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FFMpeg plugin: now build against the optional package for 0.10

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* added optional feature package for ffmpeg 0.10 gcc4 and gcc2
* increased probe buffer size to 8K for now.
* built the optional packages the default way, it might need adjustments
  like dependencies on speex, ogg, theora, vorbis, which I left out for now.
This commit is contained in:
Jérôme Duval
2012-03-07 02:05:50 +01:00
parent 08601a8488
commit e40d995cf4
1041 changed files with 65 additions and 427218 deletions
Download Patch File Download Diff File Expand all files Collapse all files
build/jam/OptionalBuildFeatures
+36
View File
@@ -211,6 +211,42 @@ if $(TARGET_ARCH) = x86 {
}
# FFmpeg
local ffmpegBaseURL = http://haiku-files.org/files/optional-packages ;
if $(TARGET_ARCH) = x86 {
if $(HAIKU_GCC_VERSION[1]) >= 4 {
HAIKU_FFMPEG_FILE = ffmpeg-0.10-x86-gcc4-2012-02-19.zip ;
} else {
HAIKU_FFMPEG_FILE = ffmpeg-0.10-x86-gcc2-2012-03-07.zip ;
}
local zipFile = [ DownloadFile $(HAIKU_FFMPEG_FILE)
: $(ffmpegBaseURL)/$(HAIKU_FFMPEG_FILE) ] ;
HAIKU_FFMPEG_DIR = [ FDirName $(HAIKU_OPTIONAL_BUILD_PACKAGES_DIR)
$(HAIKU_FFMPEG_FILE:B) ] ;
HAIKU_FFMPEG_HEADERS_DEPENDENCY = [ ExtractArchive $(HAIKU_FFMPEG_DIR)
: common/include/ : $(zipFile) : extracted-ffmpeg ] ;
HAIKU_FFMPEG_LIBS = [ ExtractArchive $(HAIKU_FFMPEG_DIR)
:
common/lib/libavformat.a
common/lib/libavcodec.a
common/lib/libavdevice.a
common/lib/libavfilter.a
common/lib/libswscale.a
common/lib/libavutil.a
: $(zipFile)
: extracted-ffmpeg ] ;
HAIKU_FFMPEG_HEADERS = [ FDirName $(HAIKU_FFMPEG_DIR) common include ] ;
} else {
Echo "FFMpeg support not available on $(TARGET_ARCH)" ;
}
# TagLib
# Automatically install the TagLib feature, when the optional TagLib optional
src/add-ons/media/plugins/ffmpeg/AVFormatReader.cpp
+5 -5
View File
@@ -255,7 +255,7 @@ StreamBase::Open()
if (buffer == NULL)
return B_NO_MEMORY;
size_t probeSize = 2048;
size_t probeSize = 8192;
AVProbeData probeData;
probeData.filename = "";
probeData.buf = buffer;
@@ -417,10 +417,10 @@ StreamBase::FrameRate() const
// TODO: Find a way to always calculate a correct frame rate...
double frameRate = 1.0;
switch (fStream->codec->codec_type) {
case CODEC_TYPE_AUDIO:
case AVMEDIA_TYPE_AUDIO:
frameRate = (double)fStream->codec->sample_rate;
break;
case CODEC_TYPE_VIDEO:
case AVMEDIA_TYPE_VIDEO:
if (fStream->avg_frame_rate.den && fStream->avg_frame_rate.num)
frameRate = av_q2d(fStream->avg_frame_rate);
else if (fStream->r_frame_rate.den && fStream->r_frame_rate.num)
@@ -734,7 +734,7 @@ StreamBase::GetNextChunk(const void** chunkBuffer,
break;
case B_MEDIA_ENCODED_AUDIO:
mediaHeader->u.encoded_audio.buffer_flags
= (fPacket.flags & PKT_FLAG_KEY) ? B_MEDIA_KEY_FRAME : 0;
= (fPacket.flags & AV_PKT_FLAG_KEY) ? B_MEDIA_KEY_FRAME : 0;
break;
case B_MEDIA_RAW_VIDEO:
mediaHeader->u.raw_video.line_count
@@ -742,7 +742,7 @@ StreamBase::GetNextChunk(const void** chunkBuffer,
break;
case B_MEDIA_ENCODED_VIDEO:
mediaHeader->u.encoded_video.field_flags
= (fPacket.flags & PKT_FLAG_KEY) ? B_MEDIA_KEY_FRAME : 0;
= (fPacket.flags & AV_PKT_FLAG_KEY) ? B_MEDIA_KEY_FRAME : 0;
mediaHeader->u.encoded_video.line_count
= fFormat.u.encoded_video.output.display.line_count;
break;
src/add-ons/media/plugins/ffmpeg/CodecTable.cpp
-1
View File
@@ -8,7 +8,6 @@
extern "C" {
#include "avcodec.h"
#include "libavformat/internal.h"
#include "avformat.h"
}
src/add-ons/media/plugins/ffmpeg/Jamfile
+24 -18
View File
@@ -4,15 +4,10 @@ SetSubDirSupportedPlatformsBeOSCompatible ;
UsePrivateHeaders media shared ;
SubDirHdrs [ FDirName $(SUBDIR) libavcodec ] ;
SubDirHdrs [ FDirName $(SUBDIR) libavformat ] ;
SubDirHdrs [ FDirName $(SUBDIR) libavutil ] ;
SubDirHdrs [ FDirName $(SUBDIR) libswscale ] ;
SubDirC++Flags -D__STDC_CONSTANT_MACROS -Wno-deprecated-declarations ;
SubDirC++Flags -D__STDC_CONSTANT_MACROS ;
Addon ffmpeg :
AVCodecDecoder.cpp
local sources =
AVCodecDecoder.cpp
AVCodecEncoder.cpp
AVFormatReader.cpp
AVFormatWriter.cpp
@@ -22,6 +17,22 @@ Addon ffmpeg :
FFmpegPlugin.cpp
MuxerTable.cpp
CpuCapabilities.cpp
;
SubDirSysHdrs $(HAIKU_FFMPEG_HEADERS) ;
Includes [ FGristFiles $(sources) ] : $(HAIKU_FFMPEG_HEADERS_DEPENDENCY) ;
UseHeaders [ FDirName $(HAIKU_FFMPEG_HEADERS) ] ;
UseHeaders [ FDirName $(HAIKU_FFMPEG_HEADERS) libavcodec ] ;
UseHeaders [ FDirName $(HAIKU_FFMPEG_HEADERS) libavdevice ] ;
UseHeaders [ FDirName $(HAIKU_FFMPEG_HEADERS) libavfilter ] ;
UseHeaders [ FDirName $(HAIKU_FFMPEG_HEADERS) libavformat ] ;
UseHeaders [ FDirName $(HAIKU_FFMPEG_HEADERS) libavutil ] ;
UseHeaders [ FDirName $(HAIKU_FFMPEG_HEADERS) libswscale ] ;
Addon ffmpeg :
$(sources)
gfx_conv_c.cpp
gfx_conv_c_lookup.cpp
@@ -31,11 +42,9 @@ Addon ffmpeg :
yuvrgb_sse2.nasm
yuvrgb_ssse3.nasm
:
libavformat.a
libavcodec.a
libavcodec_$(TARGET_ARCH).a
libavutil.a
libswscale.a
# FFmpeg libraries (from FFmpeg optional package):
$(HAIKU_FFMPEG_LIBS)
libvorbis_ffmpeg.a
libspeex_ffmpeg.a
@@ -44,9 +53,10 @@ Addon ffmpeg :
libbz2_ffmpeg.a
be
libmedia.so
media
libz.a
$(TARGET_LIBSUPC++)
$(TARGET_NETWORK_LIBS)
;
# FFmpeg libs configuration is in the config.h header. Architecture specific
@@ -112,12 +122,8 @@ if $(HAIKU_INCLUDE_GPL_ADDONS) = 1 {
HAIKU_FFMPEG_DEFINES = [ FDefines $(HAIKU_FFMPEG_DEFINES) ] ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libavcodec ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libavformat ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libavutil ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libbz2 ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libogg ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libswscale ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libspeex ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libtheora ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libvorbis ;
src/add-ons/media/plugins/ffmpeg/libavcodec/4xm.c
-855
View File
@@ -1,855 +0,0 @@
/*
* 4XM codec
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* 4XM codec.
*/
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "dsputil.h"
#include "get_bits.h"
#include "bytestream.h"
//#undef NDEBUG
//#include <assert.h>
#define BLOCK_TYPE_VLC_BITS 5
#define ACDC_VLC_BITS 9
#define CFRAME_BUFFER_COUNT 100
static const uint8_t block_type_tab[2][4][8][2]={
{
{ //{8,4,2}x{8,4,2}
{ 0,1}, { 2,2}, { 6,3}, {14,4}, {30,5}, {31,5}, { 0,0}
},{ //{8,4}x1
{ 0,1}, { 0,0}, { 2,2}, { 6,3}, {14,4}, {15,4}, { 0,0}
},{ //1x{8,4}
{ 0,1}, { 2,2}, { 0,0}, { 6,3}, {14,4}, {15,4}, { 0,0}
},{ //1x2, 2x1
{ 0,1}, { 0,0}, { 0,0}, { 2,2}, { 6,3}, {14,4}, {15,4}
}
},{
{ //{8,4,2}x{8,4,2}
{ 1,2}, { 4,3}, { 5,3}, {0,2}, {6,3}, {7,3}, {0,0}
},{//{8,4}x1
{ 1,2}, { 0,0}, { 2,2}, {0,2}, {6,3}, {7,3}, {0,0}
},{//1x{8,4}
{ 1,2}, { 2,2}, { 0,0}, {0,2}, {6,3}, {7,3}, {0,0}
},{//1x2, 2x1
{ 1,2}, { 0,0}, { 0,0}, {0,2}, {2,2}, {6,3}, {7,3}
}
}
};
static const uint8_t size2index[4][4]={
{-1, 3, 1, 1},
{ 3, 0, 0, 0},
{ 2, 0, 0, 0},
{ 2, 0, 0, 0},
};
static const int8_t mv[256][2]={
{ 0, 0},{ 0, -1},{ -1, 0},{ 1, 0},{ 0, 1},{ -1, -1},{ 1, -1},{ -1, 1},
{ 1, 1},{ 0, -2},{ -2, 0},{ 2, 0},{ 0, 2},{ -1, -2},{ 1, -2},{ -2, -1},
{ 2, -1},{ -2, 1},{ 2, 1},{ -1, 2},{ 1, 2},{ -2, -2},{ 2, -2},{ -2, 2},
{ 2, 2},{ 0, -3},{ -3, 0},{ 3, 0},{ 0, 3},{ -1, -3},{ 1, -3},{ -3, -1},
{ 3, -1},{ -3, 1},{ 3, 1},{ -1, 3},{ 1, 3},{ -2, -3},{ 2, -3},{ -3, -2},
{ 3, -2},{ -3, 2},{ 3, 2},{ -2, 3},{ 2, 3},{ 0, -4},{ -4, 0},{ 4, 0},
{ 0, 4},{ -1, -4},{ 1, -4},{ -4, -1},{ 4, -1},{ 4, 1},{ -1, 4},{ 1, 4},
{ -3, -3},{ -3, 3},{ 3, 3},{ -2, -4},{ -4, -2},{ 4, -2},{ -4, 2},{ -2, 4},
{ 2, 4},{ -3, -4},{ 3, -4},{ 4, -3},{ -5, 0},{ -4, 3},{ -3, 4},{ 3, 4},
{ -1, -5},{ -5, -1},{ -5, 1},{ -1, 5},{ -2, -5},{ 2, -5},{ 5, -2},{ 5, 2},
{ -4, -4},{ -4, 4},{ -3, -5},{ -5, -3},{ -5, 3},{ 3, 5},{ -6, 0},{ 0, 6},
{ -6, -1},{ -6, 1},{ 1, 6},{ 2, -6},{ -6, 2},{ 2, 6},{ -5, -4},{ 5, 4},
{ 4, 5},{ -6, -3},{ 6, 3},{ -7, 0},{ -1, -7},{ 5, -5},{ -7, 1},{ -1, 7},
{ 4, -6},{ 6, 4},{ -2, -7},{ -7, 2},{ -3, -7},{ 7, -3},{ 3, 7},{ 6, -5},
{ 0, -8},{ -1, -8},{ -7, -4},{ -8, 1},{ 4, 7},{ 2, -8},{ -2, 8},{ 6, 6},
{ -8, 3},{ 5, -7},{ -5, 7},{ 8, -4},{ 0, -9},{ -9, -1},{ 1, 9},{ 7, -6},
{ -7, 6},{ -5, -8},{ -5, 8},{ -9, 3},{ 9, -4},{ 7, -7},{ 8, -6},{ 6, 8},
{ 10, 1},{-10, 2},{ 9, -5},{ 10, -3},{ -8, -7},{-10, -4},{ 6, -9},{-11, 0},
{ 11, 1},{-11, -2},{ -2, 11},{ 7, -9},{ -7, 9},{ 10, 6},{ -4, 11},{ 8, -9},
{ 8, 9},{ 5, 11},{ 7,-10},{ 12, -3},{ 11, 6},{ -9, -9},{ 8, 10},{ 5, 12},
{-11, 7},{ 13, 2},{ 6,-12},{ 10, 9},{-11, 8},{ -7, 12},{ 0, 14},{ 14, -2},
{ -9, 11},{ -6, 13},{-14, -4},{ -5,-14},{ 5, 14},{-15, -1},{-14, -6},{ 3,-15},
{ 11,-11},{ -7, 14},{ -5, 15},{ 8,-14},{ 15, 6},{ 3, 16},{ 7,-15},{-16, 5},
{ 0, 17},{-16, -6},{-10, 14},{-16, 7},{ 12, 13},{-16, 8},{-17, 6},{-18, 3},
{ -7, 17},{ 15, 11},{ 16, 10},{ 2,-19},{ 3,-19},{-11,-16},{-18, 8},{-19, -6},
{ 2,-20},{-17,-11},{-10,-18},{ 8, 19},{-21, -1},{-20, 7},{ -4, 21},{ 21, 5},
{ 15, 16},{ 2,-22},{-10,-20},{-22, 5},{ 20,-11},{ -7,-22},{-12, 20},{ 23, -5},
{ 13,-20},{ 24, -2},{-15, 19},{-11, 22},{ 16, 19},{ 23,-10},{-18,-18},{ -9,-24},
{ 24,-10},{ -3, 26},{-23, 13},{-18,-20},{ 17, 21},{ -4, 27},{ 27, 6},{ 1,-28},
{-11, 26},{-17,-23},{ 7, 28},{ 11,-27},{ 29, 5},{-23,-19},{-28,-11},{-21, 22},
{-30, 7},{-17, 26},{-27, 16},{ 13, 29},{ 19,-26},{ 10,-31},{-14,-30},{ 20,-27},
{-29, 18},{-16,-31},{-28,-22},{ 21,-30},{-25, 28},{ 26,-29},{ 25,-32},{-32,-32}
};
// this is simply the scaled down elementwise product of the standard jpeg quantizer table and the AAN premul table
static const uint8_t dequant_table[64]={
16, 15, 13, 19, 24, 31, 28, 17,
17, 23, 25, 31, 36, 63, 45, 21,
18, 24, 27, 37, 52, 59, 49, 20,
16, 28, 34, 40, 60, 80, 51, 20,
18, 31, 48, 66, 68, 86, 56, 21,
19, 38, 56, 59, 64, 64, 48, 20,
27, 48, 55, 55, 56, 51, 35, 15,
20, 35, 34, 32, 31, 22, 15, 8,
};
static VLC block_type_vlc[2][4];
typedef struct CFrameBuffer{
unsigned int allocated_size;
unsigned int size;
int id;
uint8_t *data;
}CFrameBuffer;
typedef struct FourXContext{
AVCodecContext *avctx;
DSPContext dsp;
AVFrame current_picture, last_picture;
GetBitContext pre_gb; ///< ac/dc prefix
GetBitContext gb;
const uint8_t *bytestream;
const uint16_t *wordstream;
int mv[256];
VLC pre_vlc;
int last_dc;
DECLARE_ALIGNED(16, DCTELEM, block)[6][64];
void *bitstream_buffer;
unsigned int bitstream_buffer_size;
int version;
CFrameBuffer cfrm[CFRAME_BUFFER_COUNT];
} FourXContext;
#define FIX_1_082392200 70936
#define FIX_1_414213562 92682
#define FIX_1_847759065 121095
#define FIX_2_613125930 171254
#define MULTIPLY(var,const) (((var)*(const)) >> 16)
static void idct(DCTELEM block[64]){
int tmp0, tmp1, tmp2, tmp3, tmp4, tmp5, tmp6, tmp7;
int tmp10, tmp11, tmp12, tmp13;
int z5, z10, z11, z12, z13;
int i;
int temp[64];
for(i=0; i<8; i++){
tmp10 = block[8*0 + i] + block[8*4 + i];
tmp11 = block[8*0 + i] - block[8*4 + i];
tmp13 = block[8*2 + i] + block[8*6 + i];
tmp12 = MULTIPLY(block[8*2 + i] - block[8*6 + i], FIX_1_414213562) - tmp13;
tmp0 = tmp10 + tmp13;
tmp3 = tmp10 - tmp13;
tmp1 = tmp11 + tmp12;
tmp2 = tmp11 - tmp12;
z13 = block[8*5 + i] + block[8*3 + i];
z10 = block[8*5 + i] - block[8*3 + i];
z11 = block[8*1 + i] + block[8*7 + i];
z12 = block[8*1 + i] - block[8*7 + i];
tmp7 = z11 + z13;
tmp11 = MULTIPLY(z11 - z13, FIX_1_414213562);
z5 = MULTIPLY(z10 + z12, FIX_1_847759065);
tmp10 = MULTIPLY(z12, FIX_1_082392200) - z5;
tmp12 = MULTIPLY(z10, - FIX_2_613125930) + z5;
tmp6 = tmp12 - tmp7;
tmp5 = tmp11 - tmp6;
tmp4 = tmp10 + tmp5;
temp[8*0 + i] = tmp0 + tmp7;
temp[8*7 + i] = tmp0 - tmp7;
temp[8*1 + i] = tmp1 + tmp6;
temp[8*6 + i] = tmp1 - tmp6;
temp[8*2 + i] = tmp2 + tmp5;
temp[8*5 + i] = tmp2 - tmp5;
temp[8*4 + i] = tmp3 + tmp4;
temp[8*3 + i] = tmp3 - tmp4;
}
for(i=0; i<8*8; i+=8){
tmp10 = temp[0 + i] + temp[4 + i];
tmp11 = temp[0 + i] - temp[4 + i];
tmp13 = temp[2 + i] + temp[6 + i];
tmp12 = MULTIPLY(temp[2 + i] - temp[6 + i], FIX_1_414213562) - tmp13;
tmp0 = tmp10 + tmp13;
tmp3 = tmp10 - tmp13;
tmp1 = tmp11 + tmp12;
tmp2 = tmp11 - tmp12;
z13 = temp[5 + i] + temp[3 + i];
z10 = temp[5 + i] - temp[3 + i];
z11 = temp[1 + i] + temp[7 + i];
z12 = temp[1 + i] - temp[7 + i];
tmp7 = z11 + z13;
tmp11 = MULTIPLY(z11 - z13, FIX_1_414213562);
z5 = MULTIPLY(z10 + z12, FIX_1_847759065);
tmp10 = MULTIPLY(z12, FIX_1_082392200) - z5;
tmp12 = MULTIPLY(z10, - FIX_2_613125930) + z5;
tmp6 = tmp12 - tmp7;
tmp5 = tmp11 - tmp6;
tmp4 = tmp10 + tmp5;
block[0 + i] = (tmp0 + tmp7)>>6;
block[7 + i] = (tmp0 - tmp7)>>6;
block[1 + i] = (tmp1 + tmp6)>>6;
block[6 + i] = (tmp1 - tmp6)>>6;
block[2 + i] = (tmp2 + tmp5)>>6;
block[5 + i] = (tmp2 - tmp5)>>6;
block[4 + i] = (tmp3 + tmp4)>>6;
block[3 + i] = (tmp3 - tmp4)>>6;
}
}
static av_cold void init_vlcs(FourXContext *f){
static VLC_TYPE table[8][32][2];
int i;
for(i=0; i<8; i++){
block_type_vlc[0][i].table= table[i];
block_type_vlc[0][i].table_allocated= 32;
init_vlc(&block_type_vlc[0][i], BLOCK_TYPE_VLC_BITS, 7,
&block_type_tab[0][i][0][1], 2, 1,
&block_type_tab[0][i][0][0], 2, 1, INIT_VLC_USE_NEW_STATIC);
}
}
static void init_mv(FourXContext *f){
int i;
for(i=0; i<256; i++){
if(f->version>1)
f->mv[i] = mv[i][0] + mv[i][1] *f->current_picture.linesize[0]/2;
else
f->mv[i] = (i&15) - 8 + ((i>>4)-8)*f->current_picture.linesize[0]/2;
}
}
static inline void mcdc(uint16_t *dst, uint16_t *src, int log2w, int h, int stride, int scale, int dc){
int i;
dc*= 0x10001;
switch(log2w){
case 0:
for(i=0; i<h; i++){
dst[0] = scale*src[0] + dc;
if(scale) src += stride;
dst += stride;
}
break;
case 1:
for(i=0; i<h; i++){
((uint32_t*)dst)[0] = scale*((uint32_t*)src)[0] + dc;
if(scale) src += stride;
dst += stride;
}
break;
case 2:
for(i=0; i<h; i++){
((uint32_t*)dst)[0] = scale*((uint32_t*)src)[0] + dc;
((uint32_t*)dst)[1] = scale*((uint32_t*)src)[1] + dc;
if(scale) src += stride;
dst += stride;
}
break;
case 3:
for(i=0; i<h; i++){
((uint32_t*)dst)[0] = scale*((uint32_t*)src)[0] + dc;
((uint32_t*)dst)[1] = scale*((uint32_t*)src)[1] + dc;
((uint32_t*)dst)[2] = scale*((uint32_t*)src)[2] + dc;
((uint32_t*)dst)[3] = scale*((uint32_t*)src)[3] + dc;
if(scale) src += stride;
dst += stride;
}
break;
default: assert(0);
}
}
static void decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src, int log2w, int log2h, int stride){
const int index= size2index[log2h][log2w];
const int h= 1<<log2h;
int code= get_vlc2(&f->gb, block_type_vlc[1-(f->version>1)][index].table, BLOCK_TYPE_VLC_BITS, 1);
uint16_t *start= (uint16_t*)f->last_picture.data[0];
uint16_t *end= start + stride*(f->avctx->height-h+1) - (1<<log2w);
assert(code>=0 && code<=6);
if(code == 0){
src += f->mv[ *f->bytestream++ ];
if(start > src || src > end){
av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
return;
}
mcdc(dst, src, log2w, h, stride, 1, 0);
}else if(code == 1){
log2h--;
decode_p_block(f, dst , src , log2w, log2h, stride);
decode_p_block(f, dst + (stride<<log2h), src + (stride<<log2h), log2w, log2h, stride);
}else if(code == 2){
log2w--;
decode_p_block(f, dst , src , log2w, log2h, stride);
decode_p_block(f, dst + (1<<log2w), src + (1<<log2w), log2w, log2h, stride);
}else if(code == 3 && f->version<2){
mcdc(dst, src, log2w, h, stride, 1, 0);
}else if(code == 4){
src += f->mv[ *f->bytestream++ ];
if(start > src || src > end){
av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
return;
}
mcdc(dst, src, log2w, h, stride, 1, le2me_16(*f->wordstream++));
}else if(code == 5){
mcdc(dst, src, log2w, h, stride, 0, le2me_16(*f->wordstream++));
}else if(code == 6){
if(log2w){
dst[0] = le2me_16(*f->wordstream++);
dst[1] = le2me_16(*f->wordstream++);
}else{
dst[0 ] = le2me_16(*f->wordstream++);
dst[stride] = le2me_16(*f->wordstream++);
}
}
}
static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length){
int x, y;
const int width= f->avctx->width;
const int height= f->avctx->height;
uint16_t *src= (uint16_t*)f->last_picture.data[0];
uint16_t *dst= (uint16_t*)f->current_picture.data[0];
const int stride= f->current_picture.linesize[0]>>1;
unsigned int bitstream_size, bytestream_size, wordstream_size, extra;
if(f->version>1){
extra=20;
bitstream_size= AV_RL32(buf+8);
wordstream_size= AV_RL32(buf+12);
bytestream_size= AV_RL32(buf+16);
}else{
extra=0;
bitstream_size = AV_RL16(buf-4);
wordstream_size= AV_RL16(buf-2);
bytestream_size= FFMAX(length - bitstream_size - wordstream_size, 0);
}
if(bitstream_size+ bytestream_size+ wordstream_size + extra != length
|| bitstream_size > (1<<26)
|| bytestream_size > (1<<26)
|| wordstream_size > (1<<26)
){
av_log(f->avctx, AV_LOG_ERROR, "lengths %d %d %d %d\n", bitstream_size, bytestream_size, wordstream_size,
bitstream_size+ bytestream_size+ wordstream_size - length);
return -1;
}
av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size, bitstream_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!f->bitstream_buffer)
return AVERROR(ENOMEM);
f->dsp.bswap_buf(f->bitstream_buffer, (const uint32_t*)(buf + extra), bitstream_size/4);
init_get_bits(&f->gb, f->bitstream_buffer, 8*bitstream_size);
f->wordstream= (const uint16_t*)(buf + extra + bitstream_size);
f->bytestream= buf + extra + bitstream_size + wordstream_size;
init_mv(f);
for(y=0; y<height; y+=8){
for(x=0; x<width; x+=8){
decode_p_block(f, dst + x, src + x, 3, 3, stride);
}
src += 8*stride;
dst += 8*stride;
}
if( bitstream_size != (get_bits_count(&f->gb)+31)/32*4
|| (((const char*)f->wordstream - (const char*)buf + 2)&~2) != extra + bitstream_size + wordstream_size
|| (((const char*)f->bytestream - (const char*)buf + 3)&~3) != extra + bitstream_size + wordstream_size + bytestream_size)
av_log(f->avctx, AV_LOG_ERROR, " %d %td %td bytes left\n",
bitstream_size - (get_bits_count(&f->gb)+31)/32*4,
-(((const char*)f->bytestream - (const char*)buf + 3)&~3) + (extra + bitstream_size + wordstream_size + bytestream_size),
-(((const char*)f->wordstream - (const char*)buf + 2)&~2) + (extra + bitstream_size + wordstream_size)
);
return 0;
}
/**
* decode block and dequantize.
* Note this is almost identical to MJPEG.
*/
static int decode_i_block(FourXContext *f, DCTELEM *block){
int code, i, j, level, val;
/* DC coef */
val = get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3);
if (val>>4){
av_log(f->avctx, AV_LOG_ERROR, "error dc run != 0\n");
}
if(val)
val = get_xbits(&f->gb, val);
val = val * dequant_table[0] + f->last_dc;
f->last_dc =
block[0] = val;
/* AC coefs */
i = 1;
for(;;) {
code = get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3);
/* EOB */
if (code == 0)
break;
if (code == 0xf0) {
i += 16;
} else {
level = get_xbits(&f->gb, code & 0xf);
i += code >> 4;
if (i >= 64) {
av_log(f->avctx, AV_LOG_ERROR, "run %d oveflow\n", i);
return 0;
}
j= ff_zigzag_direct[i];
block[j] = level * dequant_table[j];
i++;
if (i >= 64)
break;
}
}
return 0;
}
static inline void idct_put(FourXContext *f, int x, int y){
DCTELEM (*block)[64]= f->block;
int stride= f->current_picture.linesize[0]>>1;
int i;
uint16_t *dst = ((uint16_t*)f->current_picture.data[0]) + y * stride + x;
for(i=0; i<4; i++){
block[i][0] += 0x80*8*8;
idct(block[i]);
}
if(!(f->avctx->flags&CODEC_FLAG_GRAY)){
for(i=4; i<6; i++) idct(block[i]);
}
/* Note transform is:
y= ( 1b + 4g + 2r)/14
cb=( 3b - 2g - 1r)/14
cr=(-1b - 4g + 5r)/14
*/
for(y=0; y<8; y++){
for(x=0; x<8; x++){
DCTELEM *temp= block[(x>>2) + 2*(y>>2)] + 2*(x&3) + 2*8*(y&3); //FIXME optimize
int cb= block[4][x + 8*y];
int cr= block[5][x + 8*y];
int cg= (cb + cr)>>1;
int y;
cb+=cb;
y = temp[0];
dst[0 ]= ((y+cb)>>3) + (((y-cg)&0xFC)<<3) + (((y+cr)&0xF8)<<8);
y = temp[1];
dst[1 ]= ((y+cb)>>3) + (((y-cg)&0xFC)<<3) + (((y+cr)&0xF8)<<8);
y = temp[8];
dst[ stride]= ((y+cb)>>3) + (((y-cg)&0xFC)<<3) + (((y+cr)&0xF8)<<8);
y = temp[9];
dst[1+stride]= ((y+cb)>>3) + (((y-cg)&0xFC)<<3) + (((y+cr)&0xF8)<<8);
dst += 2;
}
dst += 2*stride - 2*8;
}
}
static int decode_i_mb(FourXContext *f){
int i;
f->dsp.clear_blocks(f->block[0]);
for(i=0; i<6; i++){
if(decode_i_block(f, f->block[i]) < 0)
return -1;
}
return 0;
}
static const uint8_t *read_huffman_tables(FourXContext *f, const uint8_t * const buf){
int frequency[512];
uint8_t flag[512];
int up[512];
uint8_t len_tab[257];
int bits_tab[257];
int start, end;
const uint8_t *ptr= buf;
int j;
memset(frequency, 0, sizeof(frequency));
memset(up, -1, sizeof(up));
start= *ptr++;
end= *ptr++;
for(;;){
int i;
for(i=start; i<=end; i++){
frequency[i]= *ptr++;
}
start= *ptr++;
if(start==0) break;
end= *ptr++;
}
frequency[256]=1;
while((ptr - buf)&3) ptr++; // 4byte align
for(j=257; j<512; j++){
int min_freq[2]= {256*256, 256*256};
int smallest[2]= {0, 0};
int i;
for(i=0; i<j; i++){
if(frequency[i] == 0) continue;
if(frequency[i] < min_freq[1]){
if(frequency[i] < min_freq[0]){
min_freq[1]= min_freq[0]; smallest[1]= smallest[0];
min_freq[0]= frequency[i];smallest[0]= i;
}else{
min_freq[1]= frequency[i];smallest[1]= i;
}
}
}
if(min_freq[1] == 256*256) break;
frequency[j]= min_freq[0] + min_freq[1];
flag[ smallest[0] ]= 0;
flag[ smallest[1] ]= 1;
up[ smallest[0] ]=
up[ smallest[1] ]= j;
frequency[ smallest[0] ]= frequency[ smallest[1] ]= 0;
}
for(j=0; j<257; j++){
int node;
int len=0;
int bits=0;
for(node= j; up[node] != -1; node= up[node]){
bits += flag[node]<<len;
len++;
if(len > 31) av_log(f->avctx, AV_LOG_ERROR, "vlc length overflow\n"); //can this happen at all ?
}
bits_tab[j]= bits;
len_tab[j]= len;
}
init_vlc(&f->pre_vlc, ACDC_VLC_BITS, 257,
len_tab , 1, 1,
bits_tab, 4, 4, 0);
return ptr;
}
static int mix(int c0, int c1){
int blue = 2*(c0&0x001F) + (c1&0x001F);
int green= (2*(c0&0x03E0) + (c1&0x03E0))>>5;
int red = 2*(c0>>10) + (c1>>10);
return red/3*1024 + green/3*32 + blue/3;
}
static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length){
int x, y, x2, y2;
const int width= f->avctx->width;
const int height= f->avctx->height;
uint16_t *dst= (uint16_t*)f->current_picture.data[0];
const int stride= f->current_picture.linesize[0]>>1;
for(y=0; y<height; y+=16){
for(x=0; x<width; x+=16){
unsigned int color[4], bits;
memset(color, 0, sizeof(color));
//warning following is purely guessed ...
color[0]= bytestream_get_le16(&buf);
color[1]= bytestream_get_le16(&buf);
if(color[0]&0x8000) av_log(NULL, AV_LOG_ERROR, "unk bit 1\n");
if(color[1]&0x8000) av_log(NULL, AV_LOG_ERROR, "unk bit 2\n");
color[2]= mix(color[0], color[1]);
color[3]= mix(color[1], color[0]);
bits= bytestream_get_le32(&buf);
for(y2=0; y2<16; y2++){
for(x2=0; x2<16; x2++){
int index= 2*(x2>>2) + 8*(y2>>2);
dst[y2*stride+x2]= color[(bits>>index)&3];
}
}
dst+=16;
}
dst += 16*stride - width;
}
return 0;
}
static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length){
int x, y;
const int width= f->avctx->width;
const int height= f->avctx->height;
uint16_t *dst= (uint16_t*)f->current_picture.data[0];
const int stride= f->current_picture.linesize[0]>>1;
const unsigned int bitstream_size= AV_RL32(buf);
const int token_count av_unused = AV_RL32(buf + bitstream_size + 8);
unsigned int prestream_size= 4*AV_RL32(buf + bitstream_size + 4);
const uint8_t *prestream= buf + bitstream_size + 12;
if(prestream_size + bitstream_size + 12 != length
|| bitstream_size > (1<<26)
|| prestream_size > (1<<26)){
av_log(f->avctx, AV_LOG_ERROR, "size mismatch %d %d %d\n", prestream_size, bitstream_size, length);
return -1;
}
prestream= read_huffman_tables(f, prestream);
init_get_bits(&f->gb, buf + 4, 8*bitstream_size);
prestream_size= length + buf - prestream;
av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size, prestream_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!f->bitstream_buffer)
return AVERROR(ENOMEM);
f->dsp.bswap_buf(f->bitstream_buffer, (const uint32_t*)prestream, prestream_size/4);
init_get_bits(&f->pre_gb, f->bitstream_buffer, 8*prestream_size);
f->last_dc= 0*128*8*8;
for(y=0; y<height; y+=16){
for(x=0; x<width; x+=16){
if(decode_i_mb(f) < 0)
return -1;
idct_put(f, x, y);
}
dst += 16*stride;
}
if(get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3) != 256)
av_log(f->avctx, AV_LOG_ERROR, "end mismatch\n");
return 0;
}
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
FourXContext * const f = avctx->priv_data;
AVFrame *picture = data;
AVFrame *p, temp;
int i, frame_4cc, frame_size;
frame_4cc= AV_RL32(buf);
if(buf_size != AV_RL32(buf+4)+8 || buf_size < 20){
av_log(f->avctx, AV_LOG_ERROR, "size mismatch %d %d\n", buf_size, AV_RL32(buf+4));
}
if(frame_4cc == AV_RL32("cfrm")){
int free_index=-1;
const int data_size= buf_size - 20;
const int id= AV_RL32(buf+12);
const int whole_size= AV_RL32(buf+16);
CFrameBuffer *cfrm;
for(i=0; i<CFRAME_BUFFER_COUNT; i++){
if(f->cfrm[i].id && f->cfrm[i].id < avctx->frame_number)
av_log(f->avctx, AV_LOG_ERROR, "lost c frame %d\n", f->cfrm[i].id);
}
for(i=0; i<CFRAME_BUFFER_COUNT; i++){
if(f->cfrm[i].id == id) break;
if(f->cfrm[i].size == 0 ) free_index= i;
}
if(i>=CFRAME_BUFFER_COUNT){
i= free_index;
f->cfrm[i].id= id;
}
cfrm= &f->cfrm[i];
cfrm->data= av_fast_realloc(cfrm->data, &cfrm->allocated_size, cfrm->size + data_size + FF_INPUT_BUFFER_PADDING_SIZE);
if(!cfrm->data){ //explicit check needed as memcpy below might not catch a NULL
av_log(f->avctx, AV_LOG_ERROR, "realloc falure");
return -1;
}
memcpy(cfrm->data + cfrm->size, buf+20, data_size);
cfrm->size += data_size;
if(cfrm->size >= whole_size){
buf= cfrm->data;
frame_size= cfrm->size;
if(id != avctx->frame_number){
av_log(f->avctx, AV_LOG_ERROR, "cframe id mismatch %d %d\n", id, avctx->frame_number);
}
cfrm->size= cfrm->id= 0;
frame_4cc= AV_RL32("pfrm");
}else
return buf_size;
}else{
buf= buf + 12;
frame_size= buf_size - 12;
}
temp= f->current_picture;
f->current_picture= f->last_picture;
f->last_picture= temp;
p= &f->current_picture;
avctx->coded_frame= p;
avctx->flags |= CODEC_FLAG_EMU_EDGE; // alternatively we would have to use our own buffer management
if(p->data[0])
avctx->release_buffer(avctx, p);
p->reference= 1;
if(avctx->get_buffer(avctx, p) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
if(frame_4cc == AV_RL32("ifr2")){
p->pict_type= FF_I_TYPE;
if(decode_i2_frame(f, buf-4, frame_size) < 0)
return -1;
}else if(frame_4cc == AV_RL32("ifrm")){
p->pict_type= FF_I_TYPE;
if(decode_i_frame(f, buf, frame_size) < 0)
return -1;
}else if(frame_4cc == AV_RL32("pfrm") || frame_4cc == AV_RL32("pfr2")){
p->pict_type= FF_P_TYPE;
if(decode_p_frame(f, buf, frame_size) < 0)
return -1;
}else if(frame_4cc == AV_RL32("snd_")){
av_log(avctx, AV_LOG_ERROR, "ignoring snd_ chunk length:%d\n", buf_size);
}else{
av_log(avctx, AV_LOG_ERROR, "ignoring unknown chunk length:%d\n", buf_size);
}
p->key_frame= p->pict_type == FF_I_TYPE;
*picture= *p;
*data_size = sizeof(AVPicture);
emms_c();
return buf_size;
}
static av_cold void common_init(AVCodecContext *avctx){
FourXContext * const f = avctx->priv_data;
dsputil_init(&f->dsp, avctx);
f->avctx= avctx;
}
static av_cold int decode_init(AVCodecContext *avctx){
FourXContext * const f = avctx->priv_data;
if(avctx->extradata_size != 4 || !avctx->extradata) {
av_log(avctx, AV_LOG_ERROR, "extradata wrong or missing\n");
return 1;
}
f->version= AV_RL32(avctx->extradata)>>16;
common_init(avctx);
init_vlcs(f);
if(f->version>2) avctx->pix_fmt= PIX_FMT_RGB565;
else avctx->pix_fmt= PIX_FMT_BGR555;
return 0;
}
static av_cold int decode_end(AVCodecContext *avctx){
FourXContext * const f = avctx->priv_data;
int i;
av_freep(&f->bitstream_buffer);
f->bitstream_buffer_size=0;
for(i=0; i<CFRAME_BUFFER_COUNT; i++){
av_freep(&f->cfrm[i].data);
f->cfrm[i].allocated_size= 0;
}
free_vlc(&f->pre_vlc);
if(f->current_picture.data[0])
avctx->release_buffer(avctx, &f->current_picture);
if(f->last_picture.data[0])
avctx->release_buffer(avctx, &f->last_picture);
return 0;
}
AVCodec fourxm_decoder = {
"4xm",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_4XM,
sizeof(FourXContext),
decode_init,
NULL,
decode_end,
decode_frame,
CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("4X Movie"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/8bps.c
-233
View File
@@ -1,233 +0,0 @@
/*
* Quicktime Planar RGB (8BPS) Video Decoder
* Copyright (C) 2003 Roberto Togni
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* QT 8BPS Video Decoder by Roberto Togni
* For more information about the 8BPS format, visit:
* http://www.pcisys.net/~melanson/codecs/
*
* Supports: PAL8 (RGB 8bpp, paletted)
* : BGR24 (RGB 24bpp) (can also output it as RGB32)
* : RGB32 (RGB 32bpp, 4th plane is probably alpha and it's ignored)
*
*/
#include <stdio.h>
#include <stdlib.h>
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
static const enum PixelFormat pixfmt_rgb24[] = {PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE};
/*
* Decoder context
*/
typedef struct EightBpsContext {
AVCodecContext *avctx;
AVFrame pic;
unsigned char planes;
unsigned char planemap[4];
} EightBpsContext;
/*
*
* Decode a frame
*
*/
static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
EightBpsContext * const c = avctx->priv_data;
const unsigned char *encoded = buf;
unsigned char *pixptr, *pixptr_end;
unsigned int height = avctx->height; // Real image height
unsigned int dlen, p, row;
const unsigned char *lp, *dp;
unsigned char count;
unsigned int px_inc;
unsigned int planes = c->planes;
unsigned char *planemap = c->planemap;
if(c->pic.data[0])
avctx->release_buffer(avctx, &c->pic);
c->pic.reference = 0;
c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
if(avctx->get_buffer(avctx, &c->pic) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
/* Set data pointer after line lengths */
dp = encoded + planes * (height << 1);
/* Ignore alpha plane, don't know what to do with it */
if (planes == 4)
planes--;
px_inc = planes + (avctx->pix_fmt == PIX_FMT_RGB32);
for (p = 0; p < planes; p++) {
/* Lines length pointer for this plane */
lp = encoded + p * (height << 1);
/* Decode a plane */
for(row = 0; row < height; row++) {
pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
pixptr_end = pixptr + c->pic.linesize[0];
dlen = be2me_16(*(const unsigned short *)(lp+row*2));
/* Decode a row of this plane */
while(dlen > 0) {
if(dp + 1 >= buf+buf_size) return -1;
if ((count = *dp++) <= 127) {
count++;
dlen -= count + 1;
if (pixptr + count * px_inc > pixptr_end)
break;
if(dp + count > buf+buf_size) return -1;
while(count--) {
*pixptr = *dp++;
pixptr += px_inc;
}
} else {
count = 257 - count;
if (pixptr + count * px_inc > pixptr_end)
break;
while(count--) {
*pixptr = *dp;
pixptr += px_inc;
}
dp++;
dlen -= 2;
}
}
}
}
if (avctx->palctrl) {
memcpy (c->pic.data[1], avctx->palctrl->palette, AVPALETTE_SIZE);
if (avctx->palctrl->palette_changed) {
c->pic.palette_has_changed = 1;
avctx->palctrl->palette_changed = 0;
} else
c->pic.palette_has_changed = 0;
}
*data_size = sizeof(AVFrame);
*(AVFrame*)data = c->pic;
/* always report that the buffer was completely consumed */
return buf_size;
}
/*
*
* Init 8BPS decoder
*
*/
static av_cold int decode_init(AVCodecContext *avctx)
{
EightBpsContext * const c = avctx->priv_data;
c->avctx = avctx;
c->pic.data[0] = NULL;
switch (avctx->bits_per_coded_sample) {
case 8:
avctx->pix_fmt = PIX_FMT_PAL8;
c->planes = 1;
c->planemap[0] = 0; // 1st plane is palette indexes
if (avctx->palctrl == NULL) {
av_log(avctx, AV_LOG_ERROR, "Error: PAL8 format but no palette from demuxer.\n");
return -1;
}
break;
case 24:
avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
c->planes = 3;
c->planemap[0] = 2; // 1st plane is red
c->planemap[1] = 1; // 2nd plane is green
c->planemap[2] = 0; // 3rd plane is blue
break;
case 32:
avctx->pix_fmt = PIX_FMT_RGB32;
c->planes = 4;
#if HAVE_BIGENDIAN
c->planemap[0] = 1; // 1st plane is red
c->planemap[1] = 2; // 2nd plane is green
c->planemap[2] = 3; // 3rd plane is blue
c->planemap[3] = 0; // 4th plane is alpha???
