
mix_rate=p_mix_rate;
offset=0;
rb_read_pos=0;
rb_write_pos=0;
//avoid maybe strange noises upon load
for (int i=0;i<(rb_len*channels);i++) {
rb[i]=0;
read_buf[i]=0;
}
return OK;
}
void AudioStreamResampled::_clear() {
if (!rb)
return;
AudioServer::get_singleton()->lock();
//should be stopped at this point but just in case
if (rb) {
memdelete_arr(rb);
memdelete_arr(read_buf);
}
rb=NULL;
offset=0;
rb_read_pos=0;
rb_write_pos=0;
read_buf=NULL;
AudioServer::get_singleton()->unlock();
}
AudioStreamResampled::AudioStreamResampled() {
rb=NULL;
offset=0;
read_buf=NULL;
rb_read_pos=0;
rb_write_pos=0;
rb_bits=0;
rb_len=0;
rb_mask=0;
read_buff_len=0;
channels=0;
mix_rate=0;
}
AudioStreamResampled::~AudioStreamResampled() {
if (rb) {
memdelete_arr(rb);
memdelete_arr(read_buf);
}
}
#endif