#else
c->planemap[0] = 2; // 1st plane is red
c->planemap[1] = 1; // 2nd plane is green
c->planemap[2] = 0; // 3rd plane is blue
c->planemap[3] = 3; // 4th plane is alpha???
#endif
break;
default:
av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n", avctx->bits_per_coded_sample);
return -1;
}
return 0;
}
/*
*
* Uninit 8BPS decoder
*
*/
static av_cold int decode_end(AVCodecContext *avctx)
{
EightBpsContext * const c = avctx->priv_data;
if (c->pic.data[0])
avctx->release_buffer(avctx, &c->pic);
return 0;
}
AVCodec eightbps_decoder = {
"8bps",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_8BPS,
sizeof(EightBpsContext),
decode_init,
NULL,
decode_end,
decode_frame,
CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/8svx.c
-113
View File
@@ -1,113 +0,0 @@
/*
* 8SVX audio decoder
* Copyright (C) 2008 Jaikrishnan Menon
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* 8svx audio decoder
* @author Jaikrishnan Menon
* supports: fibonacci delta encoding
* : exponential encoding
*/
#include "avcodec.h"
/** decoder context */
typedef struct EightSvxContext {
int16_t fib_acc;
const int16_t *table;
} EightSvxContext;
static const int16_t fibonacci[16] = { -34<<8, -21<<8, -13<<8, -8<<8, -5<<8, -3<<8, -2<<8, -1<<8,
0, 1<<8, 2<<8, 3<<8, 5<<8, 8<<8, 13<<8, 21<<8 };
static const int16_t exponential[16] = { -128<<8, -64<<8, -32<<8, -16<<8, -8<<8, -4<<8, -2<<8, -1<<8,
0, 1<<8, 2<<8, 4<<8, 8<<8, 16<<8, 32<<8, 64<<8 };
/** decode a frame */
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
EightSvxContext *esc = avctx->priv_data;
int16_t *out_data = data;
int consumed = buf_size;
const uint8_t *buf_end = buf + buf_size;
if((*data_size >> 2) < buf_size)
return -1;
if(avctx->frame_number == 0) {
esc->fib_acc = buf[1] << 8;
buf_size -= 2;
buf += 2;
}
*data_size = buf_size << 2;
while(buf < buf_end) {
uint8_t d = *buf++;
esc->fib_acc += esc->table[d & 0x0f];
*out_data++ = esc->fib_acc;
esc->fib_acc += esc->table[d >> 4];
*out_data++ = esc->fib_acc;
}
return consumed;
}
/** initialize 8svx decoder */
static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
{
EightSvxContext *esc = avctx->priv_data;
switch(avctx->codec->id) {
case CODEC_ID_8SVX_FIB:
esc->table = fibonacci;
break;
case CODEC_ID_8SVX_EXP:
esc->table = exponential;
break;
default:
return -1;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
AVCodec eightsvx_fib_decoder = {
.name = "8svx_fib",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_8SVX_FIB,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
};
AVCodec eightsvx_exp_decoder = {
.name = "8svx_exp",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_8SVX_EXP,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/Jamfile
-392
View File
@@ -1,392 +0,0 @@
SubDir HAIKU_TOP src add-ons media plugins ffmpeg libavcodec ;
SubDirHdrs [ FDirName $(SUBDIR) $(DOTDOT) ] ;
SubDirHdrs [ FDirName $(SUBDIR) $(TARGET_ARCH) ] ;
SubDirHdrs [ FDirName $(SUBDIR) $(DOTDOT) libavutil ] ;
SubDirHdrs [ FDirName $(SUBDIR) $(DOTDOT) libswscale ] ;
SubDirSysHdrs [ FDirName $(SUBDIR) $(DOTDOT) libogg include ] ;
SubDirSysHdrs [ FDirName $(SUBDIR) $(DOTDOT) libtheora include ] ;
SubDirSysHdrs [ FDirName $(SUBDIR) $(DOTDOT) libvorbis include ] ;
SubDirSysHdrs [ FDirName $(SUBDIR) $(DOTDOT) libspeex include ] ;
UseLibraryHeaders zlib ;
# filter warnings we don't want here
TARGET_WARNING_CCFLAGS = [ FFilter $(TARGET_WARNING_CCFLAGS)
: -Wall -Wmissing-prototypes -Wsign-compare -Wpointer-arith ] ;
# Some targets in this subdir fail to build with DEBUG=1
DEBUG = 0 ;
SubDirCcFlags $(HAIKU_FFMPEG_GCC_EXTRA_FLAGS) ;
SubDirCcFlags $(HAIKU_FFMPEG_DEFINES) ;
SubDirC++Flags $(HAIKU_FFMPEG_DEFINES) ;
StaticLibrary libavcodec.a :
4xm.c
8bps.c
8svx.c
aac_ac3_parser.c
aac_adtstoasc_bsf.c
aac_parser.c
aaccoder.c
aacdec.c
aacenc.c
aacps.c
aacpsy.c
aacsbr.c
aactab.c
aandcttab.c
aasc.c
ac3.c
ac3_parser.c
ac3dec.c
ac3dec_data.c
ac3enc.c
ac3tab.c
acelp_filters.c
acelp_pitch_delay.c
acelp_vectors.c
adpcm.c
adxdec.c
alac.c
allcodecs.c
alsdec.c
amrnbdec.c
anm.c
apedec.c
asv1.c
atrac.c
atrac1.c
atrac3.c
audioconvert.c
aura.c
avfft.c
avpacket.c
avs.c
beosthread.c
bethsoftvideo.c
bgmc.c
bfi.c
bink.c
binkaudio.c
binkidct.c
bitstream.c
bitstream_filter.c
bmp.c
c93.c
cabac.c
cavs.c
cavs_parser.c
cavsdec.c
cavsdsp.c
# cbrt_tablegen.c
cdgraphics.c
celp_filters.c
celp_math.c
cinepak.c
cljr.c
cook.c
# costablegen.c
cscd.c
cyuv.c
dca.c
dca_parser.c
dcadsp.c
dct.c
dirac.c
dirac_parser.c
dnxhd_parser.c
dnxhddata.c
dnxhddec.c
dpcm.c
dsicinav.c
dsputil.c
dump_extradata_bsf.c
dv.c
# dv_tablegen.c
dvbsub.c
dvbsub_parser.c
dvbsubdec.c
dvdata.c
dvdsub_parser.c
dvdsubdec.c
dwt.c
dxa.c
# dxva2.c
# dxva2_h264.c
# dxva2_vc1.c
eac3dec.c
eac3dec_data.c
eacmv.c
eaidct.c
eatgq.c
eatgv.c
eatqi.c
elbg.c
error_resilience.c
escape124.c
eval.c
faandct.c
faanidct.c
faxcompr.c
# fdctref.c # GPL, only used for DCT test program
fft.c
ffv1.c
flac.c
flacdata.c
flacdec.c
flacenc.c
flashsv.c
flicvideo.c
flvenc.c
flvdec.c
fraps.c
frwu.c
g726.c
# g729dec.c
gif.c
gifdec.c
golomb.c
h261.c
h261_parser.c
h261dec.c
h263.c
h263_parser.c
h263dec.c
h264.c
h264_cabac.c
h264_cavlc.c
h264_direct.c
h264_loopfilter.c
h264_mp4toannexb_bsf.c
h264_parser.c
h264_ps.c
h264_refs.c
h264_sei.c
h264dsp.c
# h264dspenc.c
h264idct.c
h264pred.c
huffman.c
huffyuv.c
idcinvideo.c
iff.c
iirfilter.c
imc.c
imgconvert.c
imx_dump_header_bsf.c
indeo2.c
indeo3.c
indeo5.c
intelh263dec.c
interplayvideo.c
intrax8.c
intrax8dsp.c
ituh263dec.c
ituh263enc.c
ivi_common.c
ivi_dsp.c
jfdctfst.c
jfdctint.c
jpegls.c
jpeglsdec.c
jrevdct.c
kgv1dec.c
kmvc.c
lcldec.c
# lclenc.c
libspeexdec.c
libtheoraenc.c
libvorbis.c
# ljpegenc.c
loco.c
lpc.c
lsp.c
lzw.c
# lzwenc.c
mace.c
mdct.c
# mdct_tablegen.c
mdec.c
mimic.c
mjpeg.c
mjpeg_parser.c
mjpega_dump_header_bsf.c
mjpegbdec.c
mjpegdec.c
mjpegenc.c
mlp.c
mlp_parser.c
mlpdec.c
mlpdsp.c
mmvideo.c
motion_est.c
motionpixels.c
# motionpixels_tablegen.c
movsub_bsf.c
mp3_header_compress_bsf.c
mp3_header_decompress_bsf.c
mpc.c
mpc7.c
mpc8.c
mpeg4audio.c
mpeg4video.c
mpeg4video_parser.c
mpeg4videoenc.c
mpeg4videodec.c
mpeg12.c
mpeg12data.c
mpeg12enc.c
mpegaudio.c
mpegaudio_parser.c
# mpegaudio_tablegen.c
mpegaudiodata.c
mpegaudiodec.c
mpegaudiodecheader.c
mpegaudioenc.c
mpegvideo.c
mpegvideo_enc.c
mpegvideo_parser.c
# mpegvideo_xvmc.c
msmpeg4.c
msmpeg4data.c
msrle.c
msrledec.c
msvideo1.c
nellymoser.c
nellymoserdec.c
noise_bsf.c
nuv.c
opt.c
options.c
pamenc.c
parser.c
pcm.c
pcm-mpeg.c
# pcm_tablegen.c
pcx.c
pgssubdec.c
png.c
pngdec.c
pnm.c
pnm_parser.c
pnmdec.c
# pnmenc.c
psymodel.c
pthread.c
ptx.c
qcelpdec.c
qdm2.c
# qdm2_tablegen.c
qdrw.c
qpeg.c
qtrle.c
r210dec.c
ra144.c
ra288.c
rangecoder.c
ratecontrol.c
raw.c
rawdec.c
rawenc.c
rdft.c
remove_extradata_bsf.c
resample.c
resample2.c
rl2.c
rle.c
# roqaudioenc.c
roqvideo.c
roqvideodec.c
# roqvideoenc.c
rpza.c
rtjpeg.c
rv10.c
rv30.c
rv30dsp.c
rv34.c
rv40.c
rv40dsp.c
s3tc.c
sgidec.c
shorten.c
simple_idct.c
sipr.c
sipr16k.c
smacker.c
smc.c
snow.c
sonic.c
sp5xdec.c
sunrast.c
svq1.c
svq1dec.c
svq3.c
synth_filter.c
tableprint.c
targa.c
tiertexseqv.c
tiff.c
# tiffenc.c
truemotion1.c
truemotion2.c
truespeech.c
tscc.c
tta.c
twinvq.c
txd.c
ulti.c
utils.c
# vaapi_h264.c
vb.c
vc1.c
vc1dec.c
vc1_parser.c
vc1data.c
vc1dsp.c
vcr1.c
vmdav.c
vmnc.c
vorbis.c
vorbis_data.c
vorbis_dec.c
vp3.c
vp3_parser.c
vp3dsp.c
vp5.c
vp56.c
vp56data.c
vp56dsp.c
vp6.c
vp6dsp.c
vqavideo.c
wavpack.c
wma.c
wmadec.c
wmaprodec.c
wmavoice.c
wmv2.c
wmv2dec.c
wnv1.c
ws-snd1.c
xan.c
xiph.c
xl.c
xsubdec.c
yop.c
zmbv.c
;
# GCC2 needs this in order to find codec.h in each of these directories
# respectively and to find the correct codec.h, we need to define this
# per individual file.
SourceHdrs libvorbis.c : [ FDirName $(SUBDIR) .. libvorbis include vorbis ] ;
SourceHdrs libtheoraenc.c : [ FDirName $(SUBDIR) .. libtheora include theora ] ;
SourceHdrs libspeexdec.c : [ FDirName $(SUBDIR) .. libspeex include speex ] ;
SubInclude HAIKU_TOP src add-ons media plugins ffmpeg libavcodec
$(TARGET_ARCH) ;
src/add-ons/media/plugins/ffmpeg/libavcodec/aac.h
-290
View File
@@ -1,290 +0,0 @@
/*
* AAC definitions and structures
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC definitions and structures
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef AVCODEC_AAC_H
#define AVCODEC_AAC_H
#include "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "mpeg4audio.h"
#include "sbr.h"
#include <stdint.h>
#define MAX_CHANNELS 64
#define MAX_ELEM_ID 16
#define TNS_MAX_ORDER 20
enum RawDataBlockType {
TYPE_SCE,
TYPE_CPE,
TYPE_CCE,
TYPE_LFE,
TYPE_DSE,
TYPE_PCE,
TYPE_FIL,
TYPE_END,
};
enum ExtensionPayloadID {
EXT_FILL,
EXT_FILL_DATA,
EXT_DATA_ELEMENT,
EXT_DYNAMIC_RANGE = 0xb,
EXT_SBR_DATA = 0xd,
EXT_SBR_DATA_CRC = 0xe,
};
enum WindowSequence {
ONLY_LONG_SEQUENCE,
LONG_START_SEQUENCE,
EIGHT_SHORT_SEQUENCE,
LONG_STOP_SEQUENCE,
};
enum BandType {
ZERO_BT = 0, ///< Scalefactors and spectral data are all zero.
FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word.
ESC_BT = 11, ///< Spectral data are coded with an escape sequence.
NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream.
INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions.
INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions.
};
#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
enum ChannelPosition {
AAC_CHANNEL_FRONT = 1,
AAC_CHANNEL_SIDE = 2,
AAC_CHANNEL_BACK = 3,
AAC_CHANNEL_LFE = 4,
AAC_CHANNEL_CC = 5,
};
/**
* The point during decoding at which channel coupling is applied.
*/
enum CouplingPoint {
BEFORE_TNS,
BETWEEN_TNS_AND_IMDCT,
AFTER_IMDCT = 3,
};
/**
* Output configuration status
*/
enum OCStatus {
OC_NONE, //< Output unconfigured
OC_TRIAL_PCE, //< Output configuration under trial specified by an inband PCE
OC_TRIAL_FRAME, //< Output configuration under trial specified by a frame header
OC_GLOBAL_HDR, //< Output configuration set in a global header but not yet locked
OC_LOCKED, //< Output configuration locked in place
};
/**
* Predictor State
*/
typedef struct {
float cor0;
float cor1;
float var0;
float var1;
float r0;
float r1;
} PredictorState;
#define MAX_PREDICTORS 672
#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times
#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0
#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
/**
* Individual Channel Stream
*/
typedef struct {
uint8_t max_sfb; ///< number of scalefactor bands per group
enum WindowSequence window_sequence[2];
uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
int num_window_groups;
uint8_t group_len[8];
const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window
int num_swb; ///< number of scalefactor window bands
int num_windows;
int tns_max_bands;
int predictor_present;
int predictor_initialized;
int predictor_reset_group;
uint8_t prediction_used[41];
} IndividualChannelStream;
/**
* Temporal Noise Shaping
*/
typedef struct {
int present;
int n_filt[8];
int length[8][4];
int direction[8][4];
int order[8][4];
float coef[8][4][TNS_MAX_ORDER];
} TemporalNoiseShaping;
/**
* Dynamic Range Control - decoded from the bitstream but not processed further.
*/
typedef struct {
int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
int dyn_rng_ctl[17]; ///< DRC magnitude information
int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
int prog_ref_level; /**< A reference level for the long-term program audio level for all
* channels combined.
*/
} DynamicRangeControl;
typedef struct {
int num_pulse;
int start;
int pos[4];
int amp[4];
} Pulse;
/**
* coupling parameters
*/
typedef struct {
enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
int num_coupled; ///< number of target elements
enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
int id_select[8]; ///< element id
int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
* [2] list of gains for left channel; [3] lists of gains for both channels
*/
float gain[16][120];
} ChannelCoupling;
/**
* Single Channel Element - used for both SCE and LFE elements.
*/
typedef struct {
IndividualChannelStream ics;
TemporalNoiseShaping tns;
Pulse pulse;
enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED(16, float, coeffs)[1024]; ///< coefficients for IMDCT
DECLARE_ALIGNED(16, float, saved)[1024]; ///< overlap
DECLARE_ALIGNED(16, float, ret)[2048]; ///< PCM output
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
/**
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct {
// CPE specific
int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
SpectralBandReplication sbr;
} ChannelElement;
/**
* main AAC context
*/
typedef struct {
AVCodecContext *avctx;
MPEG4AudioConfig m4ac;
int is_saved; ///< Set if elements have stored overlap from previous frame.
DynamicRangeControl che_drc;
/**
* @defgroup elements Channel element related data.
* @{
*/
enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
* first index as the first 4 raw data block types
*/
ChannelElement *che[4][MAX_ELEM_ID];
ChannelElement *tag_che_map[4][MAX_ELEM_ID];
uint8_t tags_seen_this_frame[4][MAX_ELEM_ID];
int tags_mapped;
/** @} */
/**
* @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
* @{
*/
DECLARE_ALIGNED(16, float, buf_mdct)[1024];
/** @} */
/**
* @defgroup tables Computed / set up during initialization.
* @{
*/
FFTContext mdct;
FFTContext mdct_small;
DSPContext dsp;
int random_state;
/** @} */
/**
* @defgroup output Members used for output interleaving.
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
float add_bias; ///< offset for dsp.float_to_int16
float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
DECLARE_ALIGNED(16, float, temp)[128];
enum OCStatus output_configured;
} AACContext;
#endif /* AVCODEC_AAC_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_ac3_parser.c
-102
View File
@@ -1,102 +0,0 @@
/*
* Common AAC and AC-3 parser
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "parser.h"
#include "aac_ac3_parser.h"
int ff_aac_ac3_parse(AVCodecParserContext *s1,
AVCodecContext *avctx,
const uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size)
{
AACAC3ParseContext *s = s1->priv_data;
ParseContext *pc = &s->pc;
int len, i;
int new_frame_start;
get_next:
i=END_NOT_FOUND;
if(s->remaining_size <= buf_size){
if(s->remaining_size && !s->need_next_header){
i= s->remaining_size;
s->remaining_size = 0;
}else{ //we need a header first
len=0;
for(i=s->remaining_size; i<buf_size; i++){
s->state = (s->state<<8) + buf[i];
if((len=s->sync(s->state, s, &s->need_next_header, &new_frame_start)))
break;
}
if(len<=0){
i=END_NOT_FOUND;
}else{
s->state=0;
i-= s->header_size -1;
s->remaining_size = len;
if(!new_frame_start || pc->index+i<=0){
s->remaining_size += i;
goto get_next;
}
}
}
}
if(ff_combine_frame(pc, i, &buf, &buf_size)<0){
s->remaining_size -= FFMIN(s->remaining_size, buf_size);
*poutbuf = NULL;
*poutbuf_size = 0;
return buf_size;
}
*poutbuf = buf;
*poutbuf_size = buf_size;
/* update codec info */
if(s->codec_id)
avctx->codec_id = s->codec_id;
/* Due to backwards compatible HE-AAC the sample rate, channel count,
and total number of samples found in an AAC ADTS header are not
reliable. Bit rate is still accurate because the total frame duration in
seconds is still correct (as is the number of bits in the frame). */
if (avctx->codec_id != CODEC_ID_AAC) {
avctx->sample_rate = s->sample_rate;
/* allow downmixing to stereo (or mono for AC-3) */
if(avctx->request_channels > 0 &&
avctx->request_channels < s->channels &&
(avctx->request_channels <= 2 ||
(avctx->request_channels == 1 &&
(avctx->codec_id == CODEC_ID_AC3 ||
avctx->codec_id == CODEC_ID_EAC3)))) {
avctx->channels = avctx->request_channels;
} else {
avctx->channels = s->channels;
avctx->channel_layout = s->channel_layout;
}
avctx->frame_size = s->samples;
}
avctx->bit_rate = s->bit_rate;
return i;
}
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_ac3_parser.h
-65
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@@ -1,65 +0,0 @@
/*
* Common AAC and AC-3 parser prototypes
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AAC_AC3_PARSER_H
#define AVCODEC_AAC_AC3_PARSER_H
#include <stdint.h>
#include "avcodec.h"
#include "parser.h"
typedef enum {
AAC_AC3_PARSE_ERROR_SYNC = -1,
AAC_AC3_PARSE_ERROR_BSID = -2,
AAC_AC3_PARSE_ERROR_SAMPLE_RATE = -3,
AAC_AC3_PARSE_ERROR_FRAME_SIZE = -4,
AAC_AC3_PARSE_ERROR_FRAME_TYPE = -5,
AAC_AC3_PARSE_ERROR_CRC = -6,
AAC_AC3_PARSE_ERROR_CHANNEL_CFG = -7,
} AACAC3ParseError;
typedef struct AACAC3ParseContext {
ParseContext pc;
int frame_size;
int header_size;
int (*sync)(uint64_t state, struct AACAC3ParseContext *hdr_info,
int *need_next_header, int *new_frame_start);
int channels;
int sample_rate;
int bit_rate;
int samples;
int64_t channel_layout;
int remaining_size;
uint64_t state;
int need_next_header;
enum CodecID codec_id;
} AACAC3ParseContext;
int ff_aac_ac3_parse(AVCodecParserContext *s1,
AVCodecContext *avctx,
const uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size);
#endif /* AVCODEC_AAC_AC3_PARSER_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_adtstoasc_bsf.c
-114
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@@ -1,114 +0,0 @@
/*
* MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration bitstream filter
* Copyright (c) 2009 Alex Converse <[email protected]>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "aac_parser.h"
#include "put_bits.h"
#include "get_bits.h"
#include "mpeg4audio.h"
#include "internal.h"
typedef struct AACBSFContext {
int first_frame_done;
} AACBSFContext;
/**
* This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
* ADTS header and removes the ADTS header.
*/
static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
AVCodecContext *avctx, const char *args,
uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size,
int keyframe)
{
GetBitContext gb;
PutBitContext pb;
AACADTSHeaderInfo hdr;
AACBSFContext *ctx = bsfc->priv_data;
init_get_bits(&gb, buf, AAC_ADTS_HEADER_SIZE*8);
*poutbuf = (uint8_t*) buf;
*poutbuf_size = buf_size;
if (avctx->extradata)
if (show_bits(&gb, 12) != 0xfff)
return 0;
if (ff_aac_parse_header(&gb, &hdr) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing ADTS frame header!\n");
return -1;
}
if (!hdr.crc_absent && hdr.num_aac_frames > 1) {
av_log_missing_feature(avctx, "Multiple RDBs per frame with CRC is", 0);
return -1;
}
buf += AAC_ADTS_HEADER_SIZE + 2*!hdr.crc_absent;
buf_size -= AAC_ADTS_HEADER_SIZE + 2*!hdr.crc_absent;
if (!ctx->first_frame_done) {
int pce_size = 0;
uint8_t pce_data[MAX_PCE_SIZE];
if (!hdr.chan_config) {
init_get_bits(&gb, buf, buf_size);
if (get_bits(&gb, 3) != 5) {
av_log_missing_feature(avctx, "PCE based channel configuration, where the PCE is not the first syntax element is", 0);
return -1;
}
init_put_bits(&pb, pce_data, MAX_PCE_SIZE);
pce_size = ff_copy_pce_data(&pb, &gb)/8;
flush_put_bits(&pb);
buf_size -= get_bits_count(&gb)/8;
buf += get_bits_count(&gb)/8;
}
avctx->extradata_size = 2 + pce_size;
avctx->extradata = av_mallocz(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
put_bits(&pb, 5, hdr.object_type);
put_bits(&pb, 4, hdr.sampling_index);
put_bits(&pb, 4, hdr.chan_config);
put_bits(&pb, 1, 0); //frame length - 1024 samples
put_bits(&pb, 1, 0); //does not depend on core coder
put_bits(&pb, 1, 0); //is not extension
flush_put_bits(&pb);
if (pce_size) {
memcpy(avctx->extradata + 2, pce_data, pce_size);
}
ctx->first_frame_done = 1;
}
*poutbuf = (uint8_t*) buf;
*poutbuf_size = buf_size;
return 0;
}
AVBitStreamFilter aac_adtstoasc_bsf = {
"aac_adtstoasc",
sizeof(AACBSFContext),
aac_adtstoasc_filter,
};
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_parser.c
-112
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@@ -1,112 +0,0 @@
/*
* Audio and Video frame extraction
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "parser.h"
#include "aac_ac3_parser.h"
#include "aac_parser.h"
#include "get_bits.h"
#include "mpeg4audio.h"
int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
{
int size, rdb, ch, sr;
int aot, crc_abs;
if(get_bits(gbc, 12) != 0xfff)
return AAC_AC3_PARSE_ERROR_SYNC;
skip_bits1(gbc); /* id */
skip_bits(gbc, 2); /* layer */
crc_abs = get_bits1(gbc); /* protection_absent */
aot = get_bits(gbc, 2); /* profile_objecttype */
sr = get_bits(gbc, 4); /* sample_frequency_index */
if(!ff_mpeg4audio_sample_rates[sr])
return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
skip_bits1(gbc); /* private_bit */
ch = get_bits(gbc, 3); /* channel_configuration */
skip_bits1(gbc); /* original/copy */
skip_bits1(gbc); /* home */
/* adts_variable_header */
skip_bits1(gbc); /* copyright_identification_bit */
skip_bits1(gbc); /* copyright_identification_start */
size = get_bits(gbc, 13); /* aac_frame_length */
if(size < AAC_ADTS_HEADER_SIZE)
return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
skip_bits(gbc, 11); /* adts_buffer_fullness */
rdb = get_bits(gbc, 2); /* number_of_raw_data_blocks_in_frame */
hdr->object_type = aot + 1;
hdr->chan_config = ch;
hdr->crc_absent = crc_abs;
hdr->num_aac_frames = rdb + 1;
hdr->sampling_index = sr;
hdr->sample_rate = ff_mpeg4audio_sample_rates[sr];
hdr->samples = (rdb + 1) * 1024;
hdr->bit_rate = size * 8 * hdr->sample_rate / hdr->samples;
return size;
}
static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
int *need_next_header, int *new_frame_start)
{
GetBitContext bits;
AACADTSHeaderInfo hdr;
int size;
union {
uint64_t u64;
uint8_t u8[8];
} tmp;
tmp.u64 = be2me_64(state);
init_get_bits(&bits, tmp.u8+8-AAC_ADTS_HEADER_SIZE, AAC_ADTS_HEADER_SIZE * 8);
if ((size = ff_aac_parse_header(&bits, &hdr)) < 0)
return 0;
*need_next_header = 0;
*new_frame_start = 1;
hdr_info->sample_rate = hdr.sample_rate;
hdr_info->channels = ff_mpeg4audio_channels[hdr.chan_config];
hdr_info->samples = hdr.samples;
hdr_info->bit_rate = hdr.bit_rate;
return size;
}
static av_cold int aac_parse_init(AVCodecParserContext *s1)
{
AACAC3ParseContext *s = s1->priv_data;
s->header_size = AAC_ADTS_HEADER_SIZE;
s->sync = aac_sync;
return 0;
}
AVCodecParser aac_parser = {
{ CODEC_ID_AAC },
sizeof(AACAC3ParseContext),
aac_parse_init,
ff_aac_ac3_parse,
ff_parse_close,
};
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_parser.h
-55
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@@ -1,55 +0,0 @@
/*
* AAC parser prototypes
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AAC_PARSER_H
#define AVCODEC_AAC_PARSER_H
#include <stdint.h>
#include "aac_ac3_parser.h"
#include "get_bits.h"
#define AAC_ADTS_HEADER_SIZE 7
typedef struct {
uint32_t sample_rate;
uint32_t samples;
uint32_t bit_rate;
uint8_t crc_absent;
uint8_t object_type;
uint8_t sampling_index;
uint8_t chan_config;
uint8_t num_aac_frames;
} AACADTSHeaderInfo;
/**
* Parses AAC frame header.
* Parses the ADTS frame header to the end of the variable header, which is
* the first 54 bits.
* @param gbc[in] BitContext containing the first 54 bits of the frame.
* @param hdr[out] Pointer to struct where header info is written.
* @return Returns 0 on success, -1 if there is a sync word mismatch,
* -2 if the version element is invalid, -3 if the sample rate
* element is invalid, or -4 if the bit rate element is invalid.
*/
int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr);
#endif /* AVCODEC_AAC_PARSER_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_tablegen.c
-39
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@@ -1,39 +0,0 @@
/*
* Generate a header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <[email protected]>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "aac_tablegen.h"
#include "tableprint.h"
int main(void)
{
ff_aac_tableinit();
write_fileheader();
printf("const float ff_aac_pow2sf_tab[428] = {\n");
write_float_array(ff_aac_pow2sf_tab, 428);
printf("};\n");
return 0;
}
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_tablegen.h
-42
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@@ -1,42 +0,0 @@
/*
* Header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <[email protected]>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AAC_TABLEGEN_H
#define AAC_TABLEGEN_H
#include "aac_tablegen_decl.h"
#if CONFIG_HARDCODED_TABLES
#include "libavcodec/aac_tables.h"
#else
#include "../libavutil/mathematics.h"
float ff_aac_pow2sf_tab[428];
void ff_aac_tableinit(void)
{
int i;
for (i = 0; i < 428; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AAC_TABLEGEN_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aac_tablegen_decl.h
-34
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@@ -1,34 +0,0 @@
/*
* Header file for hardcoded AAC tables
*
* Copyright (c) 2010 Alex Converse <[email protected]>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AAC_TABLEGEN_INIT_H
#define AAC_TABLEGEN_INIT_H
#if CONFIG_HARDCODED_TABLES
#define ff_aac_tableinit()
extern const float ff_aac_pow2sf_tab[428];
#else
void ff_aac_tableinit(void);
extern float ff_aac_pow2sf_tab[428];
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AAC_TABLEGEN_INIT_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aaccoder.c
-960
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@@ -1,960 +0,0 @@
/*
* AAC coefficients encoder
* Copyright (C) 2008-2009 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC coefficients encoder
*/
/***********************************
* TODOs:
* speedup quantizer selection
* add sane pulse detection
***********************************/
#include "avcodec.h"
#include "put_bits.h"
#include "aac.h"
#include "aacenc.h"
#include "aactab.h"
/** bits needed to code codebook run value for long windows */
static const uint8_t run_value_bits_long[64] = {
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
};
/** bits needed to code codebook run value for short windows */
static const uint8_t run_value_bits_short[16] = {
3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
};
static const uint8_t *run_value_bits[2] = {
run_value_bits_long, run_value_bits_short
};
/**
* Quantize one coefficient.
* @return absolute value of the quantized coefficient
* @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
*/
static av_always_inline int quant(float coef, const float Q)
{
float a = coef * Q;
return sqrtf(a * sqrtf(a)) + 0.4054;
}
static void quantize_bands(int (*out)[2], const float *in, const float *scaled,
int size, float Q34, int is_signed, int maxval)
{
int i;
double qc;
for (i = 0; i < size; i++) {
qc = scaled[i] * Q34;
out[i][0] = (int)FFMIN(qc, (double)maxval);
out[i][1] = (int)FFMIN(qc + 0.4054, (double)maxval);
if (is_signed && in[i] < 0.0f) {
out[i][0] = -out[i][0];
out[i][1] = -out[i][1];
}
}
}
static void abs_pow34_v(float *out, const float *in, const int size)
{
#ifndef USE_REALLY_FULL_SEARCH
int i;
for (i = 0; i < size; i++) {
float a = fabsf(in[i]);
out[i] = sqrtf(a * sqrtf(a));
}
#endif /* USE_REALLY_FULL_SEARCH */
}
static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
/**
* Calculate rate distortion cost for quantizing with given codebook
*
* @return quantization distortion
*/
static float quantize_and_encode_band_cost(struct AACEncContext *s,
PutBitContext *pb, const float *in,
const float *scaled, int size, int scale_idx,
int cb, const float lambda, const float uplim,
int *bits)
{
const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
const float CLIPPED_ESCAPE = 165140.0f*IQ;
int i, j, k;
float cost = 0;
const int dim = cb < FIRST_PAIR_BT ? 4 : 2;
int resbits = 0;
#ifndef USE_REALLY_FULL_SEARCH
const float Q34 = sqrtf(Q * sqrtf(Q));
const int range = aac_cb_range[cb];
const int maxval = aac_cb_maxval[cb];
int offs[4];
#endif /* USE_REALLY_FULL_SEARCH */
if (!cb) {
for (i = 0; i < size; i++)
cost += in[i]*in[i];
if (bits)
*bits = 0;
return cost * lambda;
}
#ifndef USE_REALLY_FULL_SEARCH
offs[0] = 1;
for (i = 1; i < dim; i++)
offs[i] = offs[i-1]*range;
if (!scaled) {
abs_pow34_v(s->scoefs, in, size);
scaled = s->scoefs;
}
quantize_bands(s->qcoefs, in, scaled, size, Q34, !IS_CODEBOOK_UNSIGNED(cb), maxval);
#endif /* USE_REALLY_FULL_SEARCH */
for (i = 0; i < size; i += dim) {
float mincost;
int minidx = 0;
int minbits = 0;
const float *vec;
#ifndef USE_REALLY_FULL_SEARCH
int (*quants)[2] = &s->qcoefs[i];
mincost = 0.0f;
for (j = 0; j < dim; j++)
mincost += in[i+j]*in[i+j];
minidx = IS_CODEBOOK_UNSIGNED(cb) ? 0 : 40;
minbits = ff_aac_spectral_bits[cb-1][minidx];
mincost = mincost * lambda + minbits;
for (j = 0; j < (1<<dim); j++) {
float rd = 0.0f;
int curbits;
int curidx = IS_CODEBOOK_UNSIGNED(cb) ? 0 : 40;
int same = 0;
for (k = 0; k < dim; k++) {
if ((j & (1 << k)) && quants[k][0] == quants[k][1]) {
same = 1;
break;
}
}
if (same)
continue;
for (k = 0; k < dim; k++)
curidx += quants[k][!!(j & (1 << k))] * offs[dim - 1 - k];
curbits = ff_aac_spectral_bits[cb-1][curidx];
vec = &ff_aac_codebook_vectors[cb-1][curidx*dim];
#else
mincost = INFINITY;
vec = ff_aac_codebook_vectors[cb-1];
for (j = 0; j < ff_aac_spectral_sizes[cb-1]; j++, vec += dim) {
float rd = 0.0f;
int curbits = ff_aac_spectral_bits[cb-1][j];
int curidx = j;
#endif /* USE_REALLY_FULL_SEARCH */
if (IS_CODEBOOK_UNSIGNED(cb)) {
for (k = 0; k < dim; k++) {
float t = fabsf(in[i+k]);
float di;
if (vec[k] == 64.0f) { //FIXME: slow
//do not code with escape sequence small values
if (t < 39.0f*IQ) {
rd = INFINITY;
break;
}
if (t >= CLIPPED_ESCAPE) {
di = t - CLIPPED_ESCAPE;
curbits += 21;
} else {
int c = av_clip(quant(t, Q), 0, 8191);
di = t - c*cbrtf(c)*IQ;
curbits += av_log2(c)*2 - 4 + 1;
}
} else {
di = t - vec[k]*IQ;
}
if (vec[k] != 0.0f)
curbits++;
rd += di*di;
}
} else {
for (k = 0; k < dim; k++) {
float di = in[i+k] - vec[k]*IQ;
rd += di*di;
}
}
rd = rd * lambda + curbits;
if (rd < mincost) {
mincost = rd;
minidx = curidx;
minbits = curbits;
}
}
cost += mincost;
resbits += minbits;
if (cost >= uplim)
return uplim;
if (pb) {
put_bits(pb, ff_aac_spectral_bits[cb-1][minidx], ff_aac_spectral_codes[cb-1][minidx]);
if (IS_CODEBOOK_UNSIGNED(cb))
for (j = 0; j < dim; j++)
if (ff_aac_codebook_vectors[cb-1][minidx*dim+j] != 0.0f)
put_bits(pb, 1, in[i+j] < 0.0f);
if (cb == ESC_BT) {
for (j = 0; j < 2; j++) {
if (ff_aac_codebook_vectors[cb-1][minidx*2+j] == 64.0f) {
int coef = av_clip(quant(fabsf(in[i+j]), Q), 0, 8191);
int len = av_log2(coef);
put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
put_bits(pb, len, coef & ((1 << len) - 1));
}
}
}
}
}
if (bits)
*bits = resbits;
return cost;
}
static float quantize_band_cost(struct AACEncContext *s, const float *in,
const float *scaled, int size, int scale_idx,
int cb, const float lambda, const float uplim,
int *bits)
{
return quantize_and_encode_band_cost(s, NULL, in, scaled, size, scale_idx,
cb, lambda, uplim, bits);
}
static void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb,
const float *in, int size, int scale_idx,
int cb, const float lambda)
{
quantize_and_encode_band_cost(s, pb, in, NULL, size, scale_idx, cb, lambda,
INFINITY, NULL);
}
/**
* structure used in optimal codebook search
*/
typedef struct BandCodingPath {
int prev_idx; ///< pointer to the previous path point
float cost; ///< path cost
int run;
} BandCodingPath;
/**
* Encode band info for single window group bands.
*/
static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda)
{
BandCodingPath path[120][12];
int w, swb, cb, start, start2, size;
int i, j;
const int max_sfb = sce->ics.max_sfb;
const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
const int run_esc = (1 << run_bits) - 1;
int idx, ppos, count;
int stackrun[120], stackcb[120], stack_len;
float next_minrd = INFINITY;
int next_mincb = 0;
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
start = win*128;
for (cb = 0; cb < 12; cb++) {
path[0][cb].cost = 0.0f;
path[0][cb].prev_idx = -1;
path[0][cb].run = 0;
}
for (swb = 0; swb < max_sfb; swb++) {
start2 = start;
size = sce->ics.swb_sizes[swb];
if (sce->zeroes[win*16 + swb]) {
for (cb = 0; cb < 12; cb++) {
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = path[swb][cb].cost;
path[swb+1][cb].run = path[swb][cb].run + 1;
}
} else {
float minrd = next_minrd;
int mincb = next_mincb;
next_minrd = INFINITY;
next_mincb = 0;
for (cb = 0; cb < 12; cb++) {
float cost_stay_here, cost_get_here;
float rd = 0.0f;
for (w = 0; w < group_len; w++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(win+w)*16+swb];
rd += quantize_band_cost(s, sce->coeffs + start + w*128,
s->scoefs + start + w*128, size,
sce->sf_idx[(win+w)*16+swb], cb,
lambda / band->threshold, INFINITY, NULL);
}
cost_stay_here = path[swb][cb].cost + rd;
cost_get_here = minrd + rd + run_bits + 4;
if ( run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
!= run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
cost_stay_here += run_bits;
if (cost_get_here < cost_stay_here) {
path[swb+1][cb].prev_idx = mincb;
path[swb+1][cb].cost = cost_get_here;
path[swb+1][cb].run = 1;
} else {
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = cost_stay_here;
path[swb+1][cb].run = path[swb][cb].run + 1;
}
if (path[swb+1][cb].cost < next_minrd) {
next_minrd = path[swb+1][cb].cost;
next_mincb = cb;
}
}
}
start += sce->ics.swb_sizes[swb];
}
//convert resulting path from backward-linked list
stack_len = 0;
idx = 0;
for (cb = 1; cb < 12; cb++)
if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
idx = cb;
ppos = max_sfb;
while (ppos > 0) {
cb = idx;
stackrun[stack_len] = path[ppos][cb].run;
stackcb [stack_len] = cb;
idx = path[ppos-path[ppos][cb].run+1][cb].prev_idx;
ppos -= path[ppos][cb].run;
stack_len++;
}
//perform actual band info encoding
start = 0;
for (i = stack_len - 1; i >= 0; i--) {
put_bits(&s->pb, 4, stackcb[i]);
count = stackrun[i];
memset(sce->zeroes + win*16 + start, !stackcb[i], count);
//XXX: memset when band_type is also uint8_t
for (j = 0; j < count; j++) {
sce->band_type[win*16 + start] = stackcb[i];
start++;
}
while (count >= run_esc) {
put_bits(&s->pb, run_bits, run_esc);
count -= run_esc;
}
put_bits(&s->pb, run_bits, count);
}
}
typedef struct TrellisPath {
float cost;
int prev;
int min_val;
int max_val;
} TrellisPath;
#define TRELLIS_STAGES 121
#define TRELLIS_STATES 256
static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int q, w, w2, g, start = 0;
int i, j;
int idx;
TrellisPath paths[TRELLIS_STAGES][TRELLIS_STATES];
int bandaddr[TRELLIS_STAGES];
int minq;
float mincost;
for (i = 0; i < TRELLIS_STATES; i++) {
paths[0][i].cost = 0.0f;
paths[0][i].prev = -1;
paths[0][i].min_val = i;
paths[0][i].max_val = i;
}
for (j = 1; j < TRELLIS_STAGES; j++) {
for (i = 0; i < TRELLIS_STATES; i++) {
paths[j][i].cost = INFINITY;
paths[j][i].prev = -2;
paths[j][i].min_val = INT_MAX;
paths[j][i].max_val = 0;
}
}
idx = 1;
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
const float *coefs = sce->coeffs + start;
float qmin, qmax;
int nz = 0;
bandaddr[idx] = w * 16 + g;
qmin = INT_MAX;
qmax = 0.0f;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
if (band->energy <= band->threshold || band->threshold == 0.0f) {
sce->zeroes[(w+w2)*16+g] = 1;
continue;
}
sce->zeroes[(w+w2)*16+g] = 0;
nz = 1;
for (i = 0; i < sce->ics.swb_sizes[g]; i++) {
float t = fabsf(coefs[w2*128+i]);
if (t > 0.0f)
qmin = FFMIN(qmin, t);
qmax = FFMAX(qmax, t);
}
}
if (nz) {
int minscale, maxscale;
float minrd = INFINITY;
//minimum scalefactor index is when minimum nonzero coefficient after quantizing is not clipped
minscale = av_clip_uint8(log2(qmin)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
//maximum scalefactor index is when maximum coefficient after quantizing is still not zero
maxscale = av_clip_uint8(log2(qmax)*4 + 6 + SCALE_ONE_POS - SCALE_DIV_512);
for (q = minscale; q < maxscale; q++) {
float dists[12], dist;
memset(dists, 0, sizeof(dists));
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
int cb;
for (cb = 0; cb <= ESC_BT; cb++)
dists[cb] += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
q, cb, lambda / band->threshold, INFINITY, NULL);
}
dist = dists[0];
for (i = 1; i <= ESC_BT; i++)
dist = FFMIN(dist, dists[i]);
minrd = FFMIN(minrd, dist);
for (i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, TRELLIS_STATES); i++) {
float cost;
int minv, maxv;
if (isinf(paths[idx - 1][i].cost))
continue;
cost = paths[idx - 1][i].cost + dist
+ ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
minv = FFMIN(paths[idx - 1][i].min_val, q);
maxv = FFMAX(paths[idx - 1][i].max_val, q);
if (cost < paths[idx][q].cost && maxv-minv < SCALE_MAX_DIFF) {
paths[idx][q].cost = cost;
paths[idx][q].prev = i;
paths[idx][q].min_val = minv;
paths[idx][q].max_val = maxv;
}
}
}
} else {
for (q = 0; q < TRELLIS_STATES; q++) {
if (!isinf(paths[idx - 1][q].cost)) {
paths[idx][q].cost = paths[idx - 1][q].cost + 1;
paths[idx][q].prev = q;
paths[idx][q].min_val = FFMIN(paths[idx - 1][q].min_val, q);
paths[idx][q].max_val = FFMAX(paths[idx - 1][q].max_val, q);
continue;
}
for (i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, TRELLIS_STATES); i++) {
float cost;
int minv, maxv;
if (isinf(paths[idx - 1][i].cost))
continue;
cost = paths[idx - 1][i].cost + ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
minv = FFMIN(paths[idx - 1][i].min_val, q);
maxv = FFMAX(paths[idx - 1][i].max_val, q);
if (cost < paths[idx][q].cost && maxv-minv < SCALE_MAX_DIFF) {
paths[idx][q].cost = cost;
paths[idx][q].prev = i;
paths[idx][q].min_val = minv;
paths[idx][q].max_val = maxv;
}
}
}
}
sce->zeroes[w*16+g] = !nz;
start += sce->ics.swb_sizes[g];
idx++;
}
}
idx--;
mincost = paths[idx][0].cost;
minq = 0;
for (i = 1; i < TRELLIS_STATES; i++) {
if (paths[idx][i].cost < mincost) {
mincost = paths[idx][i].cost;
minq = i;
}
}
while (idx) {
sce->sf_idx[bandaddr[idx]] = minq;
minq = paths[idx][minq].prev;
idx--;
}
//set the same quantizers inside window groups
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
for (g = 0; g < sce->ics.num_swb; g++)
for (w2 = 1; w2 < sce->ics.group_len[w]; w2++)
sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
}
/**
* two-loop quantizers search taken from ISO 13818-7 Appendix C
*/
static void search_for_quantizers_twoloop(AVCodecContext *avctx,
AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int start = 0, i, w, w2, g;
int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels;
float dists[128], uplims[128];
int fflag, minscaler;
int its = 0;
int allz = 0;
float minthr = INFINITY;
//XXX: some heuristic to determine initial quantizers will reduce search time
memset(dists, 0, sizeof(dists));
//determine zero bands and upper limits
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = 0; g < sce->ics.num_swb; g++) {
int nz = 0;
float uplim = 0.0f;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
uplim += band->threshold;
if (band->energy <= band->threshold || band->threshold == 0.0f) {
sce->zeroes[(w+w2)*16+g] = 1;
continue;
}
nz = 1;
}
uplims[w*16+g] = uplim *512;
sce->zeroes[w*16+g] = !nz;
if (nz)
minthr = FFMIN(minthr, uplim);
allz = FFMAX(allz, nz);
}
}
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = 0; g < sce->ics.num_swb; g++) {
if (sce->zeroes[w*16+g]) {
sce->sf_idx[w*16+g] = SCALE_ONE_POS;
continue;
}
sce->sf_idx[w*16+g] = SCALE_ONE_POS + FFMIN(log2(uplims[w*16+g]/minthr)*4,59);
}
}
if (!allz)
return;
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
//perform two-loop search
//outer loop - improve quality
do {
int tbits, qstep;
minscaler = sce->sf_idx[0];
//inner loop - quantize spectrum to fit into given number of bits
qstep = its ? 1 : 32;
do {
int prev = -1;
tbits = 0;
fflag = 0;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
const float *coefs = sce->coeffs + start;
const float *scaled = s->scoefs + start;
int bits = 0;
int cb;
float mindist = INFINITY;
int minbits = 0;
if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
start += sce->ics.swb_sizes[g];
continue;
}
minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]);
for (cb = 0; cb <= ESC_BT; cb++) {
float dist = 0.0f;
int bb = 0;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
int b;
dist += quantize_band_cost(s, coefs + w2*128,
scaled + w2*128,
sce->ics.swb_sizes[g],
sce->sf_idx[w*16+g],
cb,
lambda,
INFINITY,
&b);
bb += b;
}
if (dist < mindist) {
mindist = dist;
minbits = bb;
}
}
dists[w*16+g] = (mindist - minbits) / lambda;
bits = minbits;
if (prev != -1) {
bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO];
}
tbits += bits;
start += sce->ics.swb_sizes[g];
prev = sce->sf_idx[w*16+g];
}
}
if (tbits > destbits) {
for (i = 0; i < 128; i++)
if (sce->sf_idx[i] < 218 - qstep)
sce->sf_idx[i] += qstep;
} else {
for (i = 0; i < 128; i++)
if (sce->sf_idx[i] > 60 - qstep)
sce->sf_idx[i] -= qstep;
}
qstep >>= 1;
if (!qstep && tbits > destbits*1.02)
qstep = 1;
if (sce->sf_idx[0] >= 217)
break;
} while (qstep);
fflag = 0;
minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
int prevsc = sce->sf_idx[w*16+g];
if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60)
sce->sf_idx[w*16+g]--;
sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF);
sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219);
if (sce->sf_idx[w*16+g] != prevsc)
fflag = 1;
}
}
its++;
} while (fflag && its < 10);
}
static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int start = 0, i, w, w2, g;
float uplim[128], maxq[128];
int minq, maxsf;
float distfact = ((sce->ics.num_windows > 1) ? 85.80 : 147.84) / lambda;
int last = 0, lastband = 0, curband = 0;
float avg_energy = 0.0;
if (sce->ics.num_windows == 1) {
start = 0;
for (i = 0; i < 1024; i++) {
if (i - start >= sce->ics.swb_sizes[curband]) {
start += sce->ics.swb_sizes[curband];
curband++;
}
if (sce->coeffs[i]) {
avg_energy += sce->coeffs[i] * sce->coeffs[i];
last = i;
lastband = curband;
}
}
} else {
for (w = 0; w < 8; w++) {
const float *coeffs = sce->coeffs + w*128;
start = 0;
for (i = 0; i < 128; i++) {
if (i - start >= sce->ics.swb_sizes[curband]) {
start += sce->ics.swb_sizes[curband];
curband++;
}
if (coeffs[i]) {
avg_energy += coeffs[i] * coeffs[i];
last = FFMAX(last, i);
lastband = FFMAX(lastband, curband);
}
}
}
}
last++;
avg_energy /= last;
if (avg_energy == 0.0f) {
for (i = 0; i < FF_ARRAY_ELEMS(sce->sf_idx); i++)
sce->sf_idx[i] = SCALE_ONE_POS;
return;
}
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
float *coefs = sce->coeffs + start;
const int size = sce->ics.swb_sizes[g];
int start2 = start, end2 = start + size, peakpos = start;
float maxval = -1, thr = 0.0f, t;
maxq[w*16+g] = 0.0f;
if (g > lastband) {
maxq[w*16+g] = 0.0f;
start += size;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
memset(coefs + w2*128, 0, sizeof(coefs[0])*size);
continue;
}
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
for (i = 0; i < size; i++) {
float t = coefs[w2*128+i]*coefs[w2*128+i];
maxq[w*16+g] = FFMAX(maxq[w*16+g], fabsf(coefs[w2*128 + i]));
thr += t;
if (sce->ics.num_windows == 1 && maxval < t) {
maxval = t;
peakpos = start+i;
}
}
}
if (sce->ics.num_windows == 1) {
start2 = FFMAX(peakpos - 2, start2);
end2 = FFMIN(peakpos + 3, end2);
} else {
start2 -= start;
end2 -= start;
}
start += size;
thr = pow(thr / (avg_energy * (end2 - start2)), 0.3 + 0.1*(lastband - g) / lastband);
t = 1.0 - (1.0 * start2 / last);
uplim[w*16+g] = distfact / (1.4 * thr + t*t*t + 0.075);
}
}
memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
const float *coefs = sce->coeffs + start;
const float *scaled = s->scoefs + start;
const int size = sce->ics.swb_sizes[g];
int scf, prev_scf, step;
int min_scf = -1, max_scf = 256;
float curdiff;
if (maxq[w*16+g] < 21.544) {
sce->zeroes[w*16+g] = 1;
start += size;
continue;
}
sce->zeroes[w*16+g] = 0;
scf = prev_scf = av_clip(SCALE_ONE_POS - SCALE_DIV_512 - log2(1/maxq[w*16+g])*16/3, 60, 218);
step = 16;
for (;;) {
float dist = 0.0f;
int quant_max;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
int b;
dist += quantize_band_cost(s, coefs + w2*128,
scaled + w2*128,
sce->ics.swb_sizes[g],
scf,
ESC_BT,
lambda,
INFINITY,
&b);
dist -= b;
}
dist *= 1.0f / 512.0f / lambda;
quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[200 - scf + SCALE_ONE_POS - SCALE_DIV_512]);
if (quant_max >= 8191) { // too much, return to the previous quantizer
sce->sf_idx[w*16+g] = prev_scf;
break;
}
prev_scf = scf;
curdiff = fabsf(dist - uplim[w*16+g]);
if (curdiff <= 1.0f)
step = 0;
else
step = log2(curdiff);
if (dist > uplim[w*16+g])
step = -step;
scf += step;
scf = av_clip_uint8(scf);
step = scf - prev_scf;
if (FFABS(step) <= 1 || (step > 0 && scf >= max_scf) || (step < 0 && scf <= min_scf)) {
sce->sf_idx[w*16+g] = av_clip(scf, min_scf, max_scf);
break;
}
if (step > 0)
min_scf = prev_scf;
else
max_scf = prev_scf;
}
start += size;
}
}
minq = sce->sf_idx[0] ? sce->sf_idx[0] : INT_MAX;
for (i = 1; i < 128; i++) {
if (!sce->sf_idx[i])
sce->sf_idx[i] = sce->sf_idx[i-1];
else
minq = FFMIN(minq, sce->sf_idx[i]);
}
if (minq == INT_MAX)
minq = 0;
minq = FFMIN(minq, SCALE_MAX_POS);
maxsf = FFMIN(minq + SCALE_MAX_DIFF, SCALE_MAX_POS);
for (i = 126; i >= 0; i--) {
if (!sce->sf_idx[i])
sce->sf_idx[i] = sce->sf_idx[i+1];
sce->sf_idx[i] = av_clip(sce->sf_idx[i], minq, maxsf);
}
}
static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
const float lambda)
{
int start = 0, i, w, w2, g;
int minq = 255;
memset(sce->sf_idx, 0, sizeof(sce->sf_idx));
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
if (band->energy <= band->threshold) {
sce->sf_idx[(w+w2)*16+g] = 218;
sce->zeroes[(w+w2)*16+g] = 1;
} else {
sce->sf_idx[(w+w2)*16+g] = av_clip(SCALE_ONE_POS - SCALE_DIV_512 + log2(band->threshold), 80, 218);
sce->zeroes[(w+w2)*16+g] = 0;
}
minq = FFMIN(minq, sce->sf_idx[(w+w2)*16+g]);
}
}
}
for (i = 0; i < 128; i++) {
sce->sf_idx[i] = 140;
//av_clip(sce->sf_idx[i], minq, minq + SCALE_MAX_DIFF - 1);
}
//set the same quantizers inside window groups
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
for (g = 0; g < sce->ics.num_swb; g++)
for (w2 = 1; w2 < sce->ics.group_len[w]; w2++)
sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
}
static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
const float lambda)
{
int start = 0, i, w, w2, g;
float M[128], S[128];
float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3;
SingleChannelElement *sce0 = &cpe->ch[0];
SingleChannelElement *sce1 = &cpe->ch[1];
if (!cpe->common_window)
return;
for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
for (g = 0; g < sce0->ics.num_swb; g++) {
if (!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]) {
float dist1 = 0.0f, dist2 = 0.0f;
for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
FFPsyBand *band0 = &s->psy.psy_bands[(s->cur_channel+0)*PSY_MAX_BANDS+(w+w2)*16+g];
FFPsyBand *band1 = &s->psy.psy_bands[(s->cur_channel+1)*PSY_MAX_BANDS+(w+w2)*16+g];
float minthr = FFMIN(band0->threshold, band1->threshold);
float maxthr = FFMAX(band0->threshold, band1->threshold);
for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
M[i] = (sce0->coeffs[start+w2*128+i]
+ sce1->coeffs[start+w2*128+i]) * 0.5;
S[i] = sce0->coeffs[start+w2*128+i]
- sce1->coeffs[start+w2*128+i];
}
abs_pow34_v(L34, sce0->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
abs_pow34_v(R34, sce1->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
abs_pow34_v(M34, M, sce0->ics.swb_sizes[g]);
abs_pow34_v(S34, S, sce0->ics.swb_sizes[g]);
dist1 += quantize_band_cost(s, sce0->coeffs + start + w2*128,
L34,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
lambda / band0->threshold, INFINITY, NULL);
dist1 += quantize_band_cost(s, sce1->coeffs + start + w2*128,
R34,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
lambda / band1->threshold, INFINITY, NULL);
dist2 += quantize_band_cost(s, M,
M34,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
lambda / maxthr, INFINITY, NULL);
dist2 += quantize_band_cost(s, S,
S34,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
lambda / minthr, INFINITY, NULL);
}
cpe->ms_mask[w*16+g] = dist2 < dist1;
}
start += sce0->ics.swb_sizes[g];
}
}
}
AACCoefficientsEncoder ff_aac_coders[] = {
{
search_for_quantizers_faac,
encode_window_bands_info,
quantize_and_encode_band,
search_for_ms,
},
{
search_for_quantizers_anmr,
encode_window_bands_info,
quantize_and_encode_band,
search_for_ms,
},
{
search_for_quantizers_twoloop,
encode_window_bands_info,
quantize_and_encode_band,
search_for_ms,
},
{
search_for_quantizers_fast,
encode_window_bands_info,
quantize_and_encode_band,
search_for_ms,
},
};
src/add-ons/media/plugins/ffmpeg/libavcodec/aacdec.c
-2142
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src/add-ons/media/plugins/ffmpeg/libavcodec/aacdectab.h
-95
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@@ -1,95 +0,0 @@
/*
* AAC decoder data
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC decoder data
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef AVCODEC_AACDECTAB_H
#define AVCODEC_AACDECTAB_H
#include "aac.h"
#include <stdint.h>
/* @name tns_tmp2_map
* Tables of the tmp2[] arrays of LPC coefficients used for TNS.
* The suffix _M_N[] indicate the values of coef_compress and coef_res
* respectively.
* @{
*/
static const float tns_tmp2_map_1_3[4] = {
0.00000000, -0.43388373, 0.64278758, 0.34202015,
};
static const float tns_tmp2_map_0_3[8] = {
0.00000000, -0.43388373, -0.78183150, -0.97492790,
0.98480773, 0.86602539, 0.64278758, 0.34202015,
};
static const float tns_tmp2_map_1_4[8] = {
0.00000000, -0.20791170, -0.40673664, -0.58778524,
0.67369562, 0.52643216, 0.36124167, 0.18374951,
};
static const float tns_tmp2_map_0_4[16] = {
0.00000000, -0.20791170, -0.40673664, -0.58778524,
-0.74314481, -0.86602539, -0.95105654, -0.99452192,
0.99573416, 0.96182561, 0.89516330, 0.79801720,
0.67369562, 0.52643216, 0.36124167, 0.18374951,
};
static const float * const tns_tmp2_map[4] = {
tns_tmp2_map_0_3,
tns_tmp2_map_0_4,
tns_tmp2_map_1_3,
tns_tmp2_map_1_4
};
// @}
static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };
static const uint8_t aac_channel_layout_map[7][5][2] = {
{ { TYPE_SCE, 0 }, },
{ { TYPE_CPE, 0 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_SCE, 1 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_CPE, 1 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 1 }, },
{ { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 2 }, { TYPE_CPE, 1 }, },
};
static const int64_t aac_channel_layout[8] = {
CH_LAYOUT_MONO,
CH_LAYOUT_STEREO,
CH_LAYOUT_SURROUND,
CH_LAYOUT_4POINT0,
CH_LAYOUT_5POINT0_BACK,
CH_LAYOUT_5POINT1_BACK,
CH_LAYOUT_7POINT1_WIDE,
0,
};
#endif /* AVCODEC_AACDECTAB_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aacenc.c
-644
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@@ -1,644 +0,0 @@
/*
* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC encoder
*/
/***********************************
* TODOs:
* add sane pulse detection
* add temporal noise shaping
***********************************/
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "mpeg4audio.h"
#include "aac.h"
#include "aactab.h"
#include "aacenc.h"
#include "psymodel.h"
static const uint8_t swb_size_1024_96[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
};
static const uint8_t swb_size_1024_64[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
};
static const uint8_t swb_size_1024_48[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
96
};
static const uint8_t swb_size_1024_32[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
};
static const uint8_t swb_size_1024_24[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
};
static const uint8_t swb_size_1024_16[] = {
8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
};
static const uint8_t swb_size_1024_8[] = {
12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};
static const uint8_t *swb_size_1024[] = {
swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
};
static const uint8_t swb_size_128_96[] = {
4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
};
static const uint8_t swb_size_128_48[] = {
4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
};
static const uint8_t swb_size_128_24[] = {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
};
static const uint8_t swb_size_128_16[] = {
4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
};
static const uint8_t swb_size_128_8[] = {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};
static const uint8_t *swb_size_128[] = {
/* the last entry on the following row is swb_size_128_64 but is a
duplicate of swb_size_128_96 */
swb_size_128_96, swb_size_128_96, swb_size_128_96,
swb_size_128_48, swb_size_128_48, swb_size_128_48,
swb_size_128_24, swb_size_128_24, swb_size_128_16,
swb_size_128_16, swb_size_128_16, swb_size_128_8
};
/** default channel configurations */
static const uint8_t aac_chan_configs[6][5] = {
{1, TYPE_SCE}, // 1 channel - single channel element
{1, TYPE_CPE}, // 2 channels - channel pair
{2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
{3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
{3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
};
/**
* Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
*/
static void put_audio_specific_config(AVCodecContext *avctx)
{
PutBitContext pb;
AACEncContext *s = avctx->priv_data;
init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
put_bits(&pb, 5, 2); //object type - AAC-LC
put_bits(&pb, 4, s->samplerate_index); //sample rate index
put_bits(&pb, 4, avctx->channels);
//GASpecificConfig
put_bits(&pb, 1, 0); //frame length - 1024 samples
put_bits(&pb, 1, 0); //does not depend on core coder
put_bits(&pb, 1, 0); //is not extension
flush_put_bits(&pb);
}
static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
int i;
const uint8_t *sizes[2];
int lengths[2];
avctx->frame_size = 1024;
for (i = 0; i < 16; i++)
if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
break;
if (i == 16) {
av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
return -1;
}
if (avctx->channels > 6) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
return -1;
}
if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
return -1;
}
if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
return -1;
}
s->samplerate_index = i;
dsputil_init(&s->dsp, avctx);
ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
ff_mdct_init(&s->mdct128, 8, 0, 1.0);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows(7);
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
avctx->extradata = av_malloc(2);
avctx->extradata_size = 2;
put_audio_specific_config(avctx);
sizes[0] = swb_size_1024[i];
sizes[1] = swb_size_128[i];
lengths[0] = ff_aac_num_swb_1024[i];
lengths[1] = ff_aac_num_swb_128[i];
ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
s->psypp = ff_psy_preprocess_init(avctx);
s->coder = &ff_aac_coders[2];
s->lambda = avctx->global_quality ? avctx->global_quality : 120;
ff_aac_tableinit();
if (avctx->channels > 5)
av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
"The output will most likely be an illegal bitstream.\n");
return 0;
}
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce, short *audio, int channel)
{
int i, j, k;
const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
memcpy(s->output, sce->saved, sizeof(float)*1024);
if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
memset(s->output, 0, sizeof(s->output[0]) * 448);
for (i = 448; i < 576; i++)
s->output[i] = sce->saved[i] * pwindow[i - 448];
for (i = 576; i < 704; i++)
s->output[i] = sce->saved[i];
}
if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) {
s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
sce->saved[i] = audio[j] * lwindow[i];
}
} else {
for (i = 0, j = channel; i < 448; i++, j += avctx->channels)
s->output[i+1024] = audio[j];
for (; i < 576; i++, j += avctx->channels)
s->output[i+1024] = audio[j] * swindow[576 - i - 1];
memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
} else {
for (k = 0; k < 1024; k += 128) {
for (i = 448 + k; i < 448 + k + 256; i++)
s->output[i - 448 - k] = (i < 1024)
? sce->saved[i]
: audio[channel + (i-1024)*avctx->channels];
s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
}
for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
}
/**
* Encode ics_info element.
* @see Table 4.6 (syntax of ics_info)
*/
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
int w;
put_bits(&s->pb, 1, 0); // ics_reserved bit
put_bits(&s->pb, 2, info->window_sequence[0]);
put_bits(&s->pb, 1, info->use_kb_window[0]);
if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
put_bits(&s->pb, 6, info->max_sfb);
put_bits(&s->pb, 1, 0); // no prediction
} else {
put_bits(&s->pb, 4, info->max_sfb);
for (w = 1; w < 8; w++)
put_bits(&s->pb, 1, !info->group_len[w]);
}
}
/**
* Encode MS data.
* @see 4.6.8.1 "Joint Coding - M/S Stereo"
*/
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
{
int i, w;
put_bits(pb, 2, cpe->ms_mode);
if (cpe->ms_mode == 1)
for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
}
/**
* Produce integer coefficients from scalefactors provided by the model.
*/
static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
{
int i, w, w2, g, ch;
int start, sum, maxsfb, cmaxsfb;
for (ch = 0; ch < chans; ch++) {
IndividualChannelStream *ics = &cpe->ch[ch].ics;
start = 0;
maxsfb = 0;
cpe->ch[ch].pulse.num_pulse = 0;
for (w = 0; w < ics->num_windows*16; w += 16) {
for (g = 0; g < ics->num_swb; g++) {
sum = 0;
//apply M/S
if (!ch && cpe->ms_mask[w + g]) {
for (i = 0; i < ics->swb_sizes[g]; i++) {
cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
}
}
start += ics->swb_sizes[g];
}
for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
;
maxsfb = FFMAX(maxsfb, cmaxsfb);
}
ics->max_sfb = maxsfb;
//adjust zero bands for window groups
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
for (g = 0; g < ics->max_sfb; g++) {
i = 1;
for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
if (!cpe->ch[ch].zeroes[w2*16 + g]) {
i = 0;
break;
}
}
cpe->ch[ch].zeroes[w*16 + g] = i;
}
}
}
if (chans > 1 && cpe->common_window) {
IndividualChannelStream *ics0 = &cpe->ch[0].ics;
IndividualChannelStream *ics1 = &cpe->ch[1].ics;
int msc = 0;
ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
ics1->max_sfb = ics0->max_sfb;
for (w = 0; w < ics0->num_windows*16; w += 16)
for (i = 0; i < ics0->max_sfb; i++)
if (cpe->ms_mask[w+i])
msc++;
if (msc == 0 || ics0->max_sfb == 0)
cpe->ms_mode = 0;
else
cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
}
}
/**
* Encode scalefactor band coding type.
*/
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
{
int w;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
}
/**
* Encode scalefactors.
*/
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce)
{
int off = sce->sf_idx[0], diff;
int i, w;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (i = 0; i < sce->ics.max_sfb; i++) {
if (!sce->zeroes[w*16 + i]) {
diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
if (diff < 0 || diff > 120)
av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
off = sce->sf_idx[w*16 + i];
put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
}
}
}
}
/**
* Encode pulse data.
*/
static void encode_pulses(AACEncContext *s, Pulse *pulse)
{
int i;
put_bits(&s->pb, 1, !!pulse->num_pulse);
if (!pulse->num_pulse)
return;
put_bits(&s->pb, 2, pulse->num_pulse - 1);
put_bits(&s->pb, 6, pulse->start);
for (i = 0; i < pulse->num_pulse; i++) {
put_bits(&s->pb, 5, pulse->pos[i]);
put_bits(&s->pb, 4, pulse->amp[i]);
}
}
/**
* Encode spectral coefficients processed by psychoacoustic model.
*/
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
{
int start, i, w, w2;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = 0;
for (i = 0; i < sce->ics.max_sfb; i++) {
if (sce->zeroes[w*16 + i]) {
start += sce->ics.swb_sizes[i];
continue;
}
for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
sce->ics.swb_sizes[i],
sce->sf_idx[w*16 + i],
sce->band_type[w*16 + i],
s->lambda);
start += sce->ics.swb_sizes[i];
}
}
}
/**
* Encode one channel of audio data.
*/
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
int common_window)
{
put_bits(&s->pb, 8, sce->sf_idx[0]);
if (!common_window)
put_ics_info(s, &sce->ics);
encode_band_info(s, sce);
encode_scale_factors(avctx, s, sce);
encode_pulses(s, &sce->pulse);
put_bits(&s->pb, 1, 0); //tns
put_bits(&s->pb, 1, 0); //ssr
encode_spectral_coeffs(s, sce);
return 0;
}
/**
* Write some auxiliary information about the created AAC file.
*/
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
const char *name)
{
int i, namelen, padbits;
namelen = strlen(name) + 2;
put_bits(&s->pb, 3, TYPE_FIL);
put_bits(&s->pb, 4, FFMIN(namelen, 15));
if (namelen >= 15)
put_bits(&s->pb, 8, namelen - 16);
put_bits(&s->pb, 4, 0); //extension type - filler
padbits = 8 - (put_bits_count(&s->pb) & 7);
align_put_bits(&s->pb);
for (i = 0; i < namelen - 2; i++)
put_bits(&s->pb, 8, name[i]);
put_bits(&s->pb, 12 - padbits, 0);
}
static int aac_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
{
AACEncContext *s = avctx->priv_data;
int16_t *samples = s->samples, *samples2, *la;
ChannelElement *cpe;
int i, j, chans, tag, start_ch;
const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
int chan_el_counter[4];
FFPsyWindowInfo windows[avctx->channels];
if (s->last_frame)
return 0;
if (data) {
if (!s->psypp) {
memcpy(s->samples + 1024 * avctx->channels, data,
1024 * avctx->channels * sizeof(s->samples[0]));
} else {
start_ch = 0;
samples2 = s->samples + 1024 * avctx->channels;
for (i = 0; i < chan_map[0]; i++) {
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
samples2 + start_ch, start_ch, chans);
start_ch += chans;
}
}
}
if (!avctx->frame_number) {
memcpy(s->samples, s->samples + 1024 * avctx->channels,
1024 * avctx->channels * sizeof(s->samples[0]));
return 0;
}
start_ch = 0;
for (i = 0; i < chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
samples2 = samples + start_ch;
la = samples2 + 1024 * avctx->channels + start_ch;
if (!data)
la = NULL;
for (j = 0; j < chans; j++) {
IndividualChannelStream *ics = &cpe->ch[j].ics;
int k;
wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = wi[j].window_type[0];
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = wi[j].window_shape;
ics->num_windows = wi[j].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
ics->num_swb = s->psy.num_bands[ics->num_windows == 8];
for (k = 0; k < ics->num_windows; k++)
ics->group_len[k] = wi[j].grouping[k];
s->cur_channel = start_ch + j;
apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
}
start_ch += chans;
}
do {
int frame_bits;
init_put_bits(&s->pb, frame, buf_size*8);
if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
start_ch = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for (i = 0; i < chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
}
cpe->common_window = 0;
if (chans > 1
&& wi[0].window_type[0] == wi[1].window_type[0]
&& wi[0].window_shape == wi[1].window_shape) {
cpe->common_window = 1;
for (j = 0; j < wi[0].num_windows; j++) {
if (wi[0].grouping[j] != wi[1].grouping[j]) {
cpe->common_window = 0;
break;
}
}
}
s->cur_channel = start_ch;
if (cpe->common_window && s->coder->search_for_ms)
s->coder->search_for_ms(s, cpe, s->lambda);
adjust_frame_information(s, cpe, chans);
put_bits(&s->pb, 3, tag);
put_bits(&s->pb, 4, chan_el_counter[tag]++);
if (chans == 2) {
put_bits(&s->pb, 1, cpe->common_window);
if (cpe->common_window) {
put_ics_info(s, &cpe->ch[0].ics);
encode_ms_info(&s->pb, cpe);
}
}
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
}
start_ch += chans;
}
frame_bits = put_bits_count(&s->pb);
if (frame_bits <= 6144 * avctx->channels - 3)
break;
s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
} while (1);
put_bits(&s->pb, 3, TYPE_END);
flush_put_bits(&s->pb);
avctx->frame_bits = put_bits_count(&s->pb);
// rate control stuff
if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
s->lambda *= ratio;
s->lambda = FFMIN(s->lambda, 65536.f);
}
if (!data)
s->last_frame = 1;
memcpy(s->samples, s->samples + 1024 * avctx->channels,
1024 * avctx->channels * sizeof(s->samples[0]));
return put_bits_count(&s->pb)>>3;
}
static av_cold int aac_encode_end(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
ff_mdct_end(&s->mdct1024);
ff_mdct_end(&s->mdct128);
ff_psy_end(&s->psy);
ff_psy_preprocess_end(s->psypp);
av_freep(&s->samples);
av_freep(&s->cpe);
return 0;
}
AVCodec aac_encoder = {
"aac",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_AAC,
sizeof(AACEncContext),
aac_encode_init,
aac_encode_frame,
aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/aacenc.h
-71
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@@ -1,71 +0,0 @@
/*
* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AACENC_H
#define AVCODEC_AACENC_H
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "aac.h"
#include "psymodel.h"
struct AACEncContext;
typedef struct AACCoefficientsEncoder {
void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
SingleChannelElement *sce, const float lambda);
void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda);
void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
int scale_idx, int cb, const float lambda);
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
} AACCoefficientsEncoder;
extern AACCoefficientsEncoder ff_aac_coders[];
/**
* AAC encoder context
*/
typedef struct AACEncContext {
PutBitContext pb;
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp;
DECLARE_ALIGNED(16, FFTSample, output)[2048]; ///< temporary buffer for MDCT input coefficients
int16_t* samples; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
ChannelElement *cpe; ///< channel elements
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
AACCoefficientsEncoder *coder;
int cur_channel;
int last_frame;
float lambda;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(16, float, scoefs)[1024]; ///< scaled coefficients
} AACEncContext;
#endif /* AVCODEC_AACENC_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aacps.c
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src/add-ons/media/plugins/ffmpeg/libavcodec/aacps.h
-82
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@@ -1,82 +0,0 @@
/*
* MPEG-4 Parametric Stereo definitions and declarations
* Copyright (c) 2010 Alex Converse <[email protected]>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_PS_H
#define AVCODEC_PS_H
#include <stdint.h>
#include "avcodec.h"
#include "get_bits.h"
#define PS_MAX_NUM_ENV 5
#define PS_MAX_NR_IIDICC 34
#define PS_MAX_NR_IPDOPD 17
#define PS_MAX_SSB 91
#define PS_MAX_AP_BANDS 50
#define PS_QMF_TIME_SLOTS 32
#define PS_MAX_DELAY 14
#define PS_AP_LINKS 3
#define PS_MAX_AP_DELAY 5
typedef struct {
int start;
int enable_iid;
int iid_quant;
int nr_iid_par;
int nr_ipdopd_par;
int enable_icc;
int icc_mode;
int nr_icc_par;
int enable_ext;
int frame_class;
int num_env_old;
int num_env;
int enable_ipdopd;
int border_position[PS_MAX_NUM_ENV+1];
int8_t iid_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Intensity Difference Parameters
int8_t icc_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-Channel Coherence Parameters
/* ipd/opd is iid/icc sized so that the same functions can handle both */
int8_t ipd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Phase Difference Parameters
int8_t opd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Overall Phase Difference Parameters
int is34bands;
int is34bands_old;
float in_buf[5][44][2];
float delay[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
float ap_delay[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
float peak_decay_nrg[34];
float power_smooth[34];
float peak_decay_diff_smooth[34];
float H11[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H12[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H21[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
float H22[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
int8_t opd_hist[PS_MAX_NR_IIDICC];
int8_t ipd_hist[PS_MAX_NR_IIDICC];
} PSContext;
void ff_ps_init(void);
void ff_ps_ctx_init(PSContext *ps);
int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top);
#endif /* AVCODEC_PS_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aacps_tablegen.c
-93
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@@ -1,93 +0,0 @@
/*
* Generate a header file for hardcoded Parametric Stereo tables
*
* Copyright (c) 2010 Alex Converse <[email protected]>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "aacps_tablegen.h"
#include "tableprint.h"
void write_float_3d_array (const void *p, int b, int c, int d)
{
int i;
const float *f = p;
for (i = 0; i < b; i++) {
printf("{\n");
write_float_2d_array(f, c, d);
printf("},\n");
f += c * d;
}
}
void write_float_4d_array (const void *p, int a, int b, int c, int d)
{
int i;
const float *f = p;
for (i = 0; i < a; i++) {
printf("{\n");
write_float_3d_array(f, b, c, d);
printf("},\n");
f += b * c * d;
}
}
int main(void)
{
ps_tableinit();
write_fileheader();
printf("static const float pd_re_smooth[8*8*8] = {\n");
write_float_array(pd_re_smooth, 8*8*8);
printf("};\n");
printf("static const float pd_im_smooth[8*8*8] = {\n");
write_float_array(pd_im_smooth, 8*8*8);
printf("};\n");
printf("static const float HA[46][8][4] = {\n");
write_float_3d_array(HA, 46, 8, 4);
printf("};\n");
printf("static const float HB[46][8][4] = {\n");
write_float_3d_array(HB, 46, 8, 4);
printf("};\n");
printf("static const float f20_0_8[8][7][2] = {\n");
write_float_3d_array(f20_0_8, 8, 7, 2);
printf("};\n");
printf("static const float f34_0_12[12][7][2] = {\n");
write_float_3d_array(f34_0_12, 12, 7, 2);
printf("};\n");
printf("static const float f34_1_8[8][7][2] = {\n");
write_float_3d_array(f34_1_8, 8, 7, 2);
printf("};\n");
printf("static const float f34_2_4[4][7][2] = {\n");
write_float_3d_array(f34_2_4, 4, 7, 2);
printf("};\n");
printf("static const float Q_fract_allpass[2][50][3][2] = {\n");
write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
printf("};\n");
printf("static const float phi_fract[2][50][2] = {\n");
write_float_3d_array(phi_fract, 2, 50, 2);
printf("};\n");
return 0;
}
src/add-ons/media/plugins/ffmpeg/libavcodec/aacps_tablegen.h
-212
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@@ -1,212 +0,0 @@
/*
* Header file for hardcoded Parametric Stereo tables
*
* Copyright (c) 2010 Alex Converse <[email protected]>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AACPS_TABLEGEN_H
#define AACPS_TABLEGEN_H
#include <stdint.h>
#if CONFIG_HARDCODED_TABLES
#define ps_tableinit()
#include "libavcodec/aacps_tables.h"
#else
#include "../libavutil/common.h"
#include "../libavutil/mathematics.h"
#define NR_ALLPASS_BANDS20 30
#define NR_ALLPASS_BANDS34 50
#define PS_AP_LINKS 3
static float pd_re_smooth[8*8*8];
static float pd_im_smooth[8*8*8];
static float HA[46][8][4];
static float HB[46][8][4];
static float f20_0_8 [ 8][7][2];
static float f34_0_12[12][7][2];
static float f34_1_8 [ 8][7][2];
static float f34_2_4 [ 4][7][2];
static float Q_fract_allpass[2][50][3][2];
static float phi_fract[2][50][2];
static const float g0_Q8[] = {
0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
0.09885108575264f, 0.11793710567217f, 0.125f
};
static const float g0_Q12[] = {
0.04081179924692f, 0.03812810994926f, 0.05144908135699f, 0.06399831151592f,
0.07428313801106f, 0.08100347892914f, 0.08333333333333f
};
static const float g1_Q8[] = {
0.01565675600122f, 0.03752716391991f, 0.05417891378782f, 0.08417044116767f,
0.10307344158036f, 0.12222452249753f, 0.125f
};
static const float g2_Q4[] = {
-0.05908211155639f, -0.04871498374946f, 0.0f, 0.07778723915851f,
0.16486303567403f, 0.23279856662996f, 0.25f
};
static void make_filters_from_proto(float (*filter)[7][2], const float *proto, int bands)
{
int q, n;
for (q = 0; q < bands; q++) {
for (n = 0; n < 7; n++) {
double theta = 2 * M_PI * (q + 0.5) * (n - 6) / bands;
filter[q][n][0] = proto[n] * cos(theta);
filter[q][n][1] = proto[n] * -sin(theta);
}
}
}
static void ps_tableinit(void)
{
static const float ipdopd_sin[] = { 0, M_SQRT1_2, 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2 };
static const float ipdopd_cos[] = { 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2, 0, M_SQRT1_2 };
int pd0, pd1, pd2;
static const float iid_par_dequant[] = {
//iid_par_dequant_default
0.05623413251903, 0.12589254117942, 0.19952623149689, 0.31622776601684,
0.44668359215096, 0.63095734448019, 0.79432823472428, 1,
1.25892541179417, 1.58489319246111, 2.23872113856834, 3.16227766016838,
5.01187233627272, 7.94328234724282, 17.7827941003892,
//iid_par_dequant_fine
0.00316227766017, 0.00562341325190, 0.01, 0.01778279410039,
0.03162277660168, 0.05623413251903, 0.07943282347243, 0.11220184543020,
0.15848931924611, 0.22387211385683, 0.31622776601684, 0.39810717055350,
0.50118723362727, 0.63095734448019, 0.79432823472428, 1,
1.25892541179417, 1.58489319246111, 1.99526231496888, 2.51188643150958,
3.16227766016838, 4.46683592150963, 6.30957344480193, 8.91250938133745,
12.5892541179417, 17.7827941003892, 31.6227766016838, 56.2341325190349,
100, 177.827941003892, 316.227766016837,
};
static const float icc_invq[] = {
1, 0.937, 0.84118, 0.60092, 0.36764, 0, -0.589, -1
};
static const float acos_icc_invq[] = {
0, 0.35685527, 0.57133466, 0.92614472, 1.1943263, M_PI/2, 2.2006171, M_PI
};
int iid, icc;
int k, m;
static const int8_t f_center_20[] = {
-3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
};
static const int8_t f_center_34[] = {
2, 6, 10, 14, 18, 22, 26, 30,
34,-10, -6, -2, 51, 57, 15, 21,
27, 33, 39, 45, 54, 66, 78, 42,
102, 66, 78, 90,102,114,126, 90,
};
static const float fractional_delay_links[] = { 0.43f, 0.75f, 0.347f };
const float fractional_delay_gain = 0.39f;
for (pd0 = 0; pd0 < 8; pd0++) {
float pd0_re = ipdopd_cos[pd0];
float pd0_im = ipdopd_sin[pd0];
for (pd1 = 0; pd1 < 8; pd1++) {
float pd1_re = ipdopd_cos[pd1];
float pd1_im = ipdopd_sin[pd1];
for (pd2 = 0; pd2 < 8; pd2++) {
float pd2_re = ipdopd_cos[pd2];
float pd2_im = ipdopd_sin[pd2];
float re_smooth = 0.25f * pd0_re + 0.5f * pd1_re + pd2_re;
float im_smooth = 0.25f * pd0_im + 0.5f * pd1_im + pd2_im;
float pd_mag = 1 / sqrt(im_smooth * im_smooth + re_smooth * re_smooth);
pd_re_smooth[pd0*64+pd1*8+pd2] = re_smooth * pd_mag;
pd_im_smooth[pd0*64+pd1*8+pd2] = im_smooth * pd_mag;
}
}
}
for (iid = 0; iid < 46; iid++) {
float c = iid_par_dequant[iid]; //<Linear Inter-channel Intensity Difference
float c1 = (float)M_SQRT2 / sqrtf(1.0f + c*c);
float c2 = c * c1;
for (icc = 0; icc < 8; icc++) {
/*if (PS_BASELINE || ps->icc_mode < 3)*/ {
float alpha = 0.5f * acos_icc_invq[icc];
float beta = alpha * (c1 - c2) * (float)M_SQRT1_2;
HA[iid][icc][0] = c2 * cosf(beta + alpha);
HA[iid][icc][1] = c1 * cosf(beta - alpha);
HA[iid][icc][2] = c2 * sinf(beta + alpha);
HA[iid][icc][3] = c1 * sinf(beta - alpha);
} /* else */ {
float alpha, gamma, mu, rho;
float alpha_c, alpha_s, gamma_c, gamma_s;
rho = FFMAX(icc_invq[icc], 0.05f);
alpha = 0.5f * atan2f(2.0f * c * rho, c*c - 1.0f);
mu = c + 1.0f / c;
mu = sqrtf(1 + (4 * rho * rho - 4)/(mu * mu));
gamma = atanf(sqrtf((1.0f - mu)/(1.0f + mu)));
if (alpha < 0) alpha += M_PI/2;
alpha_c = cosf(alpha);
alpha_s = sinf(alpha);
gamma_c = cosf(gamma);
gamma_s = sinf(gamma);
HB[iid][icc][0] = M_SQRT2 * alpha_c * gamma_c;
HB[iid][icc][1] = M_SQRT2 * alpha_s * gamma_c;
HB[iid][icc][2] = -M_SQRT2 * alpha_s * gamma_s;
HB[iid][icc][3] = M_SQRT2 * alpha_c * gamma_s;
}
}
}
for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
double f_center, theta;
if (k < FF_ARRAY_ELEMS(f_center_20))
f_center = f_center_20[k] * 0.125;
else
f_center = k - 6.5f;
for (m = 0; m < PS_AP_LINKS; m++) {
theta = -M_PI * fractional_delay_links[m] * f_center;
Q_fract_allpass[0][k][m][0] = cos(theta);
Q_fract_allpass[0][k][m][1] = sin(theta);
}
theta = -M_PI*fractional_delay_gain*f_center;
phi_fract[0][k][0] = cos(theta);
phi_fract[0][k][1] = sin(theta);
}
for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
double f_center, theta;
if (k < FF_ARRAY_ELEMS(f_center_34))
f_center = f_center_34[k] / 24.;
else
f_center = k - 26.5f;
for (m = 0; m < PS_AP_LINKS; m++) {
theta = -M_PI * fractional_delay_links[m] * f_center;
Q_fract_allpass[1][k][m][0] = cos(theta);
Q_fract_allpass[1][k][m][1] = sin(theta);
}
theta = -M_PI*fractional_delay_gain*f_center;
phi_fract[1][k][0] = cos(theta);
phi_fract[1][k][1] = sin(theta);
}
make_filters_from_proto(f20_0_8, g0_Q8, 8);
make_filters_from_proto(f34_0_12, g0_Q12, 12);
make_filters_from_proto(f34_1_8, g1_Q8, 8);
make_filters_from_proto(f34_2_4, g2_Q4, 4);
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AACPS_TABLEGEN_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aacpsdata.c
-163
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@@ -1,163 +0,0 @@
/*
* MPEG-4 Parametric Stereo data tables
* Copyright (c) 2010 Alex Converse <[email protected]>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
static const uint8_t huff_iid_df1_bits[] = {
18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15, 14, 14,
13, 12, 12, 11, 10, 10, 8, 7, 6, 5, 4, 3, 1, 3, 4, 5, 6, 7,
8, 9, 10, 11, 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, 18, 17, 18, 18,
18, 18, 18, 18, 18, 18, 18,
};
static const uint32_t huff_iid_df1_codes[] = {
0x01FEB4, 0x01FEB5, 0x01FD76, 0x01FD77, 0x01FD74, 0x01FD75, 0x01FE8A,
0x01FE8B, 0x01FE88, 0x00FE80, 0x01FEB6, 0x00FE82, 0x00FEB8, 0x007F42,
0x007FAE, 0x003FAF, 0x001FD1, 0x001FE9, 0x000FE9, 0x0007EA, 0x0007FB,
0x0003FB, 0x0001FB, 0x0001FF, 0x00007C, 0x00003C, 0x00001C, 0x00000C,
0x000000, 0x000001, 0x000001, 0x000002, 0x000001, 0x00000D, 0x00001D,
0x00003D, 0x00007D, 0x0000FC, 0x0001FC, 0x0003FC, 0x0003F4, 0x0007EB,
0x000FEA, 0x001FEA, 0x001FD6, 0x003FD0, 0x007FAF, 0x007F43, 0x00FEB9,
0x00FE83, 0x01FEB7, 0x00FE81, 0x01FE89, 0x01FE8E, 0x01FE8F, 0x01FE8C,
0x01FE8D, 0x01FEB2, 0x01FEB3, 0x01FEB0, 0x01FEB1,
};
static const uint8_t huff_iid_dt1_bits[] = {
16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 13,
13, 13, 12, 12, 11, 10, 9, 9, 7, 6, 5, 3, 1, 2, 5, 6, 7, 8,
9, 10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, 16, 16, 16, 16,
16, 16, 16, 16, 16, 16, 16,
};
static const uint16_t huff_iid_dt1_codes[] = {
0x004ED4, 0x004ED5, 0x004ECE, 0x004ECF, 0x004ECC, 0x004ED6, 0x004ED8,
0x004F46, 0x004F60, 0x002718, 0x002719, 0x002764, 0x002765, 0x00276D,
0x0027B1, 0x0013B7, 0x0013D6, 0x0009C7, 0x0009E9, 0x0009ED, 0x0004EE,
0x0004F7, 0x000278, 0x000139, 0x00009A, 0x00009F, 0x000020, 0x000011,
0x00000A, 0x000003, 0x000001, 0x000000, 0x00000B, 0x000012, 0x000021,
0x00004C, 0x00009B, 0x00013A, 0x000279, 0x000270, 0x0004EF, 0x0004E2,
0x0009EA, 0x0009D8, 0x0013D7, 0x0013D0, 0x0027B2, 0x0027A2, 0x00271A,
0x00271B, 0x004F66, 0x004F67, 0x004F61, 0x004F47, 0x004ED9, 0x004ED7,
0x004ECD, 0x004ED2, 0x004ED3, 0x004ED0, 0x004ED1,
};
static const uint8_t huff_iid_df0_bits[] = {
17, 17, 17, 17, 16, 15, 13, 10, 9, 7, 6, 5, 4, 3, 1, 3, 4, 5,
6, 6, 8, 11, 13, 14, 14, 15, 17, 18, 18,
};
static const uint32_t huff_iid_df0_codes[] = {
0x01FFFB, 0x01FFFC, 0x01FFFD, 0x01FFFA, 0x00FFFC, 0x007FFC, 0x001FFD,
0x0003FE, 0x0001FE, 0x00007E, 0x00003C, 0x00001D, 0x00000D, 0x000005,
0x000000, 0x000004, 0x00000C, 0x00001C, 0x00003D, 0x00003E, 0x0000FE,
0x0007FE, 0x001FFC, 0x003FFC, 0x003FFD, 0x007FFD, 0x01FFFE, 0x03FFFE,
0x03FFFF,
};
static const uint8_t huff_iid_dt0_bits[] = {
19, 19, 19, 20, 20, 20, 17, 15, 12, 10, 8, 6, 4, 2, 1, 3, 5, 7,
9, 11, 13, 14, 17, 19, 20, 20, 20, 20, 20,
};
static const uint32_t huff_iid_dt0_codes[] = {
0x07FFF9, 0x07FFFA, 0x07FFFB, 0x0FFFF8, 0x0FFFF9, 0x0FFFFA, 0x01FFFD,
0x007FFE, 0x000FFE, 0x0003FE, 0x0000FE, 0x00003E, 0x00000E, 0x000002,
0x000000, 0x000006, 0x00001E, 0x00007E, 0x0001FE, 0x0007FE, 0x001FFE,
0x003FFE, 0x01FFFC, 0x07FFF8, 0x0FFFFB, 0x0FFFFC, 0x0FFFFD, 0x0FFFFE,
0x0FFFFF,
};
static const uint8_t huff_icc_df_bits[] = {
14, 14, 12, 10, 7, 5, 3, 1, 2, 4, 6, 8, 9, 11, 13,
};
static const uint16_t huff_icc_df_codes[] = {
0x3FFF, 0x3FFE, 0x0FFE, 0x03FE, 0x007E, 0x001E, 0x0006, 0x0000,
0x0002, 0x000E, 0x003E, 0x00FE, 0x01FE, 0x07FE, 0x1FFE,
};
static const uint8_t huff_icc_dt_bits[] = {
14, 13, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12, 14,
};
static const uint16_t huff_icc_dt_codes[] = {
0x3FFE, 0x1FFE, 0x07FE, 0x01FE, 0x007E, 0x001E, 0x0006, 0x0000,
0x0002, 0x000E, 0x003E, 0x00FE, 0x03FE, 0x0FFE, 0x3FFF,
};
static const uint8_t huff_ipd_df_bits[] = {
1, 3, 4, 4, 4, 4, 4, 4,
};
static const uint8_t huff_ipd_df_codes[] = {
0x01, 0x00, 0x06, 0x04, 0x02, 0x03, 0x05, 0x07,
};
static const uint8_t huff_ipd_dt_bits[] = {
1, 3, 4, 5, 5, 4, 4, 3,
};
static const uint8_t huff_ipd_dt_codes[] = {
0x01, 0x02, 0x02, 0x03, 0x02, 0x00, 0x03, 0x03,
};
static const uint8_t huff_opd_df_bits[] = {
1, 3, 4, 4, 5, 5, 4, 3,
};
static const uint8_t huff_opd_df_codes[] = {
0x01, 0x01, 0x06, 0x04, 0x0F, 0x0E, 0x05, 0x00,
};
static const uint8_t huff_opd_dt_bits[] = {
1, 3, 4, 5, 5, 4, 4, 3,
};
static const uint8_t huff_opd_dt_codes[] = {
0x01, 0x02, 0x01, 0x07, 0x06, 0x00, 0x02, 0x03,
};
static const int8_t huff_offset[] = {
30, 30,
14, 14,
7, 7,
0, 0,
0, 0,
};
///Table 8.48
static const int8_t k_to_i_20[] = {
1, 0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 14, 15,
15, 15, 16, 16, 16, 16, 17, 17, 17, 17, 17, 18, 18, 18, 18, 18, 18, 18, 18,
18, 18, 18, 18, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19
};
///Table 8.49
static const int8_t k_to_i_34[] = {
0, 1, 2, 3, 4, 5, 6, 6, 7, 2, 1, 0, 10, 10, 4, 5, 6, 7, 8,
9, 10, 11, 12, 9, 14, 11, 12, 13, 14, 15, 16, 13, 16, 17, 18, 19, 20, 21,
22, 22, 23, 23, 24, 24, 25, 25, 26, 26, 27, 27, 27, 28, 28, 28, 29, 29, 29,
30, 30, 30, 31, 31, 31, 31, 32, 32, 32, 32, 33, 33, 33, 33, 33, 33, 33, 33,
33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33
};
static const float g1_Q2[] = {
0.0f, 0.01899487526049f, 0.0f, -0.07293139167538f,
0.0f, 0.30596630545168f, 0.5f
};
src/add-ons/media/plugins/ffmpeg/libavcodec/aacpsy.c
-318
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@@ -1,318 +0,0 @@
/*
* AAC encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC encoder psychoacoustic model
*/
#include "avcodec.h"
#include "aactab.h"
#include "psymodel.h"
/***********************************
* TODOs:
* thresholds linearization after their modifications for attaining given bitrate
* try other bitrate controlling mechanism (maybe use ratecontrol.c?)
* control quality for quality-based output
**********************************/
/**
* constants for 3GPP AAC psychoacoustic model
* @{
*/
#define PSY_3GPP_SPREAD_LOW 1.5f // spreading factor for ascending threshold spreading (15 dB/Bark)
#define PSY_3GPP_SPREAD_HI 3.0f // spreading factor for descending threshold spreading (30 dB/Bark)
#define PSY_3GPP_RPEMIN 0.01f
#define PSY_3GPP_RPELEV 2.0f
/**
* @}
*/
/**
* information for single band used by 3GPP TS26.403-inspired psychoacoustic model
*/
typedef struct Psy3gppBand{
float energy; ///< band energy
float ffac; ///< form factor
float thr; ///< energy threshold
float min_snr; ///< minimal SNR
float thr_quiet; ///< threshold in quiet
}Psy3gppBand;
/**
* single/pair channel context for psychoacoustic model
*/
typedef struct Psy3gppChannel{
Psy3gppBand band[128]; ///< bands information
Psy3gppBand prev_band[128]; ///< bands information from the previous frame
float win_energy; ///< sliding average of channel energy
float iir_state[2]; ///< hi-pass IIR filter state
uint8_t next_grouping; ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
}Psy3gppChannel;
/**
* psychoacoustic model frame type-dependent coefficients
*/
typedef struct Psy3gppCoeffs{
float ath [64]; ///< absolute threshold of hearing per bands
float barks [64]; ///< Bark value for each spectral band in long frame
float spread_low[64]; ///< spreading factor for low-to-high threshold spreading in long frame
float spread_hi [64]; ///< spreading factor for high-to-low threshold spreading in long frame
}Psy3gppCoeffs;
/**
* 3GPP TS26.403-inspired psychoacoustic model specific data
*/
typedef struct Psy3gppContext{
Psy3gppCoeffs psy_coef[2];
Psy3gppChannel *ch;
}Psy3gppContext;
/**
* Calculate Bark value for given line.
*/
static av_cold float calc_bark(float f)
{
return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
}
#define ATH_ADD 4
/**
* Calculate ATH value for given frequency.
* Borrowed from Lame.
*/
static av_cold float ath(float f, float add)
{
f /= 1000.0f;
return 3.64 * pow(f, -0.8)
- 6.8 * exp(-0.6 * (f - 3.4) * (f - 3.4))
+ 6.0 * exp(-0.15 * (f - 8.7) * (f - 8.7))
+ (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
}
static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
Psy3gppContext *pctx;
float barks[1024];
int i, j, g, start;
float prev, minscale, minath;
ctx->model_priv_data = av_mallocz(sizeof(Psy3gppContext));
pctx = (Psy3gppContext*) ctx->model_priv_data;
for (i = 0; i < 1024; i++)
barks[i] = calc_bark(i * ctx->avctx->sample_rate / 2048.0);
minath = ath(3410, ATH_ADD);
for (j = 0; j < 2; j++) {
Psy3gppCoeffs *coeffs = &pctx->psy_coef[j];
i = 0;
prev = 0.0;
for (g = 0; g < ctx->num_bands[j]; g++) {
i += ctx->bands[j][g];
coeffs->barks[g] = (barks[i - 1] + prev) / 2.0;
prev = barks[i - 1];
}
for (g = 0; g < ctx->num_bands[j] - 1; g++) {
coeffs->spread_low[g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_LOW);
coeffs->spread_hi [g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_HI);
}
start = 0;
for (g = 0; g < ctx->num_bands[j]; g++) {
minscale = ath(ctx->avctx->sample_rate * start / 1024.0, ATH_ADD);
for (i = 1; i < ctx->bands[j][g]; i++)
minscale = FFMIN(minscale, ath(ctx->avctx->sample_rate * (start + i) / 1024.0 / 2.0, ATH_ADD));
coeffs->ath[g] = minscale - minath;
start += ctx->bands[j][g];
}
}
pctx->ch = av_mallocz(sizeof(Psy3gppChannel) * ctx->avctx->channels);
return 0;
}
/**
* IIR filter used in block switching decision
*/
static float iir_filter(int in, float state[2])
{
float ret;
ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
state[0] = in;
state[1] = ret;
return ret;
}
/**
* window grouping information stored as bits (0 - new group, 1 - group continues)
*/
static const uint8_t window_grouping[9] = {
0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
};
/**
* Tell encoder which window types to use.
* @see 3GPP TS26.403 5.4.1 "Blockswitching"
*/
static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type)
{
int i, j;
int br = ctx->avctx->bit_rate / ctx->avctx->channels;
int attack_ratio = br <= 16000 ? 18 : 10;
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
uint8_t grouping = 0;
FFPsyWindowInfo wi;
memset(&wi, 0, sizeof(wi));
if (la) {
float s[8], v;
int switch_to_eight = 0;
float sum = 0.0, sum2 = 0.0;
int attack_n = 0;
for (i = 0; i < 8; i++) {
for (j = 0; j < 128; j++) {
v = iir_filter(audio[(i*128+j)*ctx->avctx->channels], pch->iir_state);
sum += v*v;
}
s[i] = sum;
sum2 += sum;
}
for (i = 0; i < 8; i++) {
if (s[i] > pch->win_energy * attack_ratio) {
attack_n = i + 1;
switch_to_eight = 1;
break;
}
}
pch->win_energy = pch->win_energy*7/8 + sum2/64;
wi.window_type[1] = prev_type;
switch (prev_type) {
case ONLY_LONG_SEQUENCE:
wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
break;
case LONG_START_SEQUENCE:
wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
grouping = pch->next_grouping;
break;
case LONG_STOP_SEQUENCE:
wi.window_type[0] = ONLY_LONG_SEQUENCE;
break;
case EIGHT_SHORT_SEQUENCE:
wi.window_type[0] = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
grouping = switch_to_eight ? pch->next_grouping : 0;
break;
}
pch->next_grouping = window_grouping[attack_n];
} else {
for (i = 0; i < 3; i++)
wi.window_type[i] = prev_type;
grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
}
wi.window_shape = 1;
if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
wi.num_windows = 1;
wi.grouping[0] = 1;
} else {
int lastgrp = 0;
wi.num_windows = 8;
for (i = 0; i < 8; i++) {
if (!((grouping >> i) & 1))
lastgrp = i;
wi.grouping[lastgrp]++;
}
}
return wi;
}
/**
* Calculate band thresholds as suggested in 3GPP TS26.403
*/
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
const float *coefs, FFPsyWindowInfo *wi)
{
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
int start = 0;
int i, w, g;
const int num_bands = ctx->num_bands[wi->num_windows == 8];
const uint8_t* band_sizes = ctx->bands[wi->num_windows == 8];
Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];
//calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
for (w = 0; w < wi->num_windows*16; w += 16) {
for (g = 0; g < num_bands; g++) {
Psy3gppBand *band = &pch->band[w+g];
band->energy = 0.0f;
for (i = 0; i < band_sizes[g]; i++)
band->energy += coefs[start+i] * coefs[start+i];
band->energy *= 1.0f / (512*512);
band->thr = band->energy * 0.001258925f;
start += band_sizes[g];
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].energy = band->energy;
}
}
//modify thresholds - spread, threshold in quiet - 5.4.3 "Spreaded Energy Calculation"
for (w = 0; w < wi->num_windows*16; w += 16) {
Psy3gppBand *band = &pch->band[w];
for (g = 1; g < num_bands; g++)
band[g].thr = FFMAX(band[g].thr, band[g-1].thr * coeffs->spread_low[g-1]);
for (g = num_bands - 2; g >= 0; g--)
band[g].thr = FFMAX(band[g].thr, band[g+1].thr * coeffs->spread_hi [g]);
for (g = 0; g < num_bands; g++) {
band[g].thr_quiet = FFMAX(band[g].thr, coeffs->ath[g]);
if (wi->num_windows != 8 && wi->window_type[1] != EIGHT_SHORT_SEQUENCE)
band[g].thr_quiet = FFMAX(PSY_3GPP_RPEMIN*band[g].thr_quiet,
FFMIN(band[g].thr_quiet,
PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
band[g].thr = FFMAX(band[g].thr, band[g].thr_quiet * 0.25);
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].threshold = band[g].thr;
}
}
memcpy(pch->prev_band, pch->band, sizeof(pch->band));
}
static av_cold void psy_3gpp_end(FFPsyContext *apc)
{
Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data;
av_freep(&pctx->ch);
av_freep(&apc->model_priv_data);
}
const FFPsyModel ff_aac_psy_model =
{
.name = "3GPP TS 26.403-inspired model",
.init = psy_3gpp_init,
.window = psy_3gpp_window,
.analyze = psy_3gpp_analyze,
.end = psy_3gpp_end,
};
src/add-ons/media/plugins/ffmpeg/libavcodec/aacpsy.h
-50
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@@ -1,50 +0,0 @@
/*
* AAC encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AACPSY_H
#define AVCODEC_AACPSY_H
#include "avcodec.h"
#include "aac.h"
//#include "lowpass.h"
enum AACPsyModelType{
AAC_PSY_TEST, ///< a sample model to exercise encoder
AAC_PSY_3GPP, ///< model following recommendations from 3GPP TS 26.403
AAC_NB_PSY_MODELS ///< total number of psychoacoustic models, since it's not a part of the ABI new models can be added freely
};
/**
* context used by psychoacoustic model
*/
typedef struct AACPsyContext {
AVCodecContext *avctx; ///< encoder context
}AACPsyContext;
/**
* Cleanup model context at the end.
*
* @param ctx model context
*/
void ff_aac_psy_end(AACPsyContext *ctx);
#endif /* AVCODEC_AACPSY_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aacsbr.c
-1771
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src/add-ons/media/plugins/ffmpeg/libavcodec/aacsbr.h
-49
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@@ -1,49 +0,0 @@
/*
* AAC Spectral Band Replication function declarations
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC Spectral Band Replication function declarations
* @author Robert Swain ( rob opendot cl )
*/
#ifndef AVCODEC_AACSBR_H
#define AVCODEC_AACSBR_H
#include "get_bits.h"
#include "aac.h"
#include "sbr.h"
/** Initialize SBR. */
av_cold void ff_aac_sbr_init(void);
/** Initialize one SBR context. */
av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr);
/** Close one SBR context. */
av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
/** Decode one SBR element. */
int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
GetBitContext *gb, int crc, int cnt, int id_aac);
/** Apply one SBR element to one AAC element. */
void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
float* L, float *R);
#endif /* AVCODEC_AACSBR_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aacsbrdata.h
-614
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@@ -1,614 +0,0 @@
/*
* AAC Spectral Band Replication decoding data
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC Spectral Band Replication decoding data
* @author Robert Swain ( rob opendot cl )
*/
#ifndef AVCODEC_AACSBRDATA_H
#define AVCODEC_AACSBRDATA_H
#include <stdint.h>
#include "libavutil/mem.h"
///< Huffman tables for SBR
static const uint8_t t_huffman_env_1_5dB_bits[121] = {
18, 18, 18, 18, 18, 18, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 17, 18, 16, 17, 18, 17,
16, 16, 16, 16, 15, 14, 14, 13,
13, 12, 11, 10, 9, 8, 7, 6,
5, 4, 3, 2, 2, 3, 4, 5,
6, 7, 8, 9, 10, 12, 13, 14,
14, 15, 16, 17, 16, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19,
};
static const uint32_t t_huffman_env_1_5dB_codes[121] = {
0x3ffd6, 0x3ffd7, 0x3ffd8, 0x3ffd9, 0x3ffda, 0x3ffdb, 0x7ffb8, 0x7ffb9,
0x7ffba, 0x7ffbb, 0x7ffbc, 0x7ffbd, 0x7ffbe, 0x7ffbf, 0x7ffc0, 0x7ffc1,
0x7ffc2, 0x7ffc3, 0x7ffc4, 0x7ffc5, 0x7ffc6, 0x7ffc7, 0x7ffc8, 0x7ffc9,
0x7ffca, 0x7ffcb, 0x7ffcc, 0x7ffcd, 0x7ffce, 0x7ffcf, 0x7ffd0, 0x7ffd1,
0x7ffd2, 0x7ffd3, 0x1ffe6, 0x3ffd4, 0x0fff0, 0x1ffe9, 0x3ffd5, 0x1ffe7,
0x0fff1, 0x0ffec, 0x0ffed, 0x0ffee, 0x07ff4, 0x03ff9, 0x03ff7, 0x01ffa,
0x01ff9, 0x00ffb, 0x007fc, 0x003fc, 0x001fd, 0x000fd, 0x0007d, 0x0003d,
0x0001d, 0x0000d, 0x00005, 0x00001, 0x00000, 0x00004, 0x0000c, 0x0001c,
0x0003c, 0x0007c, 0x000fc, 0x001fc, 0x003fd, 0x00ffa, 0x01ff8, 0x03ff6,
0x03ff8, 0x07ff5, 0x0ffef, 0x1ffe8, 0x0fff2, 0x7ffd4, 0x7ffd5, 0x7ffd6,
0x7ffd7, 0x7ffd8, 0x7ffd9, 0x7ffda, 0x7ffdb, 0x7ffdc, 0x7ffdd, 0x7ffde,
0x7ffdf, 0x7ffe0, 0x7ffe1, 0x7ffe2, 0x7ffe3, 0x7ffe4, 0x7ffe5, 0x7ffe6,
0x7ffe7, 0x7ffe8, 0x7ffe9, 0x7ffea, 0x7ffeb, 0x7ffec, 0x7ffed, 0x7ffee,
0x7ffef, 0x7fff0, 0x7fff1, 0x7fff2, 0x7fff3, 0x7fff4, 0x7fff5, 0x7fff6,
0x7fff7, 0x7fff8, 0x7fff9, 0x7fffa, 0x7fffb, 0x7fffc, 0x7fffd, 0x7fffe,
0x7ffff,
};
static const uint8_t f_huffman_env_1_5dB_bits[121] = {
19, 19, 20, 20, 20, 20, 20, 20,
20, 19, 20, 20, 20, 20, 19, 20,
19, 19, 20, 18, 20, 20, 20, 19,
20, 20, 20, 19, 20, 19, 18, 19,
18, 18, 17, 18, 17, 17, 17, 16,
16, 16, 15, 15, 14, 13, 13, 12,
12, 11, 10, 9, 9, 8, 7, 6,
5, 4, 3, 2, 2, 3, 4, 5,
6, 8, 8, 9, 10, 11, 11, 11,
12, 12, 13, 13, 14, 14, 16, 16,
17, 17, 18, 18, 18, 18, 18, 18,
18, 20, 19, 20, 20, 20, 20, 20,
20, 19, 20, 20, 20, 20, 19, 20,
18, 20, 20, 19, 19, 20, 20, 20,
20, 20, 20, 20, 20, 20, 20, 20,
20,
};
static const uint32_t f_huffman_env_1_5dB_codes[121] = {
0x7ffe7, 0x7ffe8, 0xfffd2, 0xfffd3, 0xfffd4, 0xfffd5, 0xfffd6, 0xfffd7,
0xfffd8, 0x7ffda, 0xfffd9, 0xfffda, 0xfffdb, 0xfffdc, 0x7ffdb, 0xfffdd,
0x7ffdc, 0x7ffdd, 0xfffde, 0x3ffe4, 0xfffdf, 0xfffe0, 0xfffe1, 0x7ffde,
0xfffe2, 0xfffe3, 0xfffe4, 0x7ffdf, 0xfffe5, 0x7ffe0, 0x3ffe8, 0x7ffe1,
0x3ffe0, 0x3ffe9, 0x1ffef, 0x3ffe5, 0x1ffec, 0x1ffed, 0x1ffee, 0x0fff4,
0x0fff3, 0x0fff0, 0x07ff7, 0x07ff6, 0x03ffa, 0x01ffa, 0x01ff9, 0x00ffa,
0x00ff8, 0x007f9, 0x003fb, 0x001fc, 0x001fa, 0x000fb, 0x0007c, 0x0003c,
0x0001c, 0x0000c, 0x00005, 0x00001, 0x00000, 0x00004, 0x0000d, 0x0001d,
0x0003d, 0x000fa, 0x000fc, 0x001fb, 0x003fa, 0x007f8, 0x007fa, 0x007fb,
0x00ff9, 0x00ffb, 0x01ff8, 0x01ffb, 0x03ff8, 0x03ff9, 0x0fff1, 0x0fff2,
0x1ffea, 0x1ffeb, 0x3ffe1, 0x3ffe2, 0x3ffea, 0x3ffe3, 0x3ffe6, 0x3ffe7,
0x3ffeb, 0xfffe6, 0x7ffe2, 0xfffe7, 0xfffe8, 0xfffe9, 0xfffea, 0xfffeb,
0xfffec, 0x7ffe3, 0xfffed, 0xfffee, 0xfffef, 0xffff0, 0x7ffe4, 0xffff1,
0x3ffec, 0xffff2, 0xffff3, 0x7ffe5, 0x7ffe6, 0xffff4, 0xffff5, 0xffff6,
0xffff7, 0xffff8, 0xffff9, 0xffffa, 0xffffb, 0xffffc, 0xffffd, 0xffffe,
0xfffff,
};
static const uint8_t t_huffman_env_bal_1_5dB_bits[49] = {
16, 16, 16, 16, 16, 16, 16, 16,
16, 16, 16, 16, 16, 16, 16, 16,
16, 16, 12, 11, 9, 7, 5, 3,
1, 2, 4, 6, 8, 11, 12, 15,
16, 16, 16, 16, 16, 16, 16, 17,
17, 17, 17, 17, 17, 17, 17, 17,
17,
};
static const uint32_t t_huffman_env_bal_1_5dB_codes[49] = {
0x0ffe4, 0x0ffe5, 0x0ffe6, 0x0ffe7, 0x0ffe8, 0x0ffe9, 0x0ffea, 0x0ffeb,
0x0ffec, 0x0ffed, 0x0ffee, 0x0ffef, 0x0fff0, 0x0fff1, 0x0fff2, 0x0fff3,
0x0fff4, 0x0ffe2, 0x00ffc, 0x007fc, 0x001fe, 0x0007e, 0x0001e, 0x00006,
0x00000, 0x00002, 0x0000e, 0x0003e, 0x000fe, 0x007fd, 0x00ffd, 0x07ff0,
0x0ffe3, 0x0fff5, 0x0fff6, 0x0fff7, 0x0fff8, 0x0fff9, 0x0fffa, 0x1fff6,
0x1fff7, 0x1fff8, 0x1fff9, 0x1fffa, 0x1fffb, 0x1fffc, 0x1fffd, 0x1fffe,
0x1ffff,
};
static const uint8_t f_huffman_env_bal_1_5dB_bits[49] = {
18, 18, 18, 18, 18, 18, 18, 18,
18, 18, 18, 18, 18, 18, 18, 16,
17, 14, 11, 11, 8, 7, 4, 2,
1, 3, 5, 6, 9, 11, 12, 15,
16, 18, 18, 18, 18, 18, 18, 18,
18, 18, 18, 18, 18, 18, 18, 19,
19,
};
static const uint32_t f_huffman_env_bal_1_5dB_codes[49] = {
0x3ffe2, 0x3ffe3, 0x3ffe4, 0x3ffe5, 0x3ffe6, 0x3ffe7, 0x3ffe8, 0x3ffe9,
0x3ffea, 0x3ffeb, 0x3ffec, 0x3ffed, 0x3ffee, 0x3ffef, 0x3fff0, 0x0fff7,
0x1fff0, 0x03ffc, 0x007fe, 0x007fc, 0x000fe, 0x0007e, 0x0000e, 0x00002,
0x00000, 0x00006, 0x0001e, 0x0003e, 0x001fe, 0x007fd, 0x00ffe, 0x07ffa,
0x0fff6, 0x3fff1, 0x3fff2, 0x3fff3, 0x3fff4, 0x3fff5, 0x3fff6, 0x3fff7,
0x3fff8, 0x3fff9, 0x3fffa, 0x3fffb, 0x3fffc, 0x3fffd, 0x3fffe, 0x7fffe,
0x7ffff,
};
static const uint8_t t_huffman_env_3_0dB_bits[63] = {
18, 18, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 17, 16, 16, 16, 14, 14, 14,
13, 12, 11, 8, 6, 4, 2, 1,
3, 5, 7, 9, 11, 13, 14, 14,
15, 16, 17, 18, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19, 19,
19, 19, 19, 19, 19, 19, 19,
};
static const uint32_t t_huffman_env_3_0dB_codes[63] = {
0x3ffed, 0x3ffee, 0x7ffde, 0x7ffdf, 0x7ffe0, 0x7ffe1, 0x7ffe2, 0x7ffe3,
0x7ffe4, 0x7ffe5, 0x7ffe6, 0x7ffe7, 0x7ffe8, 0x7ffe9, 0x7ffea, 0x7ffeb,
0x7ffec, 0x1fff4, 0x0fff7, 0x0fff9, 0x0fff8, 0x03ffb, 0x03ffa, 0x03ff8,
0x01ffa, 0x00ffc, 0x007fc, 0x000fe, 0x0003e, 0x0000e, 0x00002, 0x00000,
0x00006, 0x0001e, 0x0007e, 0x001fe, 0x007fd, 0x01ffb, 0x03ff9, 0x03ffc,
0x07ffa, 0x0fff6, 0x1fff5, 0x3ffec, 0x7ffed, 0x7ffee, 0x7ffef, 0x7fff0,
0x7fff1, 0x7fff2, 0x7fff3, 0x7fff4, 0x7fff5, 0x7fff6, 0x7fff7, 0x7fff8,
0x7fff9, 0x7fffa, 0x7fffb, 0x7fffc, 0x7fffd, 0x7fffe, 0x7ffff,
};
static const uint8_t f_huffman_env_3_0dB_bits[63] = {
20, 20, 20, 20, 20, 20, 20, 18,
19, 19, 19, 19, 18, 18, 20, 19,
17, 18, 17, 16, 16, 15, 14, 12,
11, 10, 9, 8, 6, 4, 2, 1,
3, 5, 8, 9, 10, 11, 12, 13,
14, 15, 15, 16, 16, 17, 17, 18,
18, 18, 20, 19, 19, 19, 20, 19,
19, 20, 20, 20, 20, 20, 20,
};
static const uint32_t f_huffman_env_3_0dB_codes[63] = {
0xffff0, 0xffff1, 0xffff2, 0xffff3, 0xffff4, 0xffff5, 0xffff6, 0x3fff3,
0x7fff5, 0x7ffee, 0x7ffef, 0x7fff6, 0x3fff4, 0x3fff2, 0xffff7, 0x7fff0,
0x1fff5, 0x3fff0, 0x1fff4, 0x0fff7, 0x0fff6, 0x07ff8, 0x03ffb, 0x00ffd,
0x007fd, 0x003fd, 0x001fd, 0x000fd, 0x0003e, 0x0000e, 0x00002, 0x00000,
0x00006, 0x0001e, 0x000fc, 0x001fc, 0x003fc, 0x007fc, 0x00ffc, 0x01ffc,
0x03ffa, 0x07ff9, 0x07ffa, 0x0fff8, 0x0fff9, 0x1fff6, 0x1fff7, 0x3fff5,
0x3fff6, 0x3fff1, 0xffff8, 0x7fff1, 0x7fff2, 0x7fff3, 0xffff9, 0x7fff7,
0x7fff4, 0xffffa, 0xffffb, 0xffffc, 0xffffd, 0xffffe, 0xfffff,
};
static const uint8_t t_huffman_env_bal_3_0dB_bits[25] = {
13, 13, 13, 13, 13, 13, 13, 12,
8, 7, 4, 3, 1, 2, 5, 6,
9, 13, 13, 13, 13, 13, 13, 14,
14,
};
static const uint16_t t_huffman_env_bal_3_0dB_codes[25] = {
0x1ff2, 0x1ff3, 0x1ff4, 0x1ff5, 0x1ff6, 0x1ff7, 0x1ff8, 0x0ff8,
0x00fe, 0x007e, 0x000e, 0x0006, 0x0000, 0x0002, 0x001e, 0x003e,
0x01fe, 0x1ff9, 0x1ffa, 0x1ffb, 0x1ffc, 0x1ffd, 0x1ffe, 0x3ffe,
0x3fff,
};
static const uint8_t f_huffman_env_bal_3_0dB_bits[25] = {
13, 13, 13, 13, 13, 14, 14, 11,
8, 7, 4, 2, 1, 3, 5, 6,
9, 12, 13, 14, 14, 14, 14, 14,
14,
};
static const uint16_t f_huffman_env_bal_3_0dB_codes[25] = {
0x1ff7, 0x1ff8, 0x1ff9, 0x1ffa, 0x1ffb, 0x3ff8, 0x3ff9, 0x07fc,
0x00fe, 0x007e, 0x000e, 0x0002, 0x0000, 0x0006, 0x001e, 0x003e,
0x01fe, 0x0ffa, 0x1ff6, 0x3ffa, 0x3ffb, 0x3ffc, 0x3ffd, 0x3ffe,
0x3fff,
};
static const uint8_t t_huffman_noise_3_0dB_bits[63] = {
13, 13, 13, 13, 13, 13, 13, 13,
13, 13, 13, 13, 13, 13, 13, 13,
13, 13, 13, 13, 13, 13, 13, 13,
13, 13, 11, 8, 6, 4, 3, 1,
2, 5, 8, 10, 13, 13, 13, 13,
13, 13, 13, 13, 13, 13, 13, 13,
13, 13, 13, 13, 13, 13, 13, 13,
13, 13, 13, 13, 13, 14, 14,
};
static const uint16_t t_huffman_noise_3_0dB_codes[63] = {
0x1fce, 0x1fcf, 0x1fd0, 0x1fd1, 0x1fd2, 0x1fd3, 0x1fd4, 0x1fd5,
0x1fd6, 0x1fd7, 0x1fd8, 0x1fd9, 0x1fda, 0x1fdb, 0x1fdc, 0x1fdd,
0x1fde, 0x1fdf, 0x1fe0, 0x1fe1, 0x1fe2, 0x1fe3, 0x1fe4, 0x1fe5,
0x1fe6, 0x1fe7, 0x07f2, 0x00fd, 0x003e, 0x000e, 0x0006, 0x0000,
0x0002, 0x001e, 0x00fc, 0x03f8, 0x1fcc, 0x1fe8, 0x1fe9, 0x1fea,
0x1feb, 0x1fec, 0x1fcd, 0x1fed, 0x1fee, 0x1fef, 0x1ff0, 0x1ff1,
0x1ff2, 0x1ff3, 0x1ff4, 0x1ff5, 0x1ff6, 0x1ff7, 0x1ff8, 0x1ff9,
0x1ffa, 0x1ffb, 0x1ffc, 0x1ffd, 0x1ffe, 0x3ffe, 0x3fff,
};
static const uint8_t t_huffman_noise_bal_3_0dB_bits[25] = {
8, 8, 8, 8, 8, 8, 8, 8,
8, 8, 5, 2, 1, 3, 6, 8,
8, 8, 8, 8, 8, 8, 8, 8,
8,
};
static const uint8_t t_huffman_noise_bal_3_0dB_codes[25] = {
0xec, 0xed, 0xee, 0xef, 0xf0, 0xf1, 0xf2, 0xf3,
0xf4, 0xf5, 0x1c, 0x02, 0x00, 0x06, 0x3a, 0xf6,
0xf7, 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe,
0xff,
};
static const int8_t sbr_offset[6][16] = {
{-8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7}, // fs_sbr = 16000 Hz
{-5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13}, // fs_sbr = 22050 Hz
{-5, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}, // fs_sbr = 24000 Hz
{-6, -4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}, // fs_sbr = 32000 Hz
{-4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20}, // 44100 Hz <= fs_sbr <= 64000 Hz
{-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24}, // 64000 Hz < fs_sbr
};
///< window coefficients for analysis/synthesis QMF banks
static DECLARE_ALIGNED(16, float, sbr_qmf_window_ds)[320];
static DECLARE_ALIGNED(16, float, sbr_qmf_window_us)[640] = {
0.0000000000, -0.0005525286, -0.0005617692, -0.0004947518,
-0.0004875227, -0.0004893791, -0.0005040714, -0.0005226564,
-0.0005466565, -0.0005677802, -0.0005870930, -0.0006132747,
-0.0006312493, -0.0006540333, -0.0006777690, -0.0006941614,
-0.0007157736, -0.0007255043, -0.0007440941, -0.0007490598,
-0.0007681371, -0.0007724848, -0.0007834332, -0.0007779869,
-0.0007803664, -0.0007801449, -0.0007757977, -0.0007630793,
-0.0007530001, -0.0007319357, -0.0007215391, -0.0006917937,
-0.0006650415, -0.0006341594, -0.0005946118, -0.0005564576,
-0.0005145572, -0.0004606325, -0.0004095121, -0.0003501175,
-0.0002896981, -0.0002098337, -0.0001446380, -0.0000617334,
0.0000134949, 0.0001094383, 0.0002043017, 0.0002949531,
0.0004026540, 0.0005107388, 0.0006239376, 0.0007458025,
0.0008608443, 0.0009885988, 0.0011250155, 0.0012577884,
0.0013902494, 0.0015443219, 0.0016868083, 0.0018348265,
0.0019841140, 0.0021461583, 0.0023017254, 0.0024625616,
0.0026201758, 0.0027870464, 0.0029469447, 0.0031125420,
0.0032739613, 0.0034418874, 0.0036008268, 0.0037603922,
0.0039207432, 0.0040819753, 0.0042264269, 0.0043730719,
0.0045209852, 0.0046606460, 0.0047932560, 0.0049137603,
0.0050393022, 0.0051407353, 0.0052461166, 0.0053471681,
0.0054196775, 0.0054876040, 0.0055475714, 0.0055938023,
0.0056220643, 0.0056455196, 0.0056389199, 0.0056266114,
0.0055917128, 0.0055404363, 0.0054753783, 0.0053838975,
0.0052715758, 0.0051382275, 0.0049839687, 0.0048109469,
0.0046039530, 0.0043801861, 0.0041251642, 0.0038456408,
0.0035401246, 0.0032091885, 0.0028446757, 0.0024508540,
0.0020274176, 0.0015784682, 0.0010902329, 0.0005832264,
0.0000276045, -0.0005464280, -0.0011568135, -0.0018039472,
-0.0024826723, -0.0031933778, -0.0039401124, -0.0047222596,
-0.0055337211, -0.0063792293, -0.0072615816, -0.0081798233,
-0.0091325329, -0.0101150215, -0.0111315548, -0.0121849995,
0.0132718220, 0.0143904666, 0.0155405553, 0.0167324712,
0.0179433381, 0.0191872431, 0.0204531793, 0.0217467550,
0.0230680169, 0.0244160992, 0.0257875847, 0.0271859429,
0.0286072173, 0.0300502657, 0.0315017608, 0.0329754081,
0.0344620948, 0.0359697560, 0.0374812850, 0.0390053679,
0.0405349170, 0.0420649094, 0.0436097542, 0.0451488405,
0.0466843027, 0.0482165720, 0.0497385755, 0.0512556155,
0.0527630746, 0.0542452768, 0.0557173648, 0.0571616450,
0.0585915683, 0.0599837480, 0.0613455171, 0.0626857808,
0.0639715898, 0.0652247106, 0.0664367512, 0.0676075985,
0.0687043828, 0.0697630244, 0.0707628710, 0.0717002673,
0.0725682583, 0.0733620255, 0.0741003642, 0.0747452558,
0.0753137336, 0.0758008358, 0.0761992479, 0.0764992170,
0.0767093490, 0.0768173975, 0.0768230011, 0.0767204924,
0.0765050718, 0.0761748321, 0.0757305756, 0.0751576255,
0.0744664394, 0.0736406005, 0.0726774642, 0.0715826364,
0.0703533073, 0.0689664013, 0.0674525021, 0.0657690668,
0.0639444805, 0.0619602779, 0.0598166570, 0.0575152691,
0.0550460034, 0.0524093821, 0.0495978676, 0.0466303305,
0.0434768782, 0.0401458278, 0.0366418116, 0.0329583930,
0.0290824006, 0.0250307561, 0.0207997072, 0.0163701258,
0.0117623832, 0.0069636862, 0.0019765601, -0.0032086896,
-0.0085711749, -0.0141288827, -0.0198834129, -0.0258227288,
-0.0319531274, -0.0382776572, -0.0447806821, -0.0514804176,
-0.0583705326, -0.0654409853, -0.0726943300, -0.0801372934,
-0.0877547536, -0.0955533352, -0.1035329531, -0.1116826931,
-0.1200077984, -0.1285002850, -0.1371551761, -0.1459766491,
-0.1549607071, -0.1640958855, -0.1733808172, -0.1828172548,
-0.1923966745, -0.2021250176, -0.2119735853, -0.2219652696,
-0.2320690870, -0.2423016884, -0.2526480309, -0.2631053299,
-0.2736634040, -0.2843214189, -0.2950716717, -0.3059098575,
-0.3168278913, -0.3278113727, -0.3388722693, -0.3499914122,
0.3611589903, 0.3723795546, 0.3836350013, 0.3949211761,
0.4062317676, 0.4175696896, 0.4289119920, 0.4402553754,
0.4515996535, 0.4629308085, 0.4742453214, 0.4855253091,
0.4967708254, 0.5079817500, 0.5191234970, 0.5302240895,
0.5412553448, 0.5522051258, 0.5630789140, 0.5738524131,
0.5845403235, 0.5951123086, 0.6055783538, 0.6159109932,
0.6261242695, 0.6361980107, 0.6461269695, 0.6559016302,
0.6655139880, 0.6749663190, 0.6842353293, 0.6933282376,
0.7022388719, 0.7109410426, 0.7194462634, 0.7277448900,
0.7358211758, 0.7436827863, 0.7513137456, 0.7587080760,
0.7658674865, 0.7727780881, 0.7794287519, 0.7858353120,
0.7919735841, 0.7978466413, 0.8034485751, 0.8087695004,
0.8138191270, 0.8185776004, 0.8230419890, 0.8272275347,
0.8311038457, 0.8346937361, 0.8379717337, 0.8409541392,
0.8436238281, 0.8459818469, 0.8480315777, 0.8497805198,
0.8511971524, 0.8523047035, 0.8531020949, 0.8535720573,
0.8537385600,
};
static const float sbr_noise_table[512][2] = {
{-0.99948153278296, -0.59483417516607}, { 0.97113454393991, -0.67528515225647},
{ 0.14130051758487, -0.95090983575689}, {-0.47005496701697, -0.37340549728647},
{ 0.80705063769351, 0.29653668284408}, {-0.38981478896926, 0.89572605717087},
{-0.01053049862020, -0.66959058036166}, {-0.91266367957293, -0.11522938140034},
{ 0.54840422910309, 0.75221367176302}, { 0.40009252867955, -0.98929400334421},
{-0.99867974711855, -0.88147068645358}, {-0.95531076805040, 0.90908757154593},
{-0.45725933317144, -0.56716323646760}, {-0.72929675029275, -0.98008272727324},
{ 0.75622801399036, 0.20950329995549}, { 0.07069442601050, -0.78247898470706},
{ 0.74496252926055, -0.91169004445807}, {-0.96440182703856, -0.94739918296622},
{ 0.30424629369539, -0.49438267012479}, { 0.66565033746925, 0.64652935542491},
{ 0.91697008020594, 0.17514097332009}, {-0.70774918760427, 0.52548653416543},
{-0.70051415345560, -0.45340028808763}, {-0.99496513054797, -0.90071908066973},
{ 0.98164490790123, -0.77463155528697}, {-0.54671580548181, -0.02570928536004},
{-0.01689629065389, 0.00287506445732}, {-0.86110349531986, 0.42548583726477},
{-0.98892980586032, -0.87881132267556}, { 0.51756627678691, 0.66926784710139},
{-0.99635026409640, -0.58107730574765}, {-0.99969370862163, 0.98369989360250},
{ 0.55266258627194, 0.59449057465591}, { 0.34581177741673, 0.94879421061866},
{ 0.62664209577999, -0.74402970906471}, {-0.77149701404973, -0.33883658042801},
{-0.91592244254432, 0.03687901376713}, {-0.76285492357887, -0.91371867919124},
{ 0.79788337195331, -0.93180971199849}, { 0.54473080610200, -0.11919206037186},
{-0.85639281671058, 0.42429854760451}, {-0.92882402971423, 0.27871809078609},
{-0.11708371046774, -0.99800843444966}, { 0.21356749817493, -0.90716295627033},
{-0.76191692573909, 0.99768118356265}, { 0.98111043100884, -0.95854459734407},
{-0.85913269895572, 0.95766566168880}, {-0.93307242253692, 0.49431757696466},
{ 0.30485754879632, -0.70540034357529}, { 0.85289650925190, 0.46766131791044},
{ 0.91328082618125, -0.99839597361769}, {-0.05890199924154, 0.70741827819497},
{ 0.28398686150148, 0.34633555702188}, { 0.95258164539612, -0.54893416026939},
{-0.78566324168507, -0.75568541079691}, {-0.95789495447877, -0.20423194696966},
{ 0.82411158711197, 0.96654618432562}, {-0.65185446735885, -0.88734990773289},
{-0.93643603134666, 0.99870790442385}, { 0.91427159529618, -0.98290505544444},
{-0.70395684036886, 0.58796798221039}, { 0.00563771969365, 0.61768196727244},
{ 0.89065051931895, 0.52783352697585}, {-0.68683707712762, 0.80806944710339},
{ 0.72165342518718, -0.69259857349564}, {-0.62928247730667, 0.13627037407335},
{ 0.29938434065514, -0.46051329682246}, {-0.91781958879280, -0.74012716684186},
{ 0.99298717043688, 0.40816610075661}, { 0.82368298622748, -0.74036047190173},
{-0.98512833386833, -0.99972330709594}, {-0.95915368242257, -0.99237800466040},
{-0.21411126572790, -0.93424819052545}, {-0.68821476106884, -0.26892306315457},
{ 0.91851997982317, 0.09358228901785}, {-0.96062769559127, 0.36099095133739},
{ 0.51646184922287, -0.71373332873917}, { 0.61130721139669, 0.46950141175917},
{ 0.47336129371299, -0.27333178296162}, { 0.90998308703519, 0.96715662938132},
{ 0.44844799194357, 0.99211574628306}, { 0.66614891079092, 0.96590176169121},
{ 0.74922239129237, -0.89879858826087}, {-0.99571588506485, 0.52785521494349},
{ 0.97401082477563, -0.16855870075190}, { 0.72683747733879, -0.48060774432251},
{ 0.95432193457128, 0.68849603408441}, {-0.72962208425191, -0.76608443420917},
{-0.85359479233537, 0.88738125901579}, {-0.81412430338535, -0.97480768049637},
{-0.87930772356786, 0.74748307690436}, {-0.71573331064977, -0.98570608178923},
{ 0.83524300028228, 0.83702537075163}, {-0.48086065601423, -0.98848504923531},
{ 0.97139128574778, 0.80093621198236}, { 0.51992825347895, 0.80247631400510},
{-0.00848591195325, -0.76670128000486}, {-0.70294374303036, 0.55359910445577},
{-0.95894428168140, -0.43265504344783}, { 0.97079252950321, 0.09325857238682},
{-0.92404293670797, 0.85507704027855}, {-0.69506469500450, 0.98633412625459},
{ 0.26559203620024, 0.73314307966524}, { 0.28038443336943, 0.14537913654427},
{-0.74138124825523, 0.99310339807762}, {-0.01752795995444, -0.82616635284178},
{-0.55126773094930, -0.98898543862153}, { 0.97960898850996, -0.94021446752851},
{-0.99196309146936, 0.67019017358456}, {-0.67684928085260, 0.12631491649378},
{ 0.09140039465500, -0.20537731453108}, {-0.71658965751996, -0.97788200391224},
{ 0.81014640078925, 0.53722648362443}, { 0.40616991671205, -0.26469008598449},
{-0.67680188682972, 0.94502052337695}, { 0.86849774348749, -0.18333598647899},
{-0.99500381284851, -0.02634122068550}, { 0.84329189340667, 0.10406957462213},
{-0.09215968531446, 0.69540012101253}, { 0.99956173327206, -0.12358542001404},
{-0.79732779473535, -0.91582524736159}, { 0.96349973642406, 0.96640458041000},
{-0.79942778496547, 0.64323902822857}, {-0.11566039853896, 0.28587846253726},
{-0.39922954514662, 0.94129601616966}, { 0.99089197565987, -0.92062625581587},
{ 0.28631285179909, -0.91035047143603}, {-0.83302725605608, -0.67330410892084},
{ 0.95404443402072, 0.49162765398743}, {-0.06449863579434, 0.03250560813135},
{-0.99575054486311, 0.42389784469507}, {-0.65501142790847, 0.82546114655624},
{-0.81254441908887, -0.51627234660629}, {-0.99646369485481, 0.84490533520752},
{ 0.00287840603348, 0.64768261158166}, { 0.70176989408455, -0.20453028573322},
{ 0.96361882270190, 0.40706967140989}, {-0.68883758192426, 0.91338958840772},
{-0.34875585502238, 0.71472290693300}, { 0.91980081243087, 0.66507455644919},
{-0.99009048343881, 0.85868021604848}, { 0.68865791458395, 0.55660316809678},
{-0.99484402129368, -0.20052559254934}, { 0.94214511408023, -0.99696425367461},
{-0.67414626793544, 0.49548221180078}, {-0.47339353684664, -0.85904328834047},
{ 0.14323651387360, -0.94145598222488}, {-0.29268293575672, 0.05759224927952},
{ 0.43793861458754, -0.78904969892724}, {-0.36345126374441, 0.64874435357162},
{-0.08750604656825, 0.97686944362527}, {-0.96495267812511, -0.53960305946511},
{ 0.55526940659947, 0.78891523734774}, { 0.73538215752630, 0.96452072373404},
{-0.30889773919437, -0.80664389776860}, { 0.03574995626194, -0.97325616900959},
{ 0.98720684660488, 0.48409133691962}, {-0.81689296271203, -0.90827703628298},
{ 0.67866860118215, 0.81284503870856}, {-0.15808569732583, 0.85279555024382},
{ 0.80723395114371, -0.24717418514605}, { 0.47788757329038, -0.46333147839295},
{ 0.96367554763201, 0.38486749303242}, {-0.99143875716818, -0.24945277239809},
{ 0.83081876925833, -0.94780851414763}, {-0.58753191905341, 0.01290772389163},
{ 0.95538108220960, -0.85557052096538}, {-0.96490920476211, -0.64020970923102},
{-0.97327101028521, 0.12378128133110}, { 0.91400366022124, 0.57972471346930},
{-0.99925837363824, 0.71084847864067}, {-0.86875903507313, -0.20291699203564},
{-0.26240034795124, -0.68264554369108}, {-0.24664412953388, -0.87642273115183},
{ 0.02416275806869, 0.27192914288905}, { 0.82068619590515, -0.85087787994476},
{ 0.88547373760759, -0.89636802901469}, {-0.18173078152226, -0.26152145156800},
{ 0.09355476558534, 0.54845123045604}, {-0.54668414224090, 0.95980774020221},
{ 0.37050990604091, -0.59910140383171}, {-0.70373594262891, 0.91227665827081},
{-0.34600785879594, -0.99441426144200}, {-0.68774481731008, -0.30238837956299},
{-0.26843291251234, 0.83115668004362}, { 0.49072334613242, -0.45359708737775},
{ 0.38975993093975, 0.95515358099121}, {-0.97757125224150, 0.05305894580606},
{-0.17325552859616, -0.92770672250494}, { 0.99948035025744, 0.58285545563426},
{-0.64946246527458, 0.68645507104960}, {-0.12016920576437, -0.57147322153312},
{-0.58947456517751, -0.34847132454388}, {-0.41815140454465, 0.16276422358861},
{ 0.99885650204884, 0.11136095490444}, {-0.56649614128386, -0.90494866361587},
{ 0.94138021032330, 0.35281916733018}, {-0.75725076534641, 0.53650549640587},
{ 0.20541973692630, -0.94435144369918}, { 0.99980371023351, 0.79835913565599},
{ 0.29078277605775, 0.35393777921520}, {-0.62858772103030, 0.38765693387102},
{ 0.43440904467688, -0.98546330463232}, {-0.98298583762390, 0.21021524625209},
{ 0.19513029146934, -0.94239832251867}, {-0.95476662400101, 0.98364554179143},
{ 0.93379635304810, -0.70881994583682}, {-0.85235410573336, -0.08342347966410},
{-0.86425093011245, -0.45795025029466}, { 0.38879779059045, 0.97274429344593},
{ 0.92045124735495, -0.62433652524220}, { 0.89162532251878, 0.54950955570563},
{-0.36834336949252, 0.96458298020975}, { 0.93891760988045, -0.89968353740388},
{ 0.99267657565094, -0.03757034316958}, {-0.94063471614176, 0.41332338538963},
{ 0.99740224117019, -0.16830494996370}, {-0.35899413170555, -0.46633226649613},
{ 0.05237237274947, -0.25640361602661}, { 0.36703583957424, -0.38653265641875},
{ 0.91653180367913, -0.30587628726597}, { 0.69000803499316, 0.90952171386132},
{-0.38658751133527, 0.99501571208985}, {-0.29250814029851, 0.37444994344615},
{-0.60182204677608, 0.86779651036123}, {-0.97418588163217, 0.96468523666475},
{ 0.88461574003963, 0.57508405276414}, { 0.05198933055162, 0.21269661669964},
{-0.53499621979720, 0.97241553731237}, {-0.49429560226497, 0.98183865291903},
{-0.98935142339139, -0.40249159006933}, {-0.98081380091130, -0.72856895534041},
{-0.27338148835532, 0.99950922447209}, { 0.06310802338302, -0.54539587529618},
{-0.20461677199539, -0.14209977628489}, { 0.66223843141647, 0.72528579940326},
{-0.84764345483665, 0.02372316801261}, {-0.89039863483811, 0.88866581484602},
{ 0.95903308477986, 0.76744927173873}, { 0.73504123909879, -0.03747203173192},
{-0.31744434966056, -0.36834111883652}, {-0.34110827591623, 0.40211222807691},
{ 0.47803883714199, -0.39423219786288}, { 0.98299195879514, 0.01989791390047},
{-0.30963073129751, -0.18076720599336}, { 0.99992588229018, -0.26281872094289},
{-0.93149731080767, -0.98313162570490}, { 0.99923472302773, -0.80142993767554},
{-0.26024169633417, -0.75999759855752}, {-0.35712514743563, 0.19298963768574},
{-0.99899084509530, 0.74645156992493}, { 0.86557171579452, 0.55593866696299},
{ 0.33408042438752, 0.86185953874709}, { 0.99010736374716, 0.04602397576623},
{-0.66694269691195, -0.91643611810148}, { 0.64016792079480, 0.15649530836856},
{ 0.99570534804836, 0.45844586038111}, {-0.63431466947340, 0.21079116459234},
{-0.07706847005931, -0.89581437101329}, { 0.98590090577724, 0.88241721133981},
{ 0.80099335254678, -0.36851896710853}, { 0.78368131392666, 0.45506999802597},
{ 0.08707806671691, 0.80938994918745}, {-0.86811883080712, 0.39347308654705},
{-0.39466529740375, -0.66809432114456}, { 0.97875325649683, -0.72467840967746},
{-0.95038560288864, 0.89563219587625}, { 0.17005239424212, 0.54683053962658},
{-0.76910792026848, -0.96226617549298}, { 0.99743281016846, 0.42697157037567},
{ 0.95437383549973, 0.97002324109952}, { 0.99578905365569, -0.54106826257356},
{ 0.28058259829990, -0.85361420634036}, { 0.85256524470573, -0.64567607735589},
{-0.50608540105128, -0.65846015480300}, {-0.97210735183243, -0.23095213067791},
{ 0.95424048234441, -0.99240147091219}, {-0.96926570524023, 0.73775654896574},
{ 0.30872163214726, 0.41514960556126}, {-0.24523839572639, 0.63206633394807},
{-0.33813265086024, -0.38661779441897}, {-0.05826828420146, -0.06940774188029},
{-0.22898461455054, 0.97054853316316}, {-0.18509915019881, 0.47565762892084},
{-0.10488238045009, -0.87769947402394}, {-0.71886586182037, 0.78030982480538},
{ 0.99793873738654, 0.90041310491497}, { 0.57563307626120, -0.91034337352097},
{ 0.28909646383717, 0.96307783970534}, { 0.42188998312520, 0.48148651230437},
{ 0.93335049681047, -0.43537023883588}, {-0.97087374418267, 0.86636445711364},
{ 0.36722871286923, 0.65291654172961}, {-0.81093025665696, 0.08778370229363},
{-0.26240603062237, -0.92774095379098}, { 0.83996497984604, 0.55839849139647},
{-0.99909615720225, -0.96024605713970}, { 0.74649464155061, 0.12144893606462},
{-0.74774595569805, -0.26898062008959}, { 0.95781667469567, -0.79047927052628},
{ 0.95472308713099, -0.08588776019550}, { 0.48708332746299, 0.99999041579432},
{ 0.46332038247497, 0.10964126185063}, {-0.76497004940162, 0.89210929242238},
{ 0.57397389364339, 0.35289703373760}, { 0.75374316974495, 0.96705214651335},
{-0.59174397685714, -0.89405370422752}, { 0.75087906691890, -0.29612672982396},
{-0.98607857336230, 0.25034911730023}, {-0.40761056640505, -0.90045573444695},
{ 0.66929266740477, 0.98629493401748}, {-0.97463695257310, -0.00190223301301},
{ 0.90145509409859, 0.99781390365446}, {-0.87259289048043, 0.99233587353666},
{-0.91529461447692, -0.15698707534206}, {-0.03305738840705, -0.37205262859764},
{ 0.07223051368337, -0.88805001733626}, { 0.99498012188353, 0.97094358113387},
{-0.74904939500519, 0.99985483641521}, { 0.04585228574211, 0.99812337444082},
{-0.89054954257993, -0.31791913188064}, {-0.83782144651251, 0.97637632547466},
{ 0.33454804933804, -0.86231516800408}, {-0.99707579362824, 0.93237990079441},
{-0.22827527843994, 0.18874759397997}, { 0.67248046289143, -0.03646211390569},
{-0.05146538187944, -0.92599700120679}, { 0.99947295749905, 0.93625229707912},
{ 0.66951124390363, 0.98905825623893}, {-0.99602956559179, -0.44654715757688},
{ 0.82104905483590, 0.99540741724928}, { 0.99186510988782, 0.72023001312947},
{-0.65284592392918, 0.52186723253637}, { 0.93885443798188, -0.74895312615259},
{ 0.96735248738388, 0.90891816978629}, {-0.22225968841114, 0.57124029781228},
{-0.44132783753414, -0.92688840659280}, {-0.85694974219574, 0.88844532719844},
{ 0.91783042091762, -0.46356892383970}, { 0.72556974415690, -0.99899555770747},
{-0.99711581834508, 0.58211560180426}, { 0.77638976371966, 0.94321834873819},
{ 0.07717324253925, 0.58638399856595}, {-0.56049829194163, 0.82522301569036},
{ 0.98398893639988, 0.39467440420569}, { 0.47546946844938, 0.68613044836811},
{ 0.65675089314631, 0.18331637134880}, { 0.03273375457980, -0.74933109564108},
{-0.38684144784738, 0.51337349030406}, {-0.97346267944545, -0.96549364384098},
{-0.53282156061942, -0.91423265091354}, { 0.99817310731176, 0.61133572482148},
{-0.50254500772635, -0.88829338134294}, { 0.01995873238855, 0.85223515096765},
{ 0.99930381973804, 0.94578896296649}, { 0.82907767600783, -0.06323442598128},
{-0.58660709669728, 0.96840773806582}, {-0.17573736667267, -0.48166920859485},
{ 0.83434292401346, -0.13023450646997}, { 0.05946491307025, 0.20511047074866},
{ 0.81505484574602, -0.94685947861369}, {-0.44976380954860, 0.40894572671545},
{-0.89746474625671, 0.99846578838537}, { 0.39677256130792, -0.74854668609359},
{-0.07588948563079, 0.74096214084170}, { 0.76343198951445, 0.41746629422634},
{-0.74490104699626, 0.94725911744610}, { 0.64880119792759, 0.41336660830571},
{ 0.62319537462542, -0.93098313552599}, { 0.42215817594807, -0.07712787385208},
{ 0.02704554141885, -0.05417518053666}, { 0.80001773566818, 0.91542195141039},
{-0.79351832348816, -0.36208897989136}, { 0.63872359151636, 0.08128252493444},
{ 0.52890520960295, 0.60048872455592}, { 0.74238552914587, 0.04491915291044},
{ 0.99096131449250, -0.19451182854402}, {-0.80412329643109, -0.88513818199457},
{-0.64612616129736, 0.72198674804544}, { 0.11657770663191, -0.83662833815041},
{-0.95053182488101, -0.96939905138082}, {-0.62228872928622, 0.82767262846661},
{ 0.03004475787316, -0.99738896333384}, {-0.97987214341034, 0.36526129686425},
{-0.99986980746200, -0.36021610299715}, { 0.89110648599879, -0.97894250343044},
{ 0.10407960510582, 0.77357793811619}, { 0.95964737821728, -0.35435818285502},
{ 0.50843233159162, 0.96107691266205}, { 0.17006334670615, -0.76854025314829},
{ 0.25872675063360, 0.99893303933816}, {-0.01115998681937, 0.98496019742444},
{-0.79598702973261, 0.97138411318894}, {-0.99264708948101, -0.99542822402536},
{-0.99829663752818, 0.01877138824311}, {-0.70801016548184, 0.33680685948117},
{-0.70467057786826, 0.93272777501857}, { 0.99846021905254, -0.98725746254433},
{-0.63364968534650, -0.16473594423746}, {-0.16258217500792, -0.95939125400802},
{-0.43645594360633, -0.94805030113284}, {-0.99848471702976, 0.96245166923809},
{-0.16796458968998, -0.98987511890470}, {-0.87979225745213, -0.71725725041680},
{ 0.44183099021786, -0.93568974498761}, { 0.93310180125532, -0.99913308068246},
{-0.93941931782002, -0.56409379640356}, {-0.88590003188677, 0.47624600491382},
{ 0.99971463703691, -0.83889954253462}, {-0.75376385639978, 0.00814643438625},
{ 0.93887685615875, -0.11284528204636}, { 0.85126435782309, 0.52349251543547},
{ 0.39701421446381, 0.81779634174316}, {-0.37024464187437, -0.87071656222959},
{-0.36024828242896, 0.34655735648287}, {-0.93388812549209, -0.84476541096429},
{-0.65298804552119, -0.18439575450921}, { 0.11960319006843, 0.99899346780168},
{ 0.94292565553160, 0.83163906518293}, { 0.75081145286948, -0.35533223142265},
{ 0.56721979748394, -0.24076836414499}, { 0.46857766746029, -0.30140233457198},
{ 0.97312313923635, -0.99548191630031}, {-0.38299976567017, 0.98516909715427},
{ 0.41025800019463, 0.02116736935734}, { 0.09638062008048, 0.04411984381457},
{-0.85283249275397, 0.91475563922421}, { 0.88866808958124, -0.99735267083226},
{-0.48202429536989, -0.96805608884164}, { 0.27572582416567, 0.58634753335832},
{-0.65889129659168, 0.58835634138583}, { 0.98838086953732, 0.99994349600236},
{-0.20651349620689, 0.54593044066355}, {-0.62126416356920, -0.59893681700392},
{ 0.20320105410437, -0.86879180355289}, {-0.97790548600584, 0.96290806999242},
{ 0.11112534735126, 0.21484763313301}, {-0.41368337314182, 0.28216837680365},
{ 0.24133038992960, 0.51294362630238}, {-0.66393410674885, -0.08249679629081},
{-0.53697829178752, -0.97649903936228}, {-0.97224737889348, 0.22081333579837},
{ 0.87392477144549, -0.12796173740361}, { 0.19050361015753, 0.01602615387195},
{-0.46353441212724, -0.95249041539006}, {-0.07064096339021, -0.94479803205886},
{-0.92444085484466, -0.10457590187436}, {-0.83822593578728, -0.01695043208885},
{ 0.75214681811150, -0.99955681042665}, {-0.42102998829339, 0.99720941999394},
{-0.72094786237696, -0.35008961934255}, { 0.78843311019251, 0.52851398958271},
{ 0.97394027897442, -0.26695944086561}, { 0.99206463477946, -0.57010120849429},
{ 0.76789609461795, -0.76519356730966}, {-0.82002421836409, -0.73530179553767},
{ 0.81924990025724, 0.99698425250579}, {-0.26719850873357, 0.68903369776193},
{-0.43311260380975, 0.85321815947490}, { 0.99194979673836, 0.91876249766422},
{-0.80692001248487, -0.32627540663214}, { 0.43080003649976, -0.21919095636638},
{ 0.67709491937357, -0.95478075822906}, { 0.56151770568316, -0.70693811747778},
{ 0.10831862810749, -0.08628837174592}, { 0.91229417540436, -0.65987351408410},
{-0.48972893932274, 0.56289246362686}, {-0.89033658689697, -0.71656563987082},
{ 0.65269447475094, 0.65916004833932}, { 0.67439478141121, -0.81684380846796},
{-0.47770832416973, -0.16789556203025}, {-0.99715979260878, -0.93565784007648},
{-0.90889593602546, 0.62034397054380}, {-0.06618622548177, -0.23812217221359},
{ 0.99430266919728, 0.18812555317553}, { 0.97686402381843, -0.28664534366620},
{ 0.94813650221268, -0.97506640027128}, {-0.95434497492853, -0.79607978501983},
{-0.49104783137150, 0.32895214359663}, { 0.99881175120751, 0.88993983831354},
{ 0.50449166760303, -0.85995072408434}, { 0.47162891065108, -0.18680204049569},
{-0.62081581361840, 0.75000676218956}, {-0.43867015250812, 0.99998069244322},
{ 0.98630563232075, -0.53578899600662}, {-0.61510362277374, -0.89515019899997},
{-0.03841517601843, -0.69888815681179}, {-0.30102157304644, -0.07667808922205},
{ 0.41881284182683, 0.02188098922282}, {-0.86135454941237, 0.98947480909359},
{ 0.67226861393788, -0.13494389011014}, {-0.70737398842068, -0.76547349325992},
{ 0.94044946687963, 0.09026201157416}, {-0.82386352534327, 0.08924768823676},
{-0.32070666698656, 0.50143421908753}, { 0.57593163224487, -0.98966422921509},
{-0.36326018419965, 0.07440243123228}, { 0.99979044674350, -0.14130287347405},
{-0.92366023326932, -0.97979298068180}, {-0.44607178518598, -0.54233252016394},
{ 0.44226800932956, 0.71326756742752}, { 0.03671907158312, 0.63606389366675},
{ 0.52175424682195, -0.85396826735705}, {-0.94701139690956, -0.01826348194255},
{-0.98759606946049, 0.82288714303073}, { 0.87434794743625, 0.89399495655433},
{-0.93412041758744, 0.41374052024363}, { 0.96063943315511, 0.93116709541280},
{ 0.97534253457837, 0.86150930812689}, { 0.99642466504163, 0.70190043427512},
{-0.94705089665984, -0.29580042814306}, { 0.91599807087376, -0.98147830385781},
};
#endif /* AVCODEC_AACSBRDATA_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aactab.c
-1207
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src/add-ons/media/plugins/ffmpeg/libavcodec/aactab.h
-77
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@@ -1,77 +0,0 @@
/*
* AAC data declarations
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC data declarations
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef AVCODEC_AACTAB_H
#define AVCODEC_AACTAB_H
#include "libavutil/mem.h"
#include "aac.h"
#include "aac_tablegen_decl.h"
#include <stdint.h>
/* NOTE:
* Tables in this file are used by the AAC decoder and will be used by the AAC
* encoder.
*/
/* @name window coefficients
* @{
*/
DECLARE_ALIGNED(16, extern float, ff_aac_kbd_long_1024)[1024];
DECLARE_ALIGNED(16, extern float, ff_aac_kbd_short_128)[128];
// @}
/* @name number of scalefactor window bands for long and short transform windows respectively
* @{
*/
extern const uint8_t ff_aac_num_swb_1024[];
extern const uint8_t ff_aac_num_swb_128 [];
// @}
extern const uint8_t ff_aac_pred_sfb_max [];
extern const uint32_t ff_aac_scalefactor_code[121];
extern const uint8_t ff_aac_scalefactor_bits[121];
extern const uint16_t * const ff_aac_spectral_codes[11];
extern const uint8_t * const ff_aac_spectral_bits [11];
extern const uint16_t ff_aac_spectral_sizes[11];
extern const float *ff_aac_codebook_vectors[];
extern const float *ff_aac_codebook_vector_vals[];
extern const uint16_t *ff_aac_codebook_vector_idx[];
extern const uint16_t * const ff_swb_offset_1024[13];
extern const uint16_t * const ff_swb_offset_128 [13];
extern const uint8_t ff_tns_max_bands_1024[13];
extern const uint8_t ff_tns_max_bands_128 [13];
#endif /* AVCODEC_AACTAB_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aandcttab.c
-47
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@@ -1,47 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAN (Arai Agui Aakajima) (I)DCT tables
*/
#include <stdint.h>
const uint16_t ff_aanscales[64] = {
/* precomputed values scaled up by 14 bits */
16384, 22725, 21407, 19266, 16384, 12873, 8867, 4520,
22725, 31521, 29692, 26722, 22725, 17855, 12299, 6270,
21407, 29692, 27969, 25172, 21407, 16819, 11585, 5906,
19266, 26722, 25172, 22654, 19266, 15137, 10426, 5315,
16384, 22725, 21407, 19266, 16384, 12873, 8867, 4520,
12873, 17855, 16819, 15137, 12873, 10114, 6967, 3552,
8867 , 12299, 11585, 10426, 8867, 6967, 4799, 2446,
4520 , 6270, 5906, 5315, 4520, 3552, 2446, 1247
};
const uint16_t ff_inv_aanscales[64] = {
4096, 2953, 3135, 3483, 4096, 5213, 7568, 14846,
2953, 2129, 2260, 2511, 2953, 3759, 5457, 10703,
3135, 2260, 2399, 2666, 3135, 3990, 5793, 11363,
3483, 2511, 2666, 2962, 3483, 4433, 6436, 12625,
4096, 2953, 3135, 3483, 4096, 5213, 7568, 14846,
5213, 3759, 3990, 4433, 5213, 6635, 9633, 18895,
7568, 5457, 5793, 6436, 7568, 9633, 13985, 27432,
14846, 10703, 11363, 12625, 14846, 18895, 27432, 53809,
};
src/add-ons/media/plugins/ffmpeg/libavcodec/aandcttab.h
-32
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@@ -1,32 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAN (Arai Agui Nakajima) (I)DCT tables
*/
#ifndef AVCODEC_AANDCTTAB_H
#define AVCODEC_AANDCTTAB_H
#include <stdint.h>
extern const uint16_t ff_aanscales[64];
extern const uint16_t ff_inv_aanscales[64];
#endif /* AVCODEC_AANDCTTAB_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aasc.c
-123
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@@ -1,123 +0,0 @@
/*
* Autodesk RLE Decoder
* Copyright (C) 2005 the ffmpeg project
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Autodesk RLE Video Decoder by Konstantin Shishkov
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "avcodec.h"
#include "dsputil.h"
#include "msrledec.h"
typedef struct AascContext {
AVCodecContext *avctx;
AVFrame frame;
} AascContext;
#define FETCH_NEXT_STREAM_BYTE() \
if (stream_ptr >= buf_size) \
{ \
av_log(s->avctx, AV_LOG_ERROR, " AASC: stream ptr just went out of bounds (fetch)\n"); \
break; \
} \
stream_byte = buf[stream_ptr++];
static av_cold int aasc_decode_init(AVCodecContext *avctx)
{
AascContext *s = avctx->priv_data;
s->avctx = avctx;
avctx->pix_fmt = PIX_FMT_BGR24;
return 0;
}
static int aasc_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AascContext *s = avctx->priv_data;
int compr, i, stride;
s->frame.reference = 1;
s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
if (avctx->reget_buffer(avctx, &s->frame)) {
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
return -1;
}
compr = AV_RL32(buf);
buf += 4;
buf_size -= 4;
switch(compr){
case 0:
stride = (avctx->width * 3 + 3) & ~3;
for(i = avctx->height - 1; i >= 0; i--){
memcpy(s->frame.data[0] + i*s->frame.linesize[0], buf, avctx->width*3);
buf += stride;
}
break;
case 1:
ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, buf - 4, buf_size + 4);
break;
default:
av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d\n", compr);
return -1;
}
*data_size = sizeof(AVFrame);
*(AVFrame*)data = s->frame;
/* report that the buffer was completely consumed */
return buf_size;
}
static av_cold int aasc_decode_end(AVCodecContext *avctx)
{
AascContext *s = avctx->priv_data;
/* release the last frame */
if (s->frame.data[0])
avctx->release_buffer(avctx, &s->frame);
return 0;
}
AVCodec aasc_decoder = {
"aasc",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_AASC,
sizeof(AascContext),
aasc_decode_init,
NULL,
aasc_decode_end,
aasc_decode_frame,
CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Autodesk RLE"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3.c
-281
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@@ -1,281 +0,0 @@
/*
* Common code between the AC-3 encoder and decoder
* Copyright (c) 2000 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Common code between the AC-3 encoder and decoder.
*/
#include "avcodec.h"
#include "ac3.h"
#include "get_bits.h"
#if CONFIG_HARDCODED_TABLES
/**
* Starting frequency coefficient bin for each critical band.
*/
static const uint8_t band_start_tab[51] = {
0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
10, 11, 12, 13, 14, 15, 16, 17, 18, 19,
20, 21, 22, 23, 24, 25, 26, 27, 28, 31,
34, 37, 40, 43, 46, 49, 55, 61, 67, 73,
79, 85, 97, 109, 121, 133, 157, 181, 205, 229, 253
};
/**
* Maps each frequency coefficient bin to the critical band that contains it.
*/
static const uint8_t bin_to_band_tab[253] = {
0,
1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12,
13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24,
25, 26, 27, 28, 28, 28, 29, 29, 29, 30, 30, 30,
31, 31, 31, 32, 32, 32, 33, 33, 33, 34, 34, 34,
35, 35, 35, 35, 35, 35, 36, 36, 36, 36, 36, 36,
37, 37, 37, 37, 37, 37, 38, 38, 38, 38, 38, 38,
39, 39, 39, 39, 39, 39, 40, 40, 40, 40, 40, 40,
41, 41, 41, 41, 41, 41, 41, 41, 41, 41, 41, 41,
42, 42, 42, 42, 42, 42, 42, 42, 42, 42, 42, 42,
43, 43, 43, 43, 43, 43, 43, 43, 43, 43, 43, 43,
44, 44, 44, 44, 44, 44, 44, 44, 44, 44, 44, 44,
45, 45, 45, 45, 45, 45, 45, 45, 45, 45, 45, 45,
45, 45, 45, 45, 45, 45, 45, 45, 45, 45, 45, 45,
46, 46, 46, 46, 46, 46, 46, 46, 46, 46, 46, 46,
46, 46, 46, 46, 46, 46, 46, 46, 46, 46, 46, 46,
47, 47, 47, 47, 47, 47, 47, 47, 47, 47, 47, 47,
47, 47, 47, 47, 47, 47, 47, 47, 47, 47, 47, 47,
48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48,
48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48,
49, 49, 49, 49, 49, 49, 49, 49, 49, 49, 49, 49,
49, 49, 49, 49, 49, 49, 49, 49, 49, 49, 49, 49
};
#else /* CONFIG_HARDCODED_TABLES */
static uint8_t band_start_tab[51];
static uint8_t bin_to_band_tab[253];
#endif
static inline int calc_lowcomp1(int a, int b0, int b1, int c)
{
if ((b0 + 256) == b1) {
a = c;
} else if (b0 > b1) {
a = FFMAX(a - 64, 0);
}
return a;
}
static inline int calc_lowcomp(int a, int b0, int b1, int bin)
{
if (bin < 7) {
return calc_lowcomp1(a, b0, b1, 384);
} else if (bin < 20) {
return calc_lowcomp1(a, b0, b1, 320);
} else {
return FFMAX(a - 128, 0);
}
}
void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
int16_t *band_psd)
{
int bin, band;
/* exponent mapping to PSD */
for (bin = start; bin < end; bin++) {
psd[bin]=(3072 - (exp[bin] << 7));
}
/* PSD integration */
bin = start;
band = bin_to_band_tab[start];
do {
int v = psd[bin++];
int band_end = FFMIN(band_start_tab[band+1], end);
for (; bin < band_end; bin++) {
int max = FFMAX(v, psd[bin]);
/* logadd */
int adr = FFMIN(max - ((v + psd[bin] + 1) >> 1), 255);
v = max + ff_ac3_log_add_tab[adr];
}
band_psd[band++] = v;
} while (end > band_start_tab[band]);
}
int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
int start, int end, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs, uint8_t *dba_offsets,
uint8_t *dba_lengths, uint8_t *dba_values,
int16_t *mask)
{
int16_t excite[50]; /* excitation */
int band;
int band_start, band_end, begin, end1;
int lowcomp, fastleak, slowleak;
/* excitation function */
band_start = bin_to_band_tab[start];
band_end = bin_to_band_tab[end-1] + 1;
if (band_start == 0) {
lowcomp = 0;
lowcomp = calc_lowcomp1(lowcomp, band_psd[0], band_psd[1], 384);
excite[0] = band_psd[0] - fast_gain - lowcomp;
lowcomp = calc_lowcomp1(lowcomp, band_psd[1], band_psd[2], 384);
excite[1] = band_psd[1] - fast_gain - lowcomp;
begin = 7;
for (band = 2; band < 7; band++) {
if (!(is_lfe && band == 6))
lowcomp = calc_lowcomp1(lowcomp, band_psd[band], band_psd[band+1], 384);
fastleak = band_psd[band] - fast_gain;
slowleak = band_psd[band] - s->slow_gain;
excite[band] = fastleak - lowcomp;
if (!(is_lfe && band == 6)) {
if (band_psd[band] <= band_psd[band+1]) {
begin = band + 1;
break;
}
}
}
end1 = FFMIN(band_end, 22);
for (band = begin; band < end1; band++) {
if (!(is_lfe && band == 6))
lowcomp = calc_lowcomp(lowcomp, band_psd[band], band_psd[band+1], band);
fastleak = FFMAX(fastleak - s->fast_decay, band_psd[band] - fast_gain);
slowleak = FFMAX(slowleak - s->slow_decay, band_psd[band] - s->slow_gain);
excite[band] = FFMAX(fastleak - lowcomp, slowleak);
}
begin = 22;
} else {
/* coupling channel */
begin = band_start;
fastleak = (s->cpl_fast_leak << 8) + 768;
slowleak = (s->cpl_slow_leak << 8) + 768;
}
for (band = begin; band < band_end; band++) {
fastleak = FFMAX(fastleak - s->fast_decay, band_psd[band] - fast_gain);
slowleak = FFMAX(slowleak - s->slow_decay, band_psd[band] - s->slow_gain);
excite[band] = FFMAX(fastleak, slowleak);
}
/* compute masking curve */
for (band = band_start; band < band_end; band++) {
int tmp = s->db_per_bit - band_psd[band];
if (tmp > 0) {
excite[band] += tmp >> 2;
}
mask[band] = FFMAX(ff_ac3_hearing_threshold_tab[band >> s->sr_shift][s->sr_code], excite[band]);
}
/* delta bit allocation */
if (dba_mode == DBA_REUSE || dba_mode == DBA_NEW) {
int i, seg, delta;
if (dba_nsegs >= 8)
return -1;
band = 0;
for (seg = 0; seg < dba_nsegs; seg++) {
band += dba_offsets[seg];
if (band >= 50 || dba_lengths[seg] > 50-band)
return -1;
if (dba_values[seg] >= 4) {
delta = (dba_values[seg] - 3) << 7;
} else {
delta = (dba_values[seg] - 4) << 7;
}
for (i = 0; i < dba_lengths[seg]; i++) {
mask[band++] += delta;
}
}
}
return 0;
}
void ff_ac3_bit_alloc_calc_bap(int16_t *mask, int16_t *psd, int start, int end,
int snr_offset, int floor,
const uint8_t *bap_tab, uint8_t *bap)
{
int bin, band;
/* special case, if snr offset is -960, set all bap's to zero */
if (snr_offset == -960) {
memset(bap, 0, 256);
return;
}
bin = start;
band = bin_to_band_tab[start];
do {
int m = (FFMAX(mask[band] - snr_offset - floor, 0) & 0x1FE0) + floor;
int band_end = FFMIN(band_start_tab[band+1], end);
for (; bin < band_end; bin++) {
int address = av_clip((psd[bin] - m) >> 5, 0, 63);
bap[bin] = bap_tab[address];
}
} while (end > band_start_tab[band++]);
}
/* AC-3 bit allocation. The algorithm is the one described in the AC-3
spec. */
void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
int8_t *exp, int start, int end,
int snr_offset, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs,
uint8_t *dba_offsets, uint8_t *dba_lengths,
uint8_t *dba_values)
{
int16_t psd[256]; /* scaled exponents */
int16_t band_psd[50]; /* interpolated exponents */
int16_t mask[50]; /* masking value */
ff_ac3_bit_alloc_calc_psd(exp, start, end, psd, band_psd);
ff_ac3_bit_alloc_calc_mask(s, band_psd, start, end, fast_gain, is_lfe,
dba_mode, dba_nsegs, dba_offsets, dba_lengths,
dba_values, mask);
ff_ac3_bit_alloc_calc_bap(mask, psd, start, end, snr_offset, s->floor,
ff_ac3_bap_tab, bap);
}
/**
* Initializes some tables.
* note: This function must remain thread safe because it is called by the
* AVParser init code.
*/
av_cold void ac3_common_init(void)
{
#if !CONFIG_HARDCODED_TABLES
/* compute bndtab and masktab from bandsz */
int bin = 0, band;
for (band = 0; band < 50; band++) {
int band_end = bin + ff_ac3_critical_band_size_tab[band];
band_start_tab[band] = bin;
while (bin < band_end)
bin_to_band_tab[bin++] = band;
}
band_start_tab[50] = bin;
#endif /* !CONFIG_HARDCODED_TABLES */
}
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3.h
-187
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@@ -1,187 +0,0 @@
/*
* Common code between the AC-3 encoder and decoder
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Common code between the AC-3 encoder and decoder.
*/
#ifndef AVCODEC_AC3_H
#define AVCODEC_AC3_H
#include "ac3tab.h"
#define AC3_MAX_CODED_FRAME_SIZE 3840 /* in bytes */
#define AC3_MAX_CHANNELS 6 /* including LFE channel */
#define NB_BLOCKS 6 /* number of PCM blocks inside an AC-3 frame */
#define AC3_FRAME_SIZE (NB_BLOCKS * 256)
/* exponent encoding strategy */
#define EXP_REUSE 0
#define EXP_NEW 1
#define EXP_D15 1
#define EXP_D25 2
#define EXP_D45 3
/** Delta bit allocation strategy */
typedef enum {
DBA_REUSE = 0,
DBA_NEW,
DBA_NONE,
DBA_RESERVED
} AC3DeltaStrategy;
/** Channel mode (audio coding mode) */
typedef enum {
AC3_CHMODE_DUALMONO = 0,
AC3_CHMODE_MONO,
AC3_CHMODE_STEREO,
AC3_CHMODE_3F,
AC3_CHMODE_2F1R,
AC3_CHMODE_3F1R,
AC3_CHMODE_2F2R,
AC3_CHMODE_3F2R
} AC3ChannelMode;
typedef struct AC3BitAllocParameters {
int sr_code;
int sr_shift;
int slow_gain, slow_decay, fast_decay, db_per_bit, floor;
int cpl_fast_leak, cpl_slow_leak;
} AC3BitAllocParameters;
/**
* @struct AC3HeaderInfo
* Coded AC-3 header values up to the lfeon element, plus derived values.
*/
typedef struct {
/** @defgroup coded Coded elements
* @{
*/
uint16_t sync_word;
uint16_t crc1;
uint8_t sr_code;
uint8_t bitstream_id;
uint8_t channel_mode;
uint8_t lfe_on;
uint8_t frame_type;
int substreamid; ///< substream identification
int center_mix_level; ///< Center mix level index
int surround_mix_level; ///< Surround mix level index
uint16_t channel_map;
int num_blocks; ///< number of audio blocks
/** @} */
/** @defgroup derived Derived values
* @{
*/
uint8_t sr_shift;
uint16_t sample_rate;
uint32_t bit_rate;
uint8_t channels;
uint16_t frame_size;
int64_t channel_layout;
/** @} */
} AC3HeaderInfo;
typedef enum {
EAC3_FRAME_TYPE_INDEPENDENT = 0,
EAC3_FRAME_TYPE_DEPENDENT,
EAC3_FRAME_TYPE_AC3_CONVERT,
EAC3_FRAME_TYPE_RESERVED
} EAC3FrameType;
void ac3_common_init(void);
/**
* Calculates the log power-spectral density of the input signal.
* This gives a rough estimate of signal power in the frequency domain by using
* the spectral envelope (exponents). The psd is also separately grouped
* into critical bands for use in the calculating the masking curve.
* 128 units in psd = -6 dB. The dbknee parameter in AC3BitAllocParameters
* determines the reference level.
*
* @param[in] exp frequency coefficient exponents
* @param[in] start starting bin location
* @param[in] end ending bin location
* @param[out] psd signal power for each frequency bin
* @param[out] band_psd signal power for each critical band
*/
void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
int16_t *band_psd);
/**
* Calculates the masking curve.
* First, the excitation is calculated using parameters in s and the signal
* power in each critical band. The excitation is compared with a predefined
* hearing threshold table to produce the masking curve. If delta bit
* allocation information is provided, it is used for adjusting the masking
* curve, usually to give a closer match to a better psychoacoustic model.
*
* @param[in] s adjustable bit allocation parameters
* @param[in] band_psd signal power for each critical band
* @param[in] start starting bin location
* @param[in] end ending bin location
* @param[in] fast_gain fast gain (estimated signal-to-mask ratio)
* @param[in] is_lfe whether or not the channel being processed is the LFE
* @param[in] dba_mode delta bit allocation mode (none, reuse, or new)
* @param[in] dba_nsegs number of delta segments
* @param[in] dba_offsets location offsets for each segment
* @param[in] dba_lengths length of each segment
* @param[in] dba_values delta bit allocation for each segment
* @param[out] mask calculated masking curve
* @return returns 0 for success, non-zero for error
*/
int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
int start, int end, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs, uint8_t *dba_offsets,
uint8_t *dba_lengths, uint8_t *dba_values,
int16_t *mask);
/**
* Calculates bit allocation pointers.
* The SNR is the difference between the masking curve and the signal. AC-3
* uses this value for each frequency bin to allocate bits. The snroffset
* parameter is a global adjustment to the SNR for all bins.
*
* @param[in] mask masking curve
* @param[in] psd signal power for each frequency bin
* @param[in] start starting bin location
* @param[in] end ending bin location
* @param[in] snr_offset SNR adjustment
* @param[in] floor noise floor
* @param[in] bap_tab look-up table for bit allocation pointers
* @param[out] bap bit allocation pointers
*/
void ff_ac3_bit_alloc_calc_bap(int16_t *mask, int16_t *psd, int start, int end,
int snr_offset, int floor,
const uint8_t *bap_tab, uint8_t *bap);
void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
int8_t *exp, int start, int end,
int snr_offset, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs,
uint8_t *dba_offsets, uint8_t *dba_lengths,
uint8_t *dba_values);
#endif /* AVCODEC_AC3_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3_parser.c
-207
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@@ -1,207 +0,0 @@
/*
* AC-3 parser
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "parser.h"
#include "ac3_parser.h"
#include "aac_ac3_parser.h"
#include "get_bits.h"
#define AC3_HEADER_SIZE 7
static const uint8_t eac3_blocks[4] = {
1, 2, 3, 6
};
int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
{
int frame_size_code;
memset(hdr, 0, sizeof(*hdr));
hdr->sync_word = get_bits(gbc, 16);
if(hdr->sync_word != 0x0B77)
return AAC_AC3_PARSE_ERROR_SYNC;
/* read ahead to bsid to distinguish between AC-3 and E-AC-3 */
hdr->bitstream_id = show_bits_long(gbc, 29) & 0x1F;
if(hdr->bitstream_id > 16)
return AAC_AC3_PARSE_ERROR_BSID;
hdr->num_blocks = 6;
/* set default mix levels */
hdr->center_mix_level = 1; // -4.5dB
hdr->surround_mix_level = 1; // -6.0dB
if(hdr->bitstream_id <= 10) {
/* Normal AC-3 */
hdr->crc1 = get_bits(gbc, 16);
hdr->sr_code = get_bits(gbc, 2);
if(hdr->sr_code == 3)
return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
frame_size_code = get_bits(gbc, 6);
if(frame_size_code > 37)
return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
skip_bits(gbc, 5); // skip bsid, already got it
skip_bits(gbc, 3); // skip bitstream mode
hdr->channel_mode = get_bits(gbc, 3);
if(hdr->channel_mode == AC3_CHMODE_STEREO) {
skip_bits(gbc, 2); // skip dsurmod
} else {
if((hdr->channel_mode & 1) && hdr->channel_mode != AC3_CHMODE_MONO)
hdr->center_mix_level = get_bits(gbc, 2);
if(hdr->channel_mode & 4)
hdr->surround_mix_level = get_bits(gbc, 2);
}
hdr->lfe_on = get_bits1(gbc);
hdr->sr_shift = FFMAX(hdr->bitstream_id, 8) - 8;
hdr->sample_rate = ff_ac3_sample_rate_tab[hdr->sr_code] >> hdr->sr_shift;
hdr->bit_rate = (ff_ac3_bitrate_tab[frame_size_code>>1] * 1000) >> hdr->sr_shift;
hdr->channels = ff_ac3_channels_tab[hdr->channel_mode] + hdr->lfe_on;
hdr->frame_size = ff_ac3_frame_size_tab[frame_size_code][hdr->sr_code] * 2;
hdr->frame_type = EAC3_FRAME_TYPE_AC3_CONVERT; //EAC3_FRAME_TYPE_INDEPENDENT;
hdr->substreamid = 0;
} else {
/* Enhanced AC-3 */
hdr->crc1 = 0;
hdr->frame_type = get_bits(gbc, 2);
if(hdr->frame_type == EAC3_FRAME_TYPE_RESERVED)
return AAC_AC3_PARSE_ERROR_FRAME_TYPE;
hdr->substreamid = get_bits(gbc, 3);
hdr->frame_size = (get_bits(gbc, 11) + 1) << 1;
if(hdr->frame_size < AC3_HEADER_SIZE)
return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
hdr->sr_code = get_bits(gbc, 2);
if (hdr->sr_code == 3) {
int sr_code2 = get_bits(gbc, 2);
if(sr_code2 == 3)
return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
hdr->sample_rate = ff_ac3_sample_rate_tab[sr_code2] / 2;
hdr->sr_shift = 1;
} else {
hdr->num_blocks = eac3_blocks[get_bits(gbc, 2)];
hdr->sample_rate = ff_ac3_sample_rate_tab[hdr->sr_code];
hdr->sr_shift = 0;
}
hdr->channel_mode = get_bits(gbc, 3);
hdr->lfe_on = get_bits1(gbc);
hdr->bit_rate = (uint32_t)(8.0 * hdr->frame_size * hdr->sample_rate /
(hdr->num_blocks * 256.0));
hdr->channels = ff_ac3_channels_tab[hdr->channel_mode] + hdr->lfe_on;
}
hdr->channel_layout = ff_ac3_channel_layout_tab[hdr->channel_mode];
if (hdr->lfe_on)
hdr->channel_layout |= CH_LOW_FREQUENCY;
return 0;
}
int ff_ac3_parse_header_full(GetBitContext *gbc, AC3HeaderInfo *hdr){
int ret, i;
ret = ff_ac3_parse_header(gbc, hdr);
if(!ret){
if(hdr->bitstream_id>10){
/* Enhanced AC-3 */
skip_bits(gbc, 5); // skip bitstream id
/* skip dialog normalization and compression gain */
for (i = 0; i < (hdr->channel_mode ? 1 : 2); i++) {
skip_bits(gbc, 5); // skip dialog normalization
if (get_bits1(gbc)) {
skip_bits(gbc, 8); //skip Compression gain word
}
}
/* dependent stream channel map */
if (hdr->frame_type == EAC3_FRAME_TYPE_DEPENDENT && get_bits1(gbc)) {
hdr->channel_map = get_bits(gbc, 16); //custom channel map
return 0;
}
}
//default channel map based on acmod and lfeon
hdr->channel_map = ff_eac3_default_chmap[hdr->channel_mode];
if(hdr->lfe_on)
hdr->channel_map |= AC3_CHMAP_LFE;
}
return ret;
}
static int ac3_sync(uint64_t state, AACAC3ParseContext *hdr_info,
int *need_next_header, int *new_frame_start)
{
int err;
union {
uint64_t u64;
uint8_t u8[8];
} tmp = { be2me_64(state) };
AC3HeaderInfo hdr;
GetBitContext gbc;
init_get_bits(&gbc, tmp.u8+8-AC3_HEADER_SIZE, 54);
err = ff_ac3_parse_header(&gbc, &hdr);
if(err < 0)
return 0;
hdr_info->sample_rate = hdr.sample_rate;
hdr_info->bit_rate = hdr.bit_rate;
hdr_info->channels = hdr.channels;
hdr_info->channel_layout = hdr.channel_layout;
hdr_info->samples = hdr.num_blocks * 256;
if(hdr.bitstream_id>10)
hdr_info->codec_id = CODEC_ID_EAC3;
else if (hdr_info->codec_id == CODEC_ID_NONE)
hdr_info->codec_id = CODEC_ID_AC3;
*need_next_header = (hdr.frame_type != EAC3_FRAME_TYPE_AC3_CONVERT);
*new_frame_start = (hdr.frame_type != EAC3_FRAME_TYPE_DEPENDENT);
return hdr.frame_size;
}
static av_cold int ac3_parse_init(AVCodecParserContext *s1)
{
AACAC3ParseContext *s = s1->priv_data;
s->header_size = AC3_HEADER_SIZE;
s->sync = ac3_sync;
return 0;
}
AVCodecParser ac3_parser = {
{ CODEC_ID_AC3, CODEC_ID_EAC3 },
sizeof(AACAC3ParseContext),
ac3_parse_init,
ff_aac_ac3_parse,
ff_parse_close,
};
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3_parser.h
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/*
* AC-3 parser prototypes
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AC3_PARSER_H
#define AVCODEC_AC3_PARSER_H
#include "ac3.h"
#include "get_bits.h"
/**
* Parses AC-3 frame header.
* Parses the header up to the lfeon element, which is the first 52 or 54 bits
* depending on the audio coding mode.
* @param gbc[in] BitContext containing the first 54 bits of the frame.
* @param hdr[out] Pointer to struct where header info is written.
* @return Returns 0 on success, -1 if there is a sync word mismatch,
* -2 if the bsid (version) element is invalid, -3 if the fscod (sample rate)
* element is invalid, or -4 if the frmsizecod (bit rate) element is invalid.
*/
int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr);
/**
* Parses AC-3 frame header and sets channel_map
* Parses the header up to the lfeon (channel_map in E-AC-3)
* element, which is the first 52, 54 or 104 bits depending
* on the audio coding mode.
* @param gbc[in] BitContext containing the first 54 bits of the frame.
* @param hdr[out] Pointer to struct where header info is written.
* @return value returned by ff_ac3_parse_header
*/
int ff_ac3_parse_header_full(GetBitContext *gbc, AC3HeaderInfo *hdr);
#endif /* AVCODEC_AC3_PARSER_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3dec.c
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src/add-ons/media/plugins/ffmpeg/libavcodec/ac3dec.h
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/*
* Common code between the AC-3 and E-AC-3 decoders
* Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Common code between the AC-3 and E-AC-3 decoders.
*
* Summary of MDCT Coefficient Grouping:
* The individual MDCT coefficient indices are often referred to in the
* (E-)AC-3 specification as frequency bins. These bins are grouped together
* into subbands of 12 coefficients each. The subbands are grouped together
* into bands as defined in the bitstream by the band structures, which
* determine the number of bands and the size of each band. The full spectrum
* of 256 frequency bins is divided into 1 DC bin + 21 subbands = 253 bins.
* This system of grouping coefficients is used for channel bandwidth, stereo
* rematrixing, channel coupling, enhanced coupling, and spectral extension.
*
* +-+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+-+
* |1| |12| | [12|12|12|12] | | | | | | | | | | | | |3|
* +-+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+--+-+
* ~~~ ~~~~ ~~~~~~~~~~~~~ ~~~
* | | | |
* | | | 3 unused frequency bins--+
* | | |
* | | +--1 band containing 4 subbands
* | |
* | +--1 subband of 12 frequency bins
* |
* +--DC frequency bin
*/
#ifndef AVCODEC_AC3DEC_H
#define AVCODEC_AC3DEC_H
#include "libavutil/lfg.h"
#include "ac3.h"
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
/* override ac3.h to include coupling channel */
#undef AC3_MAX_CHANNELS
#define AC3_MAX_CHANNELS 7
#define CPL_CH 0
#define AC3_OUTPUT_LFEON 8
#define AC3_MAX_COEFS 256
#define AC3_BLOCK_SIZE 256
#define MAX_BLOCKS 6
#define SPX_MAX_BANDS 17
typedef struct {
AVCodecContext *avctx; ///< parent context
GetBitContext gbc; ///< bitstream reader
uint8_t *input_buffer; ///< temp buffer to prevent overread
///@defgroup bsi bit stream information
///@{
int frame_type; ///< frame type (strmtyp)
int substreamid; ///< substream identification
int frame_size; ///< current frame size, in bytes
int bit_rate; ///< stream bit rate, in bits-per-second
int sample_rate; ///< sample frequency, in Hz
int num_blocks; ///< number of audio blocks
int channel_mode; ///< channel mode (acmod)
int channel_layout; ///< channel layout
int lfe_on; ///< lfe channel in use
int channel_map; ///< custom channel map
int center_mix_level; ///< Center mix level index
int surround_mix_level; ///< Surround mix level index
int eac3; ///< indicates if current frame is E-AC-3
///@}
///@defgroup audfrm frame syntax parameters
int snr_offset_strategy; ///< SNR offset strategy (snroffststr)
int block_switch_syntax; ///< block switch syntax enabled (blkswe)
int dither_flag_syntax; ///< dither flag syntax enabled (dithflage)
int bit_allocation_syntax; ///< bit allocation model syntax enabled (bamode)
int fast_gain_syntax; ///< fast gain codes enabled (frmfgaincode)
int dba_syntax; ///< delta bit allocation syntax enabled (dbaflde)
int skip_syntax; ///< skip field syntax enabled (skipflde)
///@}
///@defgroup cpl standard coupling
int cpl_in_use[MAX_BLOCKS]; ///< coupling in use (cplinu)
int cpl_strategy_exists[MAX_BLOCKS]; ///< coupling strategy exists (cplstre)
int channel_in_cpl[AC3_MAX_CHANNELS]; ///< channel in coupling (chincpl)
int phase_flags_in_use; ///< phase flags in use (phsflginu)
int phase_flags[18]; ///< phase flags (phsflg)
int num_cpl_bands; ///< number of coupling bands (ncplbnd)
uint8_t cpl_band_sizes[18]; ///< number of coeffs in each coupling band
int firstchincpl; ///< first channel in coupling
int first_cpl_coords[AC3_MAX_CHANNELS]; ///< first coupling coordinates states (firstcplcos)
int cpl_coords[AC3_MAX_CHANNELS][18]; ///< coupling coordinates (cplco)
///@}
///@defgroup spx spectral extension
///@{
int spx_in_use; ///< spectral extension in use (spxinu)
uint8_t channel_uses_spx[AC3_MAX_CHANNELS]; ///< channel uses spectral extension (chinspx)
int8_t spx_atten_code[AC3_MAX_CHANNELS]; ///< spx attenuation code (spxattencod)
int spx_src_start_freq; ///< spx start frequency bin
int spx_dst_end_freq; ///< spx end frequency bin
int spx_dst_start_freq; ///< spx starting frequency bin for copying (copystartmant)
///< the copy region ends at the start of the spx region.
int num_spx_bands; ///< number of spx bands (nspxbnds)
uint8_t spx_band_sizes[SPX_MAX_BANDS]; ///< number of bins in each spx band
uint8_t first_spx_coords[AC3_MAX_CHANNELS]; ///< first spx coordinates states (firstspxcos)
float spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor (nblendfact)
float spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
///@}
///@defgroup aht adaptive hybrid transform
int channel_uses_aht[AC3_MAX_CHANNELS]; ///< channel AHT in use (chahtinu)
int pre_mantissa[AC3_MAX_CHANNELS][AC3_MAX_COEFS][MAX_BLOCKS]; ///< pre-IDCT mantissas
///@}
///@defgroup channel channel
int fbw_channels; ///< number of full-bandwidth channels
int channels; ///< number of total channels
int lfe_ch; ///< index of LFE channel
float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
int downmixed; ///< indicates if coeffs are currently downmixed
int output_mode; ///< output channel configuration
int out_channels; ///< number of output channels
///@}
///@defgroup dynrng dynamic range
float dynamic_range[2]; ///< dynamic range
///@}
///@defgroup bandwidth bandwidth
int start_freq[AC3_MAX_CHANNELS]; ///< start frequency bin (strtmant)
int end_freq[AC3_MAX_CHANNELS]; ///< end frequency bin (endmant)
///@}
///@defgroup rematrixing rematrixing
int num_rematrixing_bands; ///< number of rematrixing bands (nrematbnd)
int rematrixing_flags[4]; ///< rematrixing flags (rematflg)
///@}
///@defgroup exponents exponents
int num_exp_groups[AC3_MAX_CHANNELS]; ///< Number of exponent groups (nexpgrp)
int8_t dexps[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< decoded exponents
int exp_strategy[MAX_BLOCKS][AC3_MAX_CHANNELS]; ///< exponent strategies (expstr)
///@}
///@defgroup bitalloc bit allocation
AC3BitAllocParameters bit_alloc_params; ///< bit allocation parameters
int first_cpl_leak; ///< first coupling leak state (firstcplleak)
int snr_offset[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets (snroffst)
int fast_gain[AC3_MAX_CHANNELS]; ///< fast gain values/SMR's (fgain)
uint8_t bap[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< bit allocation pointers
int16_t psd[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< scaled exponents
int16_t band_psd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents
int16_t mask[AC3_MAX_CHANNELS][50]; ///< masking curve values
int dba_mode[AC3_MAX_CHANNELS]; ///< delta bit allocation mode
int dba_nsegs[AC3_MAX_CHANNELS]; ///< number of delta segments
uint8_t dba_offsets[AC3_MAX_CHANNELS][8]; ///< delta segment offsets
uint8_t dba_lengths[AC3_MAX_CHANNELS][8]; ///< delta segment lengths
uint8_t dba_values[AC3_MAX_CHANNELS][8]; ///< delta values for each segment
///@}
///@defgroup dithering zero-mantissa dithering
int dither_flag[AC3_MAX_CHANNELS]; ///< dither flags (dithflg)
AVLFG dith_state; ///< for dither generation
///@}
///@defgroup imdct IMDCT
int block_switch[AC3_MAX_CHANNELS]; ///< block switch flags (blksw)
FFTContext imdct_512; ///< for 512 sample IMDCT
FFTContext imdct_256; ///< for 256 sample IMDCT
///@}
///@defgroup opt optimization
DSPContext dsp; ///< for optimization
float add_bias; ///< offset for float_to_int16 conversion
float mul_bias; ///< scaling for float_to_int16 conversion
///@}
///@defgroup arrays aligned arrays
DECLARE_ALIGNED(16, int, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///> fixed-point transform coefficients
DECLARE_ALIGNED(16, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform coefficients
DECLARE_ALIGNED(16, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< delay - added to the next block
DECLARE_ALIGNED(16, float, window)[AC3_BLOCK_SIZE]; ///< window coefficients
DECLARE_ALIGNED(16, float, tmp_output)[AC3_BLOCK_SIZE]; ///< temporary storage for output before windowing
DECLARE_ALIGNED(16, float, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< output after imdct transform and windowing
///@}
} AC3DecodeContext;
/**
* Parse the E-AC-3 frame header.
* This parses both the bit stream info and audio frame header.
*/
int ff_eac3_parse_header(AC3DecodeContext *s);
/**
* Decode mantissas in a single channel for the entire frame.
* This is used when AHT mode is enabled.
*/
void ff_eac3_decode_transform_coeffs_aht_ch(AC3DecodeContext *s, int ch);
void ff_ac3_downmix_c(float (*samples)[256], float (*matrix)[2],
int out_ch, int in_ch, int len);
/**
* Apply spectral extension to each channel by copying lower frequency
* coefficients to higher frequency bins and applying side information to
* approximate the original high frequency signal.
*/
void ff_eac3_apply_spectral_extension(AC3DecodeContext *s);
#endif /* AVCODEC_AC3DEC_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3dec_data.c
-72
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/*
* AC-3 and E-AC-3 decoder tables
* Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Tables taken directly from the AC-3 spec.
*/
#include "ac3dec_data.h"
#include "ac3.h"
/**
* Table used to ungroup 3 values stored in 5 bits.
* Used by bap=1 mantissas and GAQ.
* ff_ac3_ungroup_3_in_5_bits_tab[i] = { i/9, (i%9)/3, (i%9)%3 }
*/
const uint8_t ff_ac3_ungroup_3_in_5_bits_tab[32][3] = {
{ 0, 0, 0 }, { 0, 0, 1 }, { 0, 0, 2 }, { 0, 1, 0 },
{ 0, 1, 1 }, { 0, 1, 2 }, { 0, 2, 0 }, { 0, 2, 1 },
{ 0, 2, 2 }, { 1, 0, 0 }, { 1, 0, 1 }, { 1, 0, 2 },
{ 1, 1, 0 }, { 1, 1, 1 }, { 1, 1, 2 }, { 1, 2, 0 },
{ 1, 2, 1 }, { 1, 2, 2 }, { 2, 0, 0 }, { 2, 0, 1 },
{ 2, 0, 2 }, { 2, 1, 0 }, { 2, 1, 1 }, { 2, 1, 2 },
{ 2, 2, 0 }, { 2, 2, 1 }, { 2, 2, 2 }, { 3, 0, 0 },
{ 3, 0, 1 }, { 3, 0, 2 }, { 3, 1, 0 }, { 3, 1, 1 }
};
/**
* Table of bin locations for rematrixing bands
* reference: Section 7.5.2 Rematrixing : Frequency Band Definitions
*/
const uint8_t ff_ac3_rematrix_band_tab[5] = { 13, 25, 37, 61, 253 };
const uint8_t ff_eac3_hebap_tab[64] = {
0, 1, 2, 3, 4, 5, 6, 7, 8, 8,
8, 8, 9, 9, 9, 10, 10, 10, 10, 11,
11, 11, 11, 12, 12, 12, 12, 13, 13, 13,
13, 14, 14, 14, 14, 15, 15, 15, 15, 16,
16, 16, 16, 17, 17, 17, 17, 18, 18, 18,
18, 18, 18, 18, 18, 19, 19, 19, 19, 19,
19, 19, 19, 19,
};
/**
* Table E2.16 Default Coupling Banding Structure
*/
const uint8_t ff_eac3_default_cpl_band_struct[18] =
{ 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 1, 1, 0, 1, 1, 1, 1, 1 };
/**
* Table E2.15 Default Spectral Extension Banding Structure
*/
const uint8_t ff_eac3_default_spx_band_struct[17] =
{ 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1 };
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3dec_data.h
-34
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/*
* AC-3 and E-AC-3 decoder tables
* Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AC3DEC_DATA_H
#define AVCODEC_AC3DEC_DATA_H
#include <stdint.h>
extern const uint8_t ff_ac3_ungroup_3_in_5_bits_tab[32][3];
extern const uint8_t ff_ac3_rematrix_band_tab[5];
extern const uint8_t ff_eac3_hebap_tab[64];
extern const uint8_t ff_eac3_default_cpl_band_struct[18];
extern const uint8_t ff_eac3_default_spx_band_struct[17];
#endif /* AVCODEC_AC3DEC_DATA_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3enc.c
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src/add-ons/media/plugins/ffmpeg/libavcodec/ac3tab.c
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/*
* AC-3 tables
* copyright (c) 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* tables taken directly from the AC-3 spec.
*/
#include "avcodec.h"
#include "ac3tab.h"
/**
* Possible frame sizes.
* from ATSC A/52 Table 5.18 Frame Size Code Table.
*/
const uint16_t ff_ac3_frame_size_tab[38][3] = {
{ 64, 69, 96 },
{ 64, 70, 96 },
{ 80, 87, 120 },
{ 80, 88, 120 },
{ 96, 104, 144 },
{ 96, 105, 144 },
{ 112, 121, 168 },
{ 112, 122, 168 },
{ 128, 139, 192 },
{ 128, 140, 192 },
{ 160, 174, 240 },
{ 160, 175, 240 },
{ 192, 208, 288 },
{ 192, 209, 288 },
{ 224, 243, 336 },
{ 224, 244, 336 },
{ 256, 278, 384 },
{ 256, 279, 384 },
{ 320, 348, 480 },
{ 320, 349, 480 },
{ 384, 417, 576 },
{ 384, 418, 576 },
{ 448, 487, 672 },
{ 448, 488, 672 },
{ 512, 557, 768 },
{ 512, 558, 768 },
{ 640, 696, 960 },
{ 640, 697, 960 },
{ 768, 835, 1152 },
{ 768, 836, 1152 },
{ 896, 975, 1344 },
{ 896, 976, 1344 },
{ 1024, 1114, 1536 },
{ 1024, 1115, 1536 },
{ 1152, 1253, 1728 },
{ 1152, 1254, 1728 },
{ 1280, 1393, 1920 },
{ 1280, 1394, 1920 },
};
/**
* Maps audio coding mode (acmod) to number of full-bandwidth channels.
* from ATSC A/52 Table 5.8 Audio Coding Mode
*/
const uint8_t ff_ac3_channels_tab[8] = {
2, 1, 2, 3, 3, 4, 4, 5
};
/**
* Maps audio coding mode (acmod) to channel layout mask.
*/
const uint16_t ff_ac3_channel_layout_tab[8] = {
CH_LAYOUT_STEREO,
CH_LAYOUT_MONO,
CH_LAYOUT_STEREO,
CH_LAYOUT_SURROUND,
CH_LAYOUT_2_1,
CH_LAYOUT_4POINT0,
CH_LAYOUT_2_2,
CH_LAYOUT_5POINT0
};
#define COMMON_CHANNEL_MAP \
{ { 0, 1, }, { 0, 1, 2, } },\
{ { 0, }, { 0, 1, } },\
{ { 0, 1, }, { 0, 1, 2, } },\
{ { 0, 2, 1, }, { 0, 2, 1, 3, } },\
{ { 0, 1, 2, }, { 0, 1, 3, 2, } },\
{ { 0, 2, 1, 3, }, { 0, 2, 1, 4, 3, } },
/**
* Table to remap channels from SMPTE order to AC-3 order.
* [channel_mode][lfe][ch]
*/
const uint8_t ff_ac3_enc_channel_map[8][2][6] = {
COMMON_CHANNEL_MAP
{ { 0, 1, 2, 3, }, { 0, 1, 3, 4, 2, } },
{ { 0, 2, 1, 3, 4, }, { 0, 2, 1, 4, 5, 3 } },
};
/**
* Table to remap channels from from AC-3 order to SMPTE order.
* [channel_mode][lfe][ch]
*/
const uint8_t ff_ac3_dec_channel_map[8][2][6] = {
COMMON_CHANNEL_MAP
{ { 0, 1, 2, 3, }, { 0, 1, 4, 2, 3, } },
{ { 0, 2, 1, 3, 4, }, { 0, 2, 1, 5, 3, 4 } },
};
/* possible frequencies */
const uint16_t ff_ac3_sample_rate_tab[3] = { 48000, 44100, 32000 };
/* possible bitrates */
const uint16_t ff_ac3_bitrate_tab[19] = {
32, 40, 48, 56, 64, 80, 96, 112, 128,
160, 192, 224, 256, 320, 384, 448, 512, 576, 640
};
/* AC-3 MDCT window */
/* MDCT window */
const int16_t ff_ac3_window[256] = {
4, 7, 12, 16, 21, 28, 34, 42,
51, 61, 72, 84, 97, 111, 127, 145,
164, 184, 207, 231, 257, 285, 315, 347,
382, 419, 458, 500, 544, 591, 641, 694,
750, 810, 872, 937, 1007, 1079, 1155, 1235,
1318, 1406, 1497, 1593, 1692, 1796, 1903, 2016,
2132, 2253, 2379, 2509, 2644, 2783, 2927, 3076,
3230, 3389, 3552, 3721, 3894, 4072, 4255, 4444,
4637, 4835, 5038, 5246, 5459, 5677, 5899, 6127,
6359, 6596, 6837, 7083, 7334, 7589, 7848, 8112,
8380, 8652, 8927, 9207, 9491, 9778,10069,10363,
10660,10960,11264,11570,11879,12190,12504,12820,
13138,13458,13780,14103,14427,14753,15079,15407,
15735,16063,16392,16720,17049,17377,17705,18032,
18358,18683,19007,19330,19651,19970,20287,20602,
20914,21225,21532,21837,22139,22438,22733,23025,
23314,23599,23880,24157,24430,24699,24964,25225,
25481,25732,25979,26221,26459,26691,26919,27142,
27359,27572,27780,27983,28180,28373,28560,28742,
28919,29091,29258,29420,29577,29729,29876,30018,
30155,30288,30415,30538,30657,30771,30880,30985,
31086,31182,31274,31363,31447,31528,31605,31678,
31747,31814,31877,31936,31993,32046,32097,32145,
32190,32232,32272,32310,32345,32378,32409,32438,
32465,32490,32513,32535,32556,32574,32592,32608,
32623,32636,32649,32661,32671,32681,32690,32698,
32705,32712,32718,32724,32729,32733,32737,32741,
32744,32747,32750,32752,32754,32756,32757,32759,
32760,32761,32762,32763,32764,32764,32765,32765,
32766,32766,32766,32766,32767,32767,32767,32767,
32767,32767,32767,32767,32767,32767,32767,32767,
32767,32767,32767,32767,32767,32767,32767,32767,
};
const uint8_t ff_ac3_log_add_tab[260]= {
0x40,0x3f,0x3e,0x3d,0x3c,0x3b,0x3a,0x39,0x38,0x37,
0x36,0x35,0x34,0x34,0x33,0x32,0x31,0x30,0x2f,0x2f,
0x2e,0x2d,0x2c,0x2c,0x2b,0x2a,0x29,0x29,0x28,0x27,
0x26,0x26,0x25,0x24,0x24,0x23,0x23,0x22,0x21,0x21,
0x20,0x20,0x1f,0x1e,0x1e,0x1d,0x1d,0x1c,0x1c,0x1b,
0x1b,0x1a,0x1a,0x19,0x19,0x18,0x18,0x17,0x17,0x16,
0x16,0x15,0x15,0x15,0x14,0x14,0x13,0x13,0x13,0x12,
0x12,0x12,0x11,0x11,0x11,0x10,0x10,0x10,0x0f,0x0f,
0x0f,0x0e,0x0e,0x0e,0x0d,0x0d,0x0d,0x0d,0x0c,0x0c,
0x0c,0x0c,0x0b,0x0b,0x0b,0x0b,0x0a,0x0a,0x0a,0x0a,
0x0a,0x09,0x09,0x09,0x09,0x09,0x08,0x08,0x08,0x08,
0x08,0x08,0x07,0x07,0x07,0x07,0x07,0x07,0x06,0x06,
0x06,0x06,0x06,0x06,0x06,0x06,0x05,0x05,0x05,0x05,
0x05,0x05,0x05,0x05,0x04,0x04,0x04,0x04,0x04,0x04,
0x04,0x04,0x04,0x04,0x04,0x03,0x03,0x03,0x03,0x03,
0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x02,
0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,
0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x01,0x01,
0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
};
const uint16_t ff_ac3_hearing_threshold_tab[50][3]= {
{ 0x04d0,0x04f0,0x0580 },
{ 0x04d0,0x04f0,0x0580 },
{ 0x0440,0x0460,0x04b0 },
{ 0x0400,0x0410,0x0450 },
{ 0x03e0,0x03e0,0x0420 },
{ 0x03c0,0x03d0,0x03f0 },
{ 0x03b0,0x03c0,0x03e0 },
{ 0x03b0,0x03b0,0x03d0 },
{ 0x03a0,0x03b0,0x03c0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03a0 },
{ 0x0390,0x03a0,0x03a0 },
{ 0x0390,0x0390,0x03a0 },
{ 0x0390,0x0390,0x03a0 },
{ 0x0380,0x0390,0x03a0 },
{ 0x0380,0x0380,0x03a0 },
{ 0x0370,0x0380,0x03a0 },
{ 0x0370,0x0380,0x03a0 },
{ 0x0360,0x0370,0x0390 },
{ 0x0360,0x0370,0x0390 },
{ 0x0350,0x0360,0x0390 },
{ 0x0350,0x0360,0x0390 },
{ 0x0340,0x0350,0x0380 },
{ 0x0340,0x0350,0x0380 },
{ 0x0330,0x0340,0x0380 },
{ 0x0320,0x0340,0x0370 },
{ 0x0310,0x0320,0x0360 },
{ 0x0300,0x0310,0x0350 },
{ 0x02f0,0x0300,0x0340 },
{ 0x02f0,0x02f0,0x0330 },
{ 0x02f0,0x02f0,0x0320 },
{ 0x02f0,0x02f0,0x0310 },
{ 0x0300,0x02f0,0x0300 },
{ 0x0310,0x0300,0x02f0 },
{ 0x0340,0x0320,0x02f0 },
{ 0x0390,0x0350,0x02f0 },
{ 0x03e0,0x0390,0x0300 },
{ 0x0420,0x03e0,0x0310 },
{ 0x0460,0x0420,0x0330 },
{ 0x0490,0x0450,0x0350 },
{ 0x04a0,0x04a0,0x03c0 },
{ 0x0460,0x0490,0x0410 },
{ 0x0440,0x0460,0x0470 },
{ 0x0440,0x0440,0x04a0 },
{ 0x0520,0x0480,0x0460 },
{ 0x0800,0x0630,0x0440 },
{ 0x0840,0x0840,0x0450 },
{ 0x0840,0x0840,0x04e0 },
};
const uint8_t ff_ac3_bap_tab[64]= {
0, 1, 1, 1, 1, 1, 2, 2, 3, 3,
3, 4, 4, 5, 5, 6, 6, 6, 6, 7,
7, 7, 7, 8, 8, 8, 8, 9, 9, 9,
9, 10, 10, 10, 10, 11, 11, 11, 11, 12,
12, 12, 12, 13, 13, 13, 13, 14, 14, 14,
14, 14, 14, 14, 14, 15, 15, 15, 15, 15,
15, 15, 15, 15,
};
const uint8_t ff_ac3_slow_decay_tab[4]={
0x0f, 0x11, 0x13, 0x15,
};
const uint8_t ff_ac3_fast_decay_tab[4]={
0x3f, 0x53, 0x67, 0x7b,
};
const uint16_t ff_ac3_slow_gain_tab[4]= {
0x540, 0x4d8, 0x478, 0x410,
};
const uint16_t ff_ac3_db_per_bit_tab[4]= {
0x000, 0x700, 0x900, 0xb00,
};
const int16_t ff_ac3_floor_tab[8]= {
0x2f0, 0x2b0, 0x270, 0x230, 0x1f0, 0x170, 0x0f0, 0xf800,
};
const uint16_t ff_ac3_fast_gain_tab[8]= {
0x080, 0x100, 0x180, 0x200, 0x280, 0x300, 0x380, 0x400,
};
const uint8_t ff_ac3_critical_band_size_tab[50]={
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 3, 3, 3, 3, 3, 3,
3, 6, 6, 6, 6, 6, 6, 12, 12, 12, 12, 24, 24, 24, 24, 24
};
/**
* Default channel map for a dependent substream defined by acmod
*/
const uint16_t ff_eac3_default_chmap[8] = {
AC3_CHMAP_L | AC3_CHMAP_R, // FIXME Ch1+Ch2
AC3_CHMAP_C,
AC3_CHMAP_L | AC3_CHMAP_R,
AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R,
AC3_CHMAP_L | AC3_CHMAP_R | AC3_CHMAP_C_SUR,
AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R | AC3_CHMAP_C_SUR,
AC3_CHMAP_L | AC3_CHMAP_R | AC3_CHMAP_L_SUR | AC3_CHMAP_R_SUR,
AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R | AC3_CHMAP_L_SUR | AC3_CHMAP_R_SUR
};
src/add-ons/media/plugins/ffmpeg/libavcodec/ac3tab.h
-62
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@@ -1,62 +0,0 @@
/*
* AC-3 tables
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AC3TAB_H
#define AVCODEC_AC3TAB_H
#include "libavutil/common.h"
extern const uint16_t ff_ac3_frame_size_tab[38][3];
extern const uint8_t ff_ac3_channels_tab[8];
extern const uint16_t ff_ac3_channel_layout_tab[8];
extern const uint8_t ff_ac3_enc_channel_map[8][2][6];
extern const uint8_t ff_ac3_dec_channel_map[8][2][6];
extern const uint16_t ff_ac3_sample_rate_tab[3];
extern const uint16_t ff_ac3_bitrate_tab[19];
extern const int16_t ff_ac3_window[256];
extern const uint8_t ff_ac3_log_add_tab[260];
extern const uint16_t ff_ac3_hearing_threshold_tab[50][3];
extern const uint8_t ff_ac3_bap_tab[64];
extern const uint8_t ff_ac3_slow_decay_tab[4];
extern const uint8_t ff_ac3_fast_decay_tab[4];
extern const uint16_t ff_ac3_slow_gain_tab[4];
extern const uint16_t ff_ac3_db_per_bit_tab[4];
extern const int16_t ff_ac3_floor_tab[8];
extern const uint16_t ff_ac3_fast_gain_tab[8];
extern const uint8_t ff_ac3_critical_band_size_tab[50];
extern const uint16_t ff_eac3_default_chmap[8];
/** Custom channel map locations bitmask
* Other channels described in documentation:
* Lc/Rc pair, Lrs/Rrs pair, Ts, Lsd/Rsd pair,
* Lw/Rw pair, Lvh/Rvh pair, Cvh, Reserved, LFE2
*/
enum CustomChannelMapLocation{
AC3_CHMAP_L= 1<<(15-0),
AC3_CHMAP_C= 1<<(15-1),
AC3_CHMAP_R= 1<<(15-2),
AC3_CHMAP_L_SUR= 1<<(15-3),
AC3_CHMAP_R_SUR = 1<<(15-4),
AC3_CHMAP_C_SUR= 1<<(15-7),
AC3_CHMAP_LFE = 1<<(15-15)
};
#endif /* AVCODEC_AC3TAB_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/acelp_filters.c
-145
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@@ -1,145 +0,0 @@
/*
* various filters for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <inttypes.h>
#include "avcodec.h"
#include "acelp_filters.h"
const int16_t ff_acelp_interp_filter[61] = { /* (0.15) */
29443, 28346, 25207, 20449, 14701, 8693,
3143, -1352, -4402, -5865, -5850, -4673,
-2783, -672, 1211, 2536, 3130, 2991,
2259, 1170, 0, -1001, -1652, -1868,
-1666, -1147, -464, 218, 756, 1060,
1099, 904, 550, 135, -245, -514,
-634, -602, -451, -231, 0, 191,
308, 340, 296, 198, 78, -36,
-120, -163, -165, -132, -79, -19,
34, 73, 91, 89, 70, 38,
0,
};
void ff_acelp_interpolate(int16_t* out, const int16_t* in,
const int16_t* filter_coeffs, int precision,
int frac_pos, int filter_length, int length)
{
int n, i;
assert(frac_pos >= 0 && frac_pos < precision);
for (n = 0; n < length; n++) {
int idx = 0;
int v = 0x4000;
for (i = 0; i < filter_length;) {
/* The reference G.729 and AMR fixed point code performs clipping after
each of the two following accumulations.
Since clipping affects only the synthetic OVERFLOW test without
causing an int type overflow, it was moved outside the loop. */
/* R(x):=ac_v[-k+x]
v += R(n-i)*ff_acelp_interp_filter(t+6i)
v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
v += in[n + i] * filter_coeffs[idx + frac_pos];
idx += precision;
i++;
v += in[n - i] * filter_coeffs[idx - frac_pos];
}
if (av_clip_int16(v >> 15) != (v >> 15))
av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n");
out[n] = v >> 15;
}
}
void ff_acelp_interpolatef(float *out, const float *in,
const float *filter_coeffs, int precision,
int frac_pos, int filter_length, int length)
{
int n, i;
for (n = 0; n < length; n++) {
int idx = 0;
float v = 0;
for (i = 0; i < filter_length;) {
v += in[n + i] * filter_coeffs[idx + frac_pos];
idx += precision;
i++;
v += in[n - i] * filter_coeffs[idx - frac_pos];
}
out[n] = v;
}
}
void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
const int16_t* in, int length)
{
int i;
int tmp;
for (i = 0; i < length; i++) {
tmp = (hpf_f[0]* 15836LL) >> 13;
tmp += (hpf_f[1]* -7667LL) >> 13;
tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]);
/* With "+0x800" rounding, clipping is needed
for ALGTHM and SPEECH tests. */
out[i] = av_clip_int16((tmp + 0x800) >> 12);
hpf_f[1] = hpf_f[0];
hpf_f[0] = tmp;
}
}
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
const float zero_coeffs[2],
const float pole_coeffs[2],
float gain, float mem[2], int n)
{
int i;
float tmp;
for (i = 0; i < n; i++) {
tmp = gain * in[i] - pole_coeffs[0] * mem[0] - pole_coeffs[1] * mem[1];
out[i] = tmp + zero_coeffs[0] * mem[0] + zero_coeffs[1] * mem[1];
mem[1] = mem[0];
mem[0] = tmp;
}
}
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
{
float new_tilt_mem = samples[size - 1];
int i;
for (i = size - 1; i > 0; i--)
samples[i] -= tilt * samples[i - 1];
samples[0] -= tilt * *mem;
*mem = new_tilt_mem;
}
src/add-ons/media/plugins/ffmpeg/libavcodec/acelp_filters.h
-120
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@@ -1,120 +0,0 @@
/*
* various filters for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_ACELP_FILTERS_H
#define AVCODEC_ACELP_FILTERS_H
#include <stdint.h>
/**
* low-pass Finite Impulse Response filter coefficients.
*
* Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
* the coefficients are scaled by 2^15.
* This array only contains the right half of the filter.
* This filter is likely identical to the one used in G.729, though this
* could not be determined from the original comments with certainity.
*/
extern const int16_t ff_acelp_interp_filter[61];
/**
* Generic FIR interpolation routine.
* @param out [out] buffer for interpolated data
* @param in input data
* @param filter_coeffs interpolation filter coefficients (0.15)
* @param precision sub sample factor, that is the precision of the position
* @param frac_pos fractional part of position [0..precision-1]
* @param filter_length filter length
* @param length length of output
*
* filter_coeffs contains coefficients of the right half of the symmetric
* interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
* See ff_acelp_interp_filter for an example.
*
*/
void ff_acelp_interpolate(int16_t* out, const int16_t* in,
const int16_t* filter_coeffs, int precision,
int frac_pos, int filter_length, int length);
/**
* Floating point version of ff_acelp_interpolate()
*/
void ff_acelp_interpolatef(float *out, const float *in,
const float *filter_coeffs, int precision,
int frac_pos, int filter_length, int length);
/**
* high-pass filtering and upscaling (4.2.5 of G.729).
* @param out [out] output buffer for filtered speech data
* @param hpf_f [in/out] past filtered data from previous (2 items long)
* frames (-0x20000000 <= (14.13) < 0x20000000)
* @param in speech data to process
* @param length input data size
*
* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
*
* The filter has a cut-off frequency of 1/80 of the sampling freq
*
* @note Two items before the top of the out buffer must contain two items from the
* tail of the previous subframe.
*
* @remark It is safe to pass the same array in in and out parameters.
*
* @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
* but constants differs in 5th sign after comma). Fortunately in
* fixed-point all coefficients are the same as in G.729. Thus this
* routine can be used for the fixed-point AMR decoder, too.
*/
void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
const int16_t* in, int length);
/**
* Apply an order 2 rational transfer function in-place.
*
* @param out output buffer for filtered speech samples
* @param in input buffer containing speech data (may be the same as out)
* @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
* @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
* @param gain scale factor for final output
* @param mem intermediate values used by filter (should be 0 initially)
* @param n number of samples
*/
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
const float zero_coeffs[2],
const float pole_coeffs[2],
float gain,
float mem[2], int n);
/**
* Apply tilt compensation filter, 1 - tilt * z-1.
*
* @param mem pointer to the filter's state (one single float)
* @param tilt tilt factor
* @param samples array where the filter is applied
* @param size the size of the samples array
*/
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
#endif /* AVCODEC_ACELP_FILTERS_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/acelp_pitch_delay.c
-186
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@@ -1,186 +0,0 @@
/*
* gain code, gain pitch and pitch delay decoding
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "dsputil.h"
#include "acelp_pitch_delay.h"
#include "celp_math.h"
int ff_acelp_decode_8bit_to_1st_delay3(int ac_index)
{
ac_index += 58;
if(ac_index > 254)
ac_index = 3 * ac_index - 510;
return ac_index;
}
int ff_acelp_decode_4bit_to_2nd_delay3(
int ac_index,
int pitch_delay_min)
{
if(ac_index < 4)
return 3 * (ac_index + pitch_delay_min);
else if(ac_index < 12)
return 3 * pitch_delay_min + ac_index + 6;
else
return 3 * (ac_index + pitch_delay_min) - 18;
}
int ff_acelp_decode_5_6_bit_to_2nd_delay3(
int ac_index,
int pitch_delay_min)
{
return 3 * pitch_delay_min + ac_index - 2;
}
int ff_acelp_decode_9bit_to_1st_delay6(int ac_index)
{
if(ac_index < 463)
return ac_index + 105;
else
return 6 * (ac_index - 368);
}
int ff_acelp_decode_6bit_to_2nd_delay6(
int ac_index,
int pitch_delay_min)
{
return 6 * pitch_delay_min + ac_index - 3;
}
void ff_acelp_update_past_gain(
int16_t* quant_energy,
int gain_corr_factor,
int log2_ma_pred_order,
int erasure)
{
int i;
int avg_gain=quant_energy[(1 << log2_ma_pred_order) - 1]; // (5.10)
for(i=(1 << log2_ma_pred_order) - 1; i>0; i--)
{
avg_gain += quant_energy[i-1];
quant_energy[i] = quant_energy[i-1];
}
if(erasure)
quant_energy[0] = FFMAX(avg_gain >> log2_ma_pred_order, -10240) - 4096; // -10 and -4 in (5.10)
else
quant_energy[0] = (6165 * ((ff_log2(gain_corr_factor) >> 2) - (13 << 13))) >> 13;
}
int16_t ff_acelp_decode_gain_code(
DSPContext *dsp,
int gain_corr_factor,
const int16_t* fc_v,
int mr_energy,
const int16_t* quant_energy,
const int16_t* ma_prediction_coeff,
int subframe_size,
int ma_pred_order)
{
int i;
mr_energy <<= 10;
for(i=0; i<ma_pred_order; i++)
mr_energy += quant_energy[i] * ma_prediction_coeff[i];
#ifdef G729_BITEXACT
mr_energy += (((-6165LL * ff_log2(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size, 0))) >> 3) & ~0x3ff);
mr_energy = (5439 * (mr_energy >> 15)) >> 8; // (0.15) = (0.15) * (7.23)
return bidir_sal(
((ff_exp2(mr_energy & 0x7fff) + 16) >> 5) * (gain_corr_factor >> 1),
(mr_energy >> 15) - 25
);
#else
mr_energy = gain_corr_factor * exp(M_LN10 / (20 << 23) * mr_energy) /
sqrt(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size, 0));
return mr_energy >> 12;
#endif
}
float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
float *prediction_error, float energy_mean,
const float *pred_table)
{
// Equations 66-69:
// ^g_c = ^gamma_gc * 100.05 (predicted dB + mean dB - dB of fixed vector)
// Note 10^(0.05 * -10log(average x2)) = 1/sqrt((average x2)).
float val = fixed_gain_factor *
exp2f(M_LOG2_10 * 0.05 *
(ff_dot_productf(pred_table, prediction_error, 4) +
energy_mean)) /
sqrtf(fixed_mean_energy);
// update quantified prediction error energy history
memmove(&prediction_error[0], &prediction_error[1],
3 * sizeof(prediction_error[0]));
prediction_error[3] = 20.0 * log10f(fixed_gain_factor);
return val;
}
void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index,
const int prev_lag_int, const int subframe,
int third_as_first, int resolution)
{
/* Note n * 10923 >> 15 is floor(x/3) for 0 <= n <= 32767 */
if (subframe == 0 || (subframe == 2 && third_as_first)) {
if (pitch_index < 197)
pitch_index += 59;
else
pitch_index = 3 * pitch_index - 335;
} else {
if (resolution == 4) {
int search_range_min = av_clip(prev_lag_int - 5, PITCH_DELAY_MIN,
PITCH_DELAY_MAX - 9);
// decoding with 4-bit resolution
if (pitch_index < 4) {
// integer only precision for [search_range_min, search_range_min+3]
pitch_index = 3 * (pitch_index + search_range_min) + 1;
} else if (pitch_index < 12) {
// 1/3 fractional precision for [search_range_min+3 1/3, search_range_min+5 2/3]
pitch_index += 3 * search_range_min + 7;
} else {
// integer only precision for [search_range_min+6, search_range_min+9]
pitch_index = 3 * (pitch_index + search_range_min - 6) + 1;
}
} else {
// decoding with 5 or 6 bit resolution, 1/3 fractional precision
pitch_index--;
if (resolution == 5) {
pitch_index += 3 * av_clip(prev_lag_int - 10, PITCH_DELAY_MIN,
PITCH_DELAY_MAX - 19);
} else
pitch_index += 3 * av_clip(prev_lag_int - 5, PITCH_DELAY_MIN,
PITCH_DELAY_MAX - 9);
}
}
*lag_int = pitch_index * 10923 >> 15;
*lag_frac = pitch_index - 3 * *lag_int - 1;
}
src/add-ons/media/plugins/ffmpeg/libavcodec/acelp_pitch_delay.h
-255
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@@ -1,255 +0,0 @@
/*
* gain code, gain pitch and pitch delay decoding
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_ACELP_PITCH_DELAY_H
#define AVCODEC_ACELP_PITCH_DELAY_H
#include <stdint.h>
#include "dsputil.h"
#define PITCH_DELAY_MIN 20
#define PITCH_DELAY_MAX 143
/**
* \brief Decode pitch delay of the first subframe encoded by 8 bits with 1/3
* resolution.
* \param ac_index adaptive codebook index (8 bits)
*
* \return pitch delay in 1/3 units
*
* Pitch delay is coded:
* with 1/3 resolution, 19 < pitch_delay < 85
* integers only, 85 <= pitch_delay <= 143
*/
int ff_acelp_decode_8bit_to_1st_delay3(int ac_index);
/**
* \brief Decode pitch delay of the second subframe encoded by 5 or 6 bits
* with 1/3 precision.
* \param ac_index adaptive codebook index (5 or 6 bits)
* \param pitch_delay_min lower bound (integer) of pitch delay interval
* for second subframe
*
* \return pitch delay in 1/3 units
*
* Pitch delay is coded:
* with 1/3 resolution, -6 < pitch_delay - int(prev_pitch_delay) < 5
*
* \remark The routine is used in G.729 @8k, AMR @10.2k, AMR @7.95k,
* AMR @7.4k for the second subframe.
*/
int ff_acelp_decode_5_6_bit_to_2nd_delay3(
int ac_index,
int pitch_delay_min);
/**
* \brief Decode pitch delay with 1/3 precision.
* \param ac_index adaptive codebook index (4 bits)
* \param pitch_delay_min lower bound (integer) of pitch delay interval for
* second subframe
*
* \return pitch delay in 1/3 units
*
* Pitch delay is coded:
* integers only, -6 < pitch_delay - int(prev_pitch_delay) <= -2
* with 1/3 resolution, -2 < pitch_delay - int(prev_pitch_delay) < 1
* integers only, 1 <= pitch_delay - int(prev_pitch_delay) < 5
*
* \remark The routine is used in G.729 @6.4k, AMR @6.7k, AMR @5.9k,
* AMR @5.15k, AMR @4.75k for the second subframe.
*/
int ff_acelp_decode_4bit_to_2nd_delay3(
int ac_index,
int pitch_delay_min);
/**
* \brief Decode pitch delay of the first subframe encoded by 9 bits
* with 1/6 precision.
* \param ac_index adaptive codebook index (9 bits)
* \param pitch_delay_min lower bound (integer) of pitch delay interval for
* second subframe
*
* \return pitch delay in 1/6 units
*
* Pitch delay is coded:
* with 1/6 resolution, 17 < pitch_delay < 95
* integers only, 95 <= pitch_delay <= 143
*
* \remark The routine is used in AMR @12.2k for the first and third subframes.
*/
int ff_acelp_decode_9bit_to_1st_delay6(int ac_index);
/**
* \brief Decode pitch delay of the second subframe encoded by 6 bits
* with 1/6 precision.
* \param ac_index adaptive codebook index (6 bits)
* \param pitch_delay_min lower bound (integer) of pitch delay interval for
* second subframe
*
* \return pitch delay in 1/6 units
*
* Pitch delay is coded:
* with 1/6 resolution, -6 < pitch_delay - int(prev_pitch_delay) < 5
*
* \remark The routine is used in AMR @12.2k for the second and fourth subframes.
*/
int ff_acelp_decode_6bit_to_2nd_delay6(
int ac_index,
int pitch_delay_min);
/**
* \brief Update past quantized energies
* \param quant_energy [in/out] past quantized energies (5.10)
* \param gain_corr_factor gain correction factor
* \param log2_ma_pred_order log2() of MA prediction order
* \param erasure frame erasure flag
*
* If frame erasure flag is not equal to zero, memory is updated with
* averaged energy, attenuated by 4dB:
* max(avg(quant_energy[i])-4, -14), i=0,ma_pred_order
*
* In normal mode memory is updated with
* Er - Ep = 20 * log10(gain_corr_factor)
*
* \remark The routine is used in G.729 and AMR (all modes).
*/
void ff_acelp_update_past_gain(
int16_t* quant_energy,
int gain_corr_factor,
int log2_ma_pred_order,
int erasure);
/**
* \brief Decode the adaptive codebook gain and add
* correction (4.1.5 and 3.9.1 of G.729).
* \param dsp initialized dsputil context
* \param gain_corr_factor gain correction factor (2.13)
* \param fc_v fixed-codebook vector (2.13)
* \param mr_energy mean innovation energy and fixed-point correction (7.13)
* \param quant_energy [in/out] past quantized energies (5.10)
* \param subframe_size length of subframe
* \param ma_pred_order MA prediction order
*
* \return quantized fixed-codebook gain (14.1)
*
* The routine implements equations 69, 66 and 71 of the G.729 specification (3.9.1)
*
* Em - mean innovation energy (dB, constant, depends on decoding algorithm)
* Ep - mean-removed predicted energy (dB)
* Er - mean-removed innovation energy (dB)
* Ei - mean energy of the fixed-codebook contribution (dB)
* N - subframe_size
* M - MA (Moving Average) prediction order
* gc - fixed-codebook gain
* gc_p - predicted fixed-codebook gain
*
* Fixed codebook gain is computed using predicted gain gc_p and
* correction factor gain_corr_factor as shown below:
*
* gc = gc_p * gain_corr_factor
*
* The predicted fixed codebook gain gc_p is found by predicting
* the energy of the fixed-codebook contribution from the energy
* of previous fixed-codebook contributions.
*
* mean = 1/N * sum(i,0,N){ fc_v[i] * fc_v[i] }
*
* Ei = 10log(mean)
*
* Er = 10log(1/N * gc^2 * mean) - Em = 20log(gc) + Ei - Em
*
* Replacing Er with Ep and gc with gc_p we will receive:
*
* Ep = 10log(1/N * gc_p^2 * mean) - Em = 20log(gc_p) + Ei - Em
*
* and from above:
*
* gc_p = 10^((Ep - Ei + Em) / 20)
*
* Ep is predicted using past energies and prediction coefficients:
*
* Ep = sum(i,0,M){ ma_prediction_coeff[i] * quant_energy[i] }
*
* gc_p in fixed-point arithmetic is calculated as following:
*
* mean = 1/N * sum(i,0,N){ (fc_v[i] / 2^13) * (fc_v[i] / 2^13) } =
* = 1/N * sum(i,0,N) { fc_v[i] * fc_v[i] } / 2^26
*
* Ei = 10log(mean) = -10log(N) - 10log(2^26) +
* + 10log(sum(i,0,N) { fc_v[i] * fc_v[i] })
*
* Ep - Ei + Em = Ep + Em + 10log(N) + 10log(2^26) -
* - 10log(sum(i,0,N) { fc_v[i] * fc_v[i] }) =
* = Ep + mr_energy - 10log(sum(i,0,N) { fc_v[i] * fc_v[i] })
*
* gc_p = 10 ^ ((Ep - Ei + Em) / 20) =
* = 2 ^ (3.3219 * (Ep - Ei + Em) / 20) = 2 ^ (0.166 * (Ep - Ei + Em))
*
* where
*
* mr_energy = Em + 10log(N) + 10log(2^26)
*
* \remark The routine is used in G.729 and AMR (all modes).
*/
int16_t ff_acelp_decode_gain_code(
DSPContext *dsp,
int gain_corr_factor,
const int16_t* fc_v,
int mr_energy,
const int16_t* quant_energy,
const int16_t* ma_prediction_coeff,
int subframe_size,
int max_pred_order);
/**
* Calculate fixed gain (part of section 6.1.3 of AMR spec)
*
* @param fixed_gain_factor gain correction factor
* @param fixed_energy decoded algebraic codebook vector energy
* @param prediction_error vector of the quantified predictor errors of
* the four previous subframes. It is updated by this function.
* @param energy_mean desired mean innovation energy
* @param pred_table table of four moving average coefficients
*/
float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
float *prediction_error, float energy_mean,
const float *pred_table);
/**
* Decode the adaptive codebook index to the integer and fractional parts
* of the pitch lag for one subframe at 1/3 fractional precision.
*
* The choice of pitch lag is described in 3GPP TS 26.090 section 5.6.1.
*
* @param lag_int integer part of pitch lag of the current subframe
* @param lag_frac fractional part of pitch lag of the current subframe
* @param pitch_index parsed adaptive codebook (pitch) index
* @param prev_lag_int integer part of pitch lag for the previous subframe
* @param subframe current subframe number
* @param third_as_first treat the third frame the same way as the first
*/
void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index,
const int prev_lag_int, const int subframe,
int third_as_first, int resolution);
#endif /* AVCODEC_ACELP_PITCH_DELAY_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/acelp_vectors.c
-270
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@@ -1,270 +0,0 @@
/*
* adaptive and fixed codebook vector operations for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <inttypes.h>
#include "avcodec.h"
#include "acelp_vectors.h"
#include "celp_math.h"
const uint8_t ff_fc_2pulses_9bits_track1[16] =
{
1, 3,
6, 8,
11, 13,
16, 18,
21, 23,
26, 28,
31, 33,
36, 38
};
const uint8_t ff_fc_2pulses_9bits_track1_gray[16] =
{
1, 3,
8, 6,
18, 16,
11, 13,
38, 36,
31, 33,
21, 23,
28, 26,
};
const uint8_t ff_fc_2pulses_9bits_track2_gray[32] =
{
0, 2,
5, 4,
12, 10,
7, 9,
25, 24,
20, 22,
14, 15,
19, 17,
36, 31,
21, 26,
1, 6,
16, 11,
27, 29,
32, 30,
39, 37,
34, 35,
};
const uint8_t ff_fc_4pulses_8bits_tracks_13[16] =
{
0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
};
const uint8_t ff_fc_4pulses_8bits_track_4[32] =
{
3, 4,
8, 9,
13, 14,
18, 19,
23, 24,
28, 29,
33, 34,
38, 39,
43, 44,
48, 49,
53, 54,
58, 59,
63, 64,
68, 69,
73, 74,
78, 79,
};
#if 0
static uint8_t gray_decode[32] =
{
0, 1, 3, 2, 7, 6, 4, 5,
15, 14, 12, 13, 8, 9, 11, 10,
31, 30, 28, 29, 24, 25, 27, 26,
16, 17, 19, 18, 23, 22, 20, 21
};
#endif
const float ff_pow_0_7[10] = {
0.700000, 0.490000, 0.343000, 0.240100, 0.168070,
0.117649, 0.082354, 0.057648, 0.040354, 0.028248
};
const float ff_pow_0_75[10] = {
0.750000, 0.562500, 0.421875, 0.316406, 0.237305,
0.177979, 0.133484, 0.100113, 0.075085, 0.056314
};
const float ff_pow_0_55[10] = {
0.550000, 0.302500, 0.166375, 0.091506, 0.050328,
0.027681, 0.015224, 0.008373, 0.004605, 0.002533
};
const float ff_b60_sinc[61] = {
0.898529 , 0.865051 , 0.769257 , 0.624054 , 0.448639 , 0.265289 ,
0.0959167 , -0.0412598 , -0.134338 , -0.178986 , -0.178528 , -0.142609 ,
-0.0849304 , -0.0205078 , 0.0369568 , 0.0773926 , 0.0955200 , 0.0912781 ,
0.0689392 , 0.0357056 , 0. , -0.0305481 , -0.0504150 , -0.0570068 ,
-0.0508423 , -0.0350037 , -0.0141602 , 0.00665283, 0.0230713 , 0.0323486 ,
0.0335388 , 0.0275879 , 0.0167847 , 0.00411987, -0.00747681, -0.0156860 ,
-0.0193481 , -0.0183716 , -0.0137634 , -0.00704956, 0. , 0.00582886 ,
0.00939941, 0.0103760 , 0.00903320, 0.00604248, 0.00238037, -0.00109863 ,
-0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
0.00103760, 0.00222778, 0.00277710, 0.00271606, 0.00213623, 0.00115967 ,
0.
};
void ff_acelp_fc_pulse_per_track(
int16_t* fc_v,
const uint8_t *tab1,
const uint8_t *tab2,
int pulse_indexes,
int pulse_signs,
int pulse_count,
int bits)
{
int mask = (1 << bits) - 1;
int i;
for(i=0; i<pulse_count; i++)
{
fc_v[i + tab1[pulse_indexes & mask]] +=
(pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13)
pulse_indexes >>= bits;
pulse_signs >>= 1;
}
fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
}
void ff_decode_10_pulses_35bits(const int16_t *fixed_index,
AMRFixed *fixed_sparse,
const uint8_t *gray_decode,
int half_pulse_count, int bits)
{
int i;
int mask = (1 << bits) - 1;
fixed_sparse->no_repeat_mask = 0;
fixed_sparse->n = 2 * half_pulse_count;
for (i = 0; i < half_pulse_count; i++) {
const int pos1 = gray_decode[fixed_index[2*i+1] & mask] + i;
const int pos2 = gray_decode[fixed_index[2*i ] & mask] + i;
const float sign = (fixed_index[2*i+1] & (1 << bits)) ? -1.0 : 1.0;
fixed_sparse->x[2*i+1] = pos1;
fixed_sparse->x[2*i ] = pos2;
fixed_sparse->y[2*i+1] = sign;
fixed_sparse->y[2*i ] = pos2 < pos1 ? -sign : sign;
}
}
void ff_acelp_weighted_vector_sum(
int16_t* out,
const int16_t *in_a,
const int16_t *in_b,
int16_t weight_coeff_a,
int16_t weight_coeff_b,
int16_t rounder,
int shift,
int length)
{
int i;
// Clipping required here; breaks OVERFLOW test.
for(i=0; i<length; i++)
out[i] = av_clip_int16((
in_a[i] * weight_coeff_a +
in_b[i] * weight_coeff_b +
rounder) >> shift);
}
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
float weight_coeff_a, float weight_coeff_b, int length)
{
int i;
for(i=0; i<length; i++)
out[i] = weight_coeff_a * in_a[i]
+ weight_coeff_b * in_b[i];
}
void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
int size, float alpha, float *gain_mem)
{
int i;
float postfilter_energ = ff_dot_productf(in, in, size);
float gain_scale_factor = 1.0;
float mem = *gain_mem;
if (postfilter_energ)
gain_scale_factor = sqrt(speech_energ / postfilter_energ);
gain_scale_factor *= 1.0 - alpha;
for (i = 0; i < size; i++) {
mem = alpha * mem + gain_scale_factor;
out[i] = in[i] * mem;
}
*gain_mem = mem;
}
void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
float sum_of_squares, const int n)
{
int i;
float scalefactor = ff_dot_productf(in, in, n);
if (scalefactor)
scalefactor = sqrt(sum_of_squares / scalefactor);
for (i = 0; i < n; i++)
out[i] = in[i] * scalefactor;
}
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
{
int i;
for (i=0; i < in->n; i++) {
int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
float y = in->y[i] * scale;
do {
out[x] += y;
y *= in->pitch_fac;
x += in->pitch_lag;
} while (x < size && repeats);
}
}
void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
{
int i;
for (i=0; i < in->n; i++) {
int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
do {
out[x] = 0.0;
x += in->pitch_lag;
} while (x < size && repeats);
}
}
src/add-ons/media/plugins/ffmpeg/libavcodec/acelp_vectors.h
-264
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@@ -1,264 +0,0 @@
/*
* adaptive and fixed codebook vector operations for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_ACELP_VECTORS_H
#define AVCODEC_ACELP_VECTORS_H
#include <stdint.h>
/** Sparse representation for the algebraic codebook (fixed) vector */
typedef struct {
int n;
int x[10];
float y[10];
int no_repeat_mask;
int pitch_lag;
float pitch_fac;
} AMRFixed;
/**
* Track|Pulse| Positions
* -------------------------------------------------------------------------
* 1 | 0 | 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75
* -------------------------------------------------------------------------
* 2 | 1 | 1, 6, 11, 16, 21, 26, 31, 36, 41, 46, 51, 56, 61, 66, 71, 76
* -------------------------------------------------------------------------
* 3 | 2 | 2, 7, 12, 17, 22, 27, 32, 37, 42, 47, 52, 57, 62, 67, 72, 77
* -------------------------------------------------------------------------
*
* Table contains only first the pulse indexes.
*
* Used in G.729 @8k, G.729 @4.4k, AMR @7.95k, AMR @7.40k
*/
extern const uint8_t ff_fc_4pulses_8bits_tracks_13[16];
/**
* Track|Pulse| Positions
* -------------------------------------------------------------------------
* 4 | 3 | 3, 8, 13, 18, 23, 28, 33, 38, 43, 48, 53, 58, 63, 68, 73, 78
* | | 4, 9, 14, 19, 24, 29, 34, 39, 44, 49, 54, 59, 64, 69, 74, 79
* -------------------------------------------------------------------------
*
* @remark Track in the table should be read top-to-bottom, left-to-right.
*
* Used in G.729 @8k, G.729 @4.4k, AMR @7.95k, AMR @7.40k
*/
extern const uint8_t ff_fc_4pulses_8bits_track_4[32];
/**
* Track|Pulse| Positions
* -----------------------------------------
* 1 | 0 | 1, 6, 11, 16, 21, 26, 31, 36
* | | 3, 8, 13, 18, 23, 28, 33, 38
* -----------------------------------------
*
* @remark Track in the table should be read top-to-bottom, left-to-right.
*
* @note (EE) Reference G.729D code also uses gray decoding for each
* pulse index before looking up the value in the table.
*
* Used in G.729 @6.4k (with gray coding), AMR @5.9k (without gray coding)
*/
extern const uint8_t ff_fc_2pulses_9bits_track1[16];
extern const uint8_t ff_fc_2pulses_9bits_track1_gray[16];
/**
* Track|Pulse| Positions
* -----------------------------------------
* 2 | 1 | 0, 7, 14, 20, 27, 34, 1, 21
* | | 2, 9, 15, 22, 29, 35, 6, 26
* | | 4,10, 17, 24, 30, 37, 11, 31
* | | 5,12, 19, 25, 32, 39, 16, 36
* -----------------------------------------
*
* @remark Track in the table should be read top-to-bottom, left-to-right.
*
* @note (EE.1) This table (from the reference code) does not comply with
* the specification.
* The specification contains the following table:
*
* Track|Pulse| Positions
* -----------------------------------------
* 2 | 1 | 0, 5, 10, 15, 20, 25, 30, 35
* | | 1, 6, 11, 16, 21, 26, 31, 36
* | | 2, 7, 12, 17, 22, 27, 32, 37
* | | 4, 9, 14, 19, 24, 29, 34, 39
*
* -----------------------------------------
*
* @note (EE.2) Reference G.729D code also uses gray decoding for each
* pulse index before looking up the value in the table.
*
* Used in G.729 @6.4k (with gray coding)
*/
extern const uint8_t ff_fc_2pulses_9bits_track2_gray[32];
/**
* b60 hamming windowed sinc function coefficients
*/
extern const float ff_b60_sinc[61];
/**
* Table of pow(0.7,n)
*/
extern const float ff_pow_0_7[10];
/**
* Table of pow(0.75,n)
*/
extern const float ff_pow_0_75[10];
/**
* Table of pow(0.55,n)
*/
extern const float ff_pow_0_55[10];
/**
* Decode fixed-codebook vector (3.8 and D.5.8 of G.729, 5.7.1 of AMR).
* @param fc_v [out] decoded fixed codebook vector (2.13)
* @param tab1 table used for first pulse_count pulses
* @param tab2 table used for last pulse
* @param pulse_indexes fixed codebook indexes
* @param pulse_signs signs of the excitation pulses (0 bit value
* means negative sign)
* @param bits number of bits per one pulse index
* @param pulse_count number of pulses decoded using first table
* @param bits length of one pulse index in bits
*
* Used in G.729 @8k, G.729 @4.4k, G.729 @6.4k, AMR @7.95k, AMR @7.40k
*/
void ff_acelp_fc_pulse_per_track(int16_t* fc_v,
const uint8_t *tab1,
const uint8_t *tab2,
int pulse_indexes,
int pulse_signs,
int pulse_count,
int bits);
/**
* Decode the algebraic codebook index to pulse positions and signs and
* construct the algebraic codebook vector for MODE_12k2.
*
* @note: The positions and signs are explicitly coded in MODE_12k2.
*
* @param fixed_index positions of the ten pulses
* @param fixed_sparse pointer to the algebraic codebook vector
* @param gray_decode gray decoding table
* @param half_pulse_count number of couples of pulses
* @param bits length of one pulse index in bits
*/
void ff_decode_10_pulses_35bits(const int16_t *fixed_index,
AMRFixed *fixed_sparse,
const uint8_t *gray_decode,
int half_pulse_count, int bits);
/**
* weighted sum of two vectors with rounding.
* @param out [out] result of addition
* @param in_a first vector
* @param in_b second vector
* @param weight_coeff_a first vector weight coefficient
* @param weight_coeff_a second vector weight coefficient
* @param rounder this value will be added to the sum of the two vectors
* @param shift result will be shifted to right by this value
* @param length vectors length
*
* @note It is safe to pass the same buffer for out and in_a or in_b.
*
* out[i] = (in_a[i]*weight_a + in_b[i]*weight_b + rounder) >> shift
*/
void ff_acelp_weighted_vector_sum(int16_t* out,
const int16_t *in_a,
const int16_t *in_b,
int16_t weight_coeff_a,
int16_t weight_coeff_b,
int16_t rounder,
int shift,
int length);
/**
* float implementation of weighted sum of two vectors.
* @param out [out] result of addition
* @param in_a first vector
* @param in_b second vector
* @param weight_coeff_a first vector weight coefficient
* @param weight_coeff_a second vector weight coefficient
* @param length vectors length
*
* @note It is safe to pass the same buffer for out and in_a or in_b.
*/
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
float weight_coeff_a, float weight_coeff_b,
int length);
/**
* Adaptive gain control (as used in AMR postfiltering)
*
* @param out output buffer for filtered speech data
* @param in the input speech buffer (may be the same as out)
* @param speech_energ input energy
* @param size the input buffer size
* @param alpha exponential filter factor
* @param gain_mem a pointer to the filter memory (single float of size)
*/
void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
int size, float alpha, float *gain_mem);
/**
* Set the sum of squares of a signal by scaling
*
* @param out output samples
* @param in input samples
* @param sum_of_squares new sum of squares
* @param n number of samples
*
* @note If the input is zero (or its energy underflows), the output is zero.
* This is the behavior of AGC in the AMR reference decoder. The QCELP
* reference decoder seems to have undefined behavior.
*
* TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
* 3GPP TS 26.090 6.1 (6)
*/
void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
float sum_of_squares, const int n);
/**
* Add fixed vector to an array from a sparse representation
*
* @param out fixed vector with pitch sharpening
* @param in sparse fixed vector
* @param scale number to multiply the fixed vector by
* @param size the output vector size
*/
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size);
/**
* Clear array values set by set_fixed_vector
*
* @param out fixed vector to be cleared
* @param in sparse fixed vector
* @param size the output vector size
*/
void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size);
#endif /* AVCODEC_ACELP_VECTORS_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/adpcm.c
-1701
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src/add-ons/media/plugins/ffmpeg/libavcodec/adx.h
-49
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@@ -1,49 +0,0 @@
/*
* ADX ADPCM codecs
* Copyright (c) 2001,2003 BERO
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* SEGA CRI adx codecs.
*
* Reference documents:
* http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
* adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
*/
#ifndef AVCODEC_ADX_H
#define AVCODEC_ADX_H
typedef struct {
int s1,s2;
} PREV;
typedef struct {
PREV prev[2];
int header_parsed;
unsigned char dec_temp[18*2];
int in_temp;
} ADXContext;
#define BASEVOL 0x4000
#define SCALE1 0x7298
#define SCALE2 0x3350
#endif /* AVCODEC_ADX_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/adxdec.c
-180
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/*
* ADX ADPCM codecs
* Copyright (c) 2001,2003 BERO
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "adx.h"
/**
* @file
* SEGA CRI adx codecs.
*
* Reference documents:
* http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
* adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
*/
static av_cold int adx_decode_init(AVCodecContext *avctx)
{
avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
/* 18 bytes <-> 32 samples */
static void adx_decode(short *out,const unsigned char *in,PREV *prev)
{
int scale = AV_RB16(in);
int i;
int s0,s1,s2,d;
// printf("%x ",scale);
in+=2;
s1 = prev->s1;
s2 = prev->s2;
for(i=0;i<16;i++) {
d = in[i];
// d>>=4; if (d&8) d-=16;
d = ((signed char)d >> 4);
s0 = (BASEVOL*d*scale + SCALE1*s1 - SCALE2*s2)>>14;
s2 = s1;
s1 = av_clip_int16(s0);
*out++=s1;
d = in[i];
//d&=15; if (d&8) d-=16;
d = ((signed char)(d<<4) >> 4);
s0 = (BASEVOL*d*scale + SCALE1*s1 - SCALE2*s2)>>14;
s2 = s1;
s1 = av_clip_int16(s0);
*out++=s1;
}
prev->s1 = s1;
prev->s2 = s2;
}
static void adx_decode_stereo(short *out,const unsigned char *in,PREV *prev)
{
short tmp[32*2];
int i;
adx_decode(tmp ,in ,prev);
adx_decode(tmp+32,in+18,prev+1);
for(i=0;i<32;i++) {
out[i*2] = tmp[i];
out[i*2+1] = tmp[i+32];
}
}
/* return data offset or 0 */
static int adx_decode_header(AVCodecContext *avctx,const unsigned char *buf,size_t bufsize)
{
int offset;
if (buf[0]!=0x80) return 0;
offset = (AV_RB32(buf)^0x80000000)+4;
if (bufsize<offset || memcmp(buf+offset-6,"(c)CRI",6)) return 0;
avctx->channels = buf[7];
avctx->sample_rate = AV_RB32(buf+8);
avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;
return offset;
}
static int adx_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf0 = avpkt->data;
int buf_size = avpkt->size;
ADXContext *c = avctx->priv_data;
short *samples = data;
const uint8_t *buf = buf0;
int rest = buf_size;
if (!c->header_parsed) {
int hdrsize = adx_decode_header(avctx,buf,rest);
if (hdrsize==0) return -1;
c->header_parsed = 1;
buf += hdrsize;
rest -= hdrsize;
}
/* 18 bytes of data are expanded into 32*2 bytes of audio,
so guard against buffer overflows */
if(rest/18 > *data_size/64)
rest = (*data_size/64) * 18;
if (c->in_temp) {
int copysize = 18*avctx->channels - c->in_temp;
memcpy(c->dec_temp+c->in_temp,buf,copysize);
rest -= copysize;
buf += copysize;
if (avctx->channels==1) {
adx_decode(samples,c->dec_temp,c->prev);
samples += 32;
} else {
adx_decode_stereo(samples,c->dec_temp,c->prev);
samples += 32*2;
}
}
//
if (avctx->channels==1) {
while(rest>=18) {
adx_decode(samples,buf,c->prev);
rest-=18;
buf+=18;
samples+=32;
}
} else {
while(rest>=18*2) {
adx_decode_stereo(samples,buf,c->prev);
rest-=18*2;
buf+=18*2;
samples+=32*2;
}
}
//
c->in_temp = rest;
if (rest) {
memcpy(c->dec_temp,buf,rest);
buf+=rest;
}
*data_size = (uint8_t*)samples - (uint8_t*)data;
// printf("%d:%d ",buf-buf0,*data_size); fflush(stdout);
return buf-buf0;
}
AVCodec adpcm_adx_decoder = {
"adpcm_adx",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_ADPCM_ADX,
sizeof(ADXContext),
adx_decode_init,
NULL,
NULL,
adx_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/adxenc.c
-197
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@@ -1,197 +0,0 @@
/*
* ADX ADPCM codecs
* Copyright (c) 2001,2003 BERO
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "adx.h"
/**
* @file
* SEGA CRI adx codecs.
*
* Reference documents:
* http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
* adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
*/
/* 18 bytes <-> 32 samples */
static void adx_encode(unsigned char *adx,const short *wav,PREV *prev)
{
int scale;
int i;
int s0,s1,s2,d;
int max=0;
int min=0;
int data[32];
s1 = prev->s1;
s2 = prev->s2;
for(i=0;i<32;i++) {
s0 = wav[i];
d = ((s0<<14) - SCALE1*s1 + SCALE2*s2)/BASEVOL;
data[i]=d;
if (max<d) max=d;
if (min>d) min=d;
s2 = s1;
s1 = s0;
}
prev->s1 = s1;
prev->s2 = s2;
/* -8..+7 */
if (max==0 && min==0) {
memset(adx,0,18);
return;
}
if (max/7>-min/8) scale = max/7;
else scale = -min/8;
if (scale==0) scale=1;
AV_WB16(adx, scale);
for(i=0;i<16;i++) {
adx[i+2] = ((data[i*2]/scale)<<4) | ((data[i*2+1]/scale)&0xf);
}
}
static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t bufsize)
{
#if 0
struct {
uint32_t offset; /* 0x80000000 + sample start - 4 */
unsigned char unknown1[3]; /* 03 12 04 */
unsigned char channel; /* 1 or 2 */
uint32_t freq;
uint32_t size;
uint32_t unknown2; /* 01 f4 03 00 */
uint32_t unknown3; /* 00 00 00 00 */
uint32_t unknown4; /* 00 00 00 00 */
/* if loop
unknown3 00 15 00 01
unknown4 00 00 00 01
long loop_start_sample;
long loop_start_byte;
long loop_end_sample;
long loop_end_byte;
long
*/
} adxhdr; /* big endian */
/* offset-6 "(c)CRI" */
#endif
AV_WB32(buf+0x00,0x80000000|0x20);
AV_WB32(buf+0x04,0x03120400|avctx->channels);
AV_WB32(buf+0x08,avctx->sample_rate);
AV_WB32(buf+0x0c,0); /* FIXME: set after */
AV_WB32(buf+0x10,0x01040300);
AV_WB32(buf+0x14,0x00000000);
AV_WB32(buf+0x18,0x00000000);
memcpy(buf+0x1c,"\0\0(c)CRI",8);
return 0x20+4;
}
static av_cold int adx_encode_init(AVCodecContext *avctx)
{
if (avctx->channels > 2)
return -1; /* only stereo or mono =) */
avctx->frame_size = 32;
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
// avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;
av_log(avctx, AV_LOG_DEBUG, "adx encode init\n");
return 0;
}
static av_cold int adx_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
static int adx_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
{
ADXContext *c = avctx->priv_data;
const short *samples = data;
unsigned char *dst = frame;
int rest = avctx->frame_size;
/*
input data size =
ffmpeg.c: do_audio_out()
frame_bytes = enc->frame_size * 2 * enc->channels;
*/
// printf("sz=%d ",buf_size); fflush(stdout);
if (!c->header_parsed) {
int hdrsize = adx_encode_header(avctx,dst,buf_size);
dst+=hdrsize;
c->header_parsed = 1;
}
if (avctx->channels==1) {
while(rest>=32) {
adx_encode(dst,samples,c->prev);
dst+=18;
samples+=32;
rest-=32;
}
} else {
while(rest>=32*2) {
short tmpbuf[32*2];
int i;
for(i=0;i<32;i++) {
tmpbuf[i] = samples[i*2];
tmpbuf[i+32] = samples[i*2+1];
}
adx_encode(dst,tmpbuf,c->prev);
adx_encode(dst+18,tmpbuf+32,c->prev+1);
dst+=18*2;
samples+=32*2;
rest-=32*2;
}
}
return dst-frame;
}
AVCodec adpcm_adx_encoder = {
"adpcm_adx",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_ADPCM_ADX,
sizeof(ADXContext),
adx_encode_init,
adx_encode_frame,
adx_encode_close,
NULL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/alac.c
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/*
* ALAC (Apple Lossless Audio Codec) decoder
* Copyright (c) 2005 David Hammerton
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ALAC (Apple Lossless Audio Codec) decoder
* @author 2005 David Hammerton
*
* For more information on the ALAC format, visit:
* http://crazney.net/programs/itunes/alac.html
*
* Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
* passed through the extradata[_size] fields. This atom is tacked onto
* the end of an 'alac' stsd atom and has the following format:
* bytes 0-3 atom size (0x24), big-endian
* bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd)
* bytes 8-35 data bytes needed by decoder
*
* Extradata:
* 32bit size
* 32bit tag (=alac)
* 32bit zero?
* 32bit max sample per frame
* 8bit ?? (zero?)
* 8bit sample size
* 8bit history mult
* 8bit initial history
* 8bit kmodifier
* 8bit channels?
* 16bit ??
* 32bit max coded frame size
* 32bit bitrate?
* 32bit samplerate
*/
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "unary.h"
#include "mathops.h"
#define ALAC_EXTRADATA_SIZE 36
#define MAX_CHANNELS 2
typedef struct {
AVCodecContext *avctx;
GetBitContext gb;
/* init to 0; first frame decode should initialize from extradata and
* set this to 1 */
int context_initialized;
int numchannels;
int bytespersample;
/* buffers */
int32_t *predicterror_buffer[MAX_CHANNELS];
int32_t *outputsamples_buffer[MAX_CHANNELS];
int32_t *wasted_bits_buffer[MAX_CHANNELS];
/* stuff from setinfo */
uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
uint8_t setinfo_sample_size; /* 0x10 */
uint8_t setinfo_rice_historymult; /* 0x28 */
uint8_t setinfo_rice_initialhistory; /* 0x0a */
uint8_t setinfo_rice_kmodifier; /* 0x0e */
/* end setinfo stuff */
int wasted_bits;
} ALACContext;
static void allocate_buffers(ALACContext *alac)
{
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
alac->predicterror_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->outputsamples_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
}
}
static int alac_set_info(ALACContext *alac)
{
const unsigned char *ptr = alac->avctx->extradata;
ptr += 4; /* size */
ptr += 4; /* alac */
ptr += 4; /* 0 ? */
if(AV_RB32(ptr) >= UINT_MAX/4){
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
return -1;
}
/* buffer size / 2 ? */
alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
ptr++; /* ??? */
alac->setinfo_sample_size = *ptr++;
if (alac->setinfo_sample_size > 32) {
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
return -1;
}
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
ptr++; /* channels? */
bytestream_get_be16(&ptr); /* ??? */
bytestream_get_be32(&ptr); /* max coded frame size */
bytestream_get_be32(&ptr); /* bitrate ? */
bytestream_get_be32(&ptr); /* samplerate */
allocate_buffers(alac);
return 0;
}
static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
/* read x - number of 1s before 0 represent the rice */
int x = get_unary_0_9(gb);
if (x > 8) { /* RICE THRESHOLD */
/* use alternative encoding */
x = get_bits(gb, readsamplesize);
} else {
if (k >= limit)
k = limit;
if (k != 1) {
int extrabits = show_bits(gb, k);
/* multiply x by 2^k - 1, as part of their strange algorithm */
x = (x << k) - x;
if (extrabits > 1) {
x += extrabits - 1;
skip_bits(gb, k);
} else
skip_bits(gb, k - 1);
}
}
return x;
}
static void bastardized_rice_decompress(ALACContext *alac,
int32_t *output_buffer,
int output_size,
int readsamplesize, /* arg_10 */
int rice_initialhistory, /* arg424->b */
int rice_kmodifier, /* arg424->d */
int rice_historymult, /* arg424->c */
int rice_kmodifier_mask /* arg424->e */
)
{
int output_count;
unsigned int history = rice_initialhistory;
int sign_modifier = 0;
for (output_count = 0; output_count < output_size; output_count++) {
int32_t x;
int32_t x_modified;
int32_t final_val;
/* standard rice encoding */
int k; /* size of extra bits */
/* read k, that is bits as is */
k = av_log2((history >> 9) + 3);
x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
x_modified = sign_modifier + x;
final_val = (x_modified + 1) / 2;
if (x_modified & 1) final_val *= -1;
output_buffer[output_count] = final_val;
sign_modifier = 0;
/* now update the history */
history += x_modified * rice_historymult
- ((history * rice_historymult) >> 9);
if (x_modified > 0xffff)
history = 0xffff;
/* special case: there may be compressed blocks of 0 */
if ((history < 128) && (output_count+1 < output_size)) {
int k;
unsigned int block_size;
sign_modifier = 1;
k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
if (block_size > 0) {
if(block_size >= output_size - output_count){
av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
block_size= output_size - output_count - 1;
}
memset(&output_buffer[output_count+1], 0, block_size * 4);
output_count += block_size;
}
if (block_size > 0xffff)
sign_modifier = 0;
history = 0;
}
}
}
static inline int sign_only(int v)
{
return v ? FFSIGN(v) : 0;
}
static void predictor_decompress_fir_adapt(int32_t *error_buffer,
int32_t *buffer_out,
int output_size,
int readsamplesize,
int16_t *predictor_coef_table,
int predictor_coef_num,
int predictor_quantitization)
{
int i;
/* first sample always copies */
*buffer_out = *error_buffer;
if (!predictor_coef_num) {
if (output_size <= 1)
return;
memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
return;
}
if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
/* second-best case scenario for fir decompression,
* error describes a small difference from the previous sample only
*/
if (output_size <= 1)
return;
for (i = 0; i < output_size - 1; i++) {
int32_t prev_value;
int32_t error_value;
prev_value = buffer_out[i];
error_value = error_buffer[i+1];
buffer_out[i+1] =
sign_extend((prev_value + error_value), readsamplesize);
}
return;
}
/* read warm-up samples */
if (predictor_coef_num > 0)
for (i = 0; i < predictor_coef_num; i++) {
int32_t val;
val = buffer_out[i] + error_buffer[i+1];
val = sign_extend(val, readsamplesize);
buffer_out[i+1] = val;
}
#if 0
/* 4 and 8 are very common cases (the only ones i've seen). these
* should be unrolled and optimized
*/
if (predictor_coef_num == 4) {
/* FIXME: optimized general case */
return;
}
if (predictor_coef_table == 8) {
/* FIXME: optimized general case */
return;
}
#endif
/* general case */
if (predictor_coef_num > 0) {
for (i = predictor_coef_num + 1; i < output_size; i++) {
int j;
int sum = 0;
int outval;
int error_val = error_buffer[i];
for (j = 0; j < predictor_coef_num; j++) {
sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
predictor_coef_table[j];
}
outval = (1 << (predictor_quantitization-1)) + sum;
outval = outval >> predictor_quantitization;
outval = outval + buffer_out[0] + error_val;
outval = sign_extend(outval, readsamplesize);
buffer_out[predictor_coef_num+1] = outval;
if (error_val > 0) {
int predictor_num = predictor_coef_num - 1;
while (predictor_num >= 0 && error_val > 0) {
int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
int sign = sign_only(val);
predictor_coef_table[predictor_num] -= sign;
val *= sign; /* absolute value */
error_val -= ((val >> predictor_quantitization) *
(predictor_coef_num - predictor_num));
predictor_num--;
}
} else if (error_val < 0) {
int predictor_num = predictor_coef_num - 1;
while (predictor_num >= 0 && error_val < 0) {
int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
int sign = - sign_only(val);
predictor_coef_table[predictor_num] -= sign;
val *= sign; /* neg value */
error_val -= ((val >> predictor_quantitization) *
(predictor_coef_num - predictor_num));
predictor_num--;
}
}
buffer_out++;
}
}
}
static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
int16_t *buffer_out,
int numchannels, int numsamples,
uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
{
int i;
if (numsamples <= 0)
return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t a, b;
a = buffer[0][i];
b = buffer[1][i];
a -= (b * interlacing_leftweight) >> interlacing_shift;
b += a;
buffer_out[i*numchannels] = b;
buffer_out[i*numchannels + 1] = a;
}
return;
}
/* otherwise basic interlacing took place */
for (i = 0; i < numsamples; i++) {
int16_t left, right;
left = buffer[0][i];
right = buffer[1][i];
buffer_out[i*numchannels] = left;
buffer_out[i*numchannels + 1] = right;
}
}
static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
int32_t *buffer_out,
int32_t *wasted_bits_buffer[MAX_CHANNELS],
int wasted_bits,
int numchannels, int numsamples,
uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
{
int i;
if (numsamples <= 0)
return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t a, b;
a = buffer[0][i];
b = buffer[1][i];
a -= (b * interlacing_leftweight) >> interlacing_shift;
b += a;
if (wasted_bits) {
b = (b << wasted_bits) | wasted_bits_buffer[0][i];
a = (a << wasted_bits) | wasted_bits_buffer[1][i];
}
buffer_out[i * numchannels] = b << 8;
buffer_out[i * numchannels + 1] = a << 8;
}
} else {
for (i = 0; i < numsamples; i++) {
int32_t left, right;
left = buffer[0][i];
right = buffer[1][i];
if (wasted_bits) {
left = (left << wasted_bits) | wasted_bits_buffer[0][i];
right = (right << wasted_bits) | wasted_bits_buffer[1][i];
}
buffer_out[i * numchannels] = left << 8;
buffer_out[i * numchannels + 1] = right << 8;
}
}
}
static int alac_decode_frame(AVCodecContext *avctx,
void *outbuffer, int *outputsize,
AVPacket *avpkt)
{
const uint8_t *inbuffer = avpkt->data;
int input_buffer_size = avpkt->size;
ALACContext *alac = avctx->priv_data;
int channels;
unsigned int outputsamples;
int hassize;
unsigned int readsamplesize;
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
/* short-circuit null buffers */
if (!inbuffer || !input_buffer_size)
return input_buffer_size;
/* initialize from the extradata */
if (!alac->context_initialized) {
if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
ALAC_EXTRADATA_SIZE);
return input_buffer_size;
}
if (alac_set_info(alac)) {
av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
return input_buffer_size;
}
alac->context_initialized = 1;
}
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
channels = get_bits(&alac->gb, 3) + 1;
if (channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
MAX_CHANNELS);
return input_buffer_size;
}
/* 2^result = something to do with output waiting.
* perhaps matters if we read > 1 frame in a pass?
*/
skip_bits(&alac->gb, 4);
skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */
/* the output sample size is stored soon */
hassize = get_bits1(&alac->gb);
alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
/* whether the frame is compressed */
isnotcompressed = get_bits1(&alac->gb);
if (hassize) {
/* now read the number of samples as a 32bit integer */
outputsamples = get_bits_long(&alac->gb, 32);
if(outputsamples > alac->setinfo_max_samples_per_frame){
av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
return -1;
}
} else
outputsamples = alac->setinfo_max_samples_per_frame;
switch (alac->setinfo_sample_size) {
case 16: avctx->sample_fmt = SAMPLE_FMT_S16;
alac->bytespersample = channels << 1;
break;
case 24: avctx->sample_fmt = SAMPLE_FMT_S32;
alac->bytespersample = channels << 2;
break;
default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
alac->setinfo_sample_size);
return -1;
}
if(outputsamples > *outputsize / alac->bytespersample){
av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
return -1;
}
*outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
return -1;
}
if (!isnotcompressed) {
/* so it is compressed */
int16_t predictor_coef_table[channels][32];
int predictor_coef_num[channels];
int prediction_type[channels];
int prediction_quantitization[channels];
int ricemodifier[channels];
int i, chan;
interlacing_shift = get_bits(&alac->gb, 8);
interlacing_leftweight = get_bits(&alac->gb, 8);
for (chan = 0; chan < channels; chan++) {
prediction_type[chan] = get_bits(&alac->gb, 4);
prediction_quantitization[chan] = get_bits(&alac->gb, 4);
ricemodifier[chan] = get_bits(&alac->gb, 3);
predictor_coef_num[chan] = get_bits(&alac->gb, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num[chan]; i++)
predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
}
if (alac->wasted_bits) {
int i, ch;
for (i = 0; i < outputsamples; i++) {
for (ch = 0; ch < channels; ch++)
alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits);
}
}
for (chan = 0; chan < channels; chan++) {
bastardized_rice_decompress(alac,
alac->predicterror_buffer[chan],
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type[chan] == 0) {
/* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
alac->outputsamples_buffer[chan],
outputsamples,
readsamplesize,
predictor_coef_table[chan],
predictor_coef_num[chan],
prediction_quantitization[chan]);
} else {
av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
/* I think the only other prediction type (or perhaps this is
* just a boolean?) runs adaptive fir twice.. like:
* predictor_decompress_fir_adapt(predictor_error, tempout, ...)
* predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
* little strange..
*/
}
}
} else {
/* not compressed, easy case */
int i, chan;
if (alac->setinfo_sample_size <= 16) {
for (i = 0; i < outputsamples; i++)
for (chan = 0; chan < channels; chan++) {
int32_t audiobits;
audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
alac->outputsamples_buffer[chan][i] = audiobits;
}
} else {
for (i = 0; i < outputsamples; i++) {
for (chan = 0; chan < channels; chan++) {
alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb,
alac->setinfo_sample_size);
alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i],
alac->setinfo_sample_size);
}
}
}
alac->wasted_bits = 0;
interlacing_shift = 0;
interlacing_leftweight = 0;
}
if (get_bits(&alac->gb, 3) != 7)
av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
switch(alac->setinfo_sample_size) {
case 16:
if (channels == 2) {
reconstruct_stereo_16(alac->outputsamples_buffer,
(int16_t*)outbuffer,
alac->numchannels,
outputsamples,
interlacing_shift,
interlacing_leftweight);
} else {
int i;
for (i = 0; i < outputsamples; i++) {
((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
}
}
break;
case 24:
if (channels == 2) {
decorrelate_stereo_24(alac->outputsamples_buffer,
outbuffer,
alac->wasted_bits_buffer,
alac->wasted_bits,
alac->numchannels,
outputsamples,
interlacing_shift,
interlacing_leftweight);
} else {
int i;
for (i = 0; i < outputsamples; i++)
((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
}
break;
}
if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
return input_buffer_size;
}
static av_cold int alac_decode_init(AVCodecContext * avctx)
{
ALACContext *alac = avctx->priv_data;
alac->avctx = avctx;
alac->context_initialized = 0;
alac->numchannels = alac->avctx->channels;
return 0;
}
static av_cold int alac_decode_close(AVCodecContext *avctx)
{
ALACContext *alac = avctx->priv_data;
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
av_freep(&alac->predicterror_buffer[chan]);
av_freep(&alac->outputsamples_buffer[chan]);
av_freep(&alac->wasted_bits_buffer[chan]);
}
return 0;
}
AVCodec alac_decoder = {
"alac",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_ALAC,
sizeof(ALACContext),
alac_decode_init,
NULL,
alac_decode_close,
alac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/alacenc.c
-532
View File
@@ -1,532 +0,0 @@
/**
* ALAC audio encoder
* Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "lpc.h"
#include "mathops.h"
#define DEFAULT_FRAME_SIZE 4096
#define DEFAULT_SAMPLE_SIZE 16
#define MAX_CHANNELS 8
#define ALAC_EXTRADATA_SIZE 36
#define ALAC_FRAME_HEADER_SIZE 55
#define ALAC_FRAME_FOOTER_SIZE 3
#define ALAC_ESCAPE_CODE 0x1FF
#define ALAC_MAX_LPC_ORDER 30
#define DEFAULT_MAX_PRED_ORDER 6
#define DEFAULT_MIN_PRED_ORDER 4
#define ALAC_MAX_LPC_PRECISION 9
#define ALAC_MAX_LPC_SHIFT 9
#define ALAC_CHMODE_LEFT_RIGHT 0
#define ALAC_CHMODE_LEFT_SIDE 1
#define ALAC_CHMODE_RIGHT_SIDE 2
#define ALAC_CHMODE_MID_SIDE 3
typedef struct RiceContext {
int history_mult;
int initial_history;
int k_modifier;
int rice_modifier;
} RiceContext;
typedef struct LPCContext {
int lpc_order;
int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
int lpc_quant;
} LPCContext;
typedef struct AlacEncodeContext {
int compression_level;
int min_prediction_order;
int max_prediction_order;
int max_coded_frame_size;
int write_sample_size;
int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
int32_t predictor_buf[DEFAULT_FRAME_SIZE];
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
RiceContext rc;
LPCContext lpc[MAX_CHANNELS];
DSPContext dspctx;
AVCodecContext *avctx;
} AlacEncodeContext;
static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
{
int ch, i;
for(ch=0;ch<s->avctx->channels;ch++) {
int16_t *sptr = input_samples + ch;
for(i=0;i<s->avctx->frame_size;i++) {
s->sample_buf[ch][i] = *sptr;
sptr += s->avctx->channels;
}
}
}
static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
{
int divisor, q, r;
k = FFMIN(k, s->rc.k_modifier);
divisor = (1<<k) - 1;
q = x / divisor;
r = x % divisor;
if(q > 8) {
// write escape code and sample value directly
put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
put_bits(&s->pbctx, write_sample_size, x);
} else {
if(q)
put_bits(&s->pbctx, q, (1<<q) - 1);
put_bits(&s->pbctx, 1, 0);
if(k != 1) {
if(r > 0)
put_bits(&s->pbctx, k, r+1);
else
put_bits(&s->pbctx, k-1, 0);
}
}
}
static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
{
put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
put_bits(&s->pbctx, 16, 0); // Seems to be zero
put_bits(&s->pbctx, 1, 1); // Sample count is in the header
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame
}
static void calc_predictor_params(AlacEncodeContext *s, int ch)
{
int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
int shift[MAX_LPC_ORDER];
int opt_order;
if (s->compression_level == 1) {
s->lpc[ch].lpc_order = 6;
s->lpc[ch].lpc_quant = 6;
s->lpc[ch].lpc_coeff[0] = 160;
s->lpc[ch].lpc_coeff[1] = -190;
s->lpc[ch].lpc_coeff[2] = 170;
s->lpc[ch].lpc_coeff[3] = -130;
s->lpc[ch].lpc_coeff[4] = 80;
s->lpc[ch].lpc_coeff[5] = -25;
} else {
opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch],
s->avctx->frame_size,
s->min_prediction_order,
s->max_prediction_order,
ALAC_MAX_LPC_PRECISION, coefs, shift, 1,
ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
s->lpc[ch].lpc_order = opt_order;
s->lpc[ch].lpc_quant = shift[opt_order-1];
memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
}
}
static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
{
int i, best;
int32_t lt, rt;
uint64_t sum[4];
uint64_t score[4];
/* calculate sum of 2nd order residual for each channel */
sum[0] = sum[1] = sum[2] = sum[3] = 0;
for(i=2; i<n; i++) {
lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
sum[2] += FFABS((lt + rt) >> 1);
sum[3] += FFABS(lt - rt);
sum[0] += FFABS(lt);
sum[1] += FFABS(rt);
}
/* calculate score for each mode */
score[0] = sum[0] + sum[1];
score[1] = sum[0] + sum[3];
score[2] = sum[1] + sum[3];
score[3] = sum[2] + sum[3];
/* return mode with lowest score */
best = 0;
for(i=1; i<4; i++) {
if(score[i] < score[best]) {
best = i;
}
}
return best;
}
static void alac_stereo_decorrelation(AlacEncodeContext *s)
{
int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
int i, mode, n = s->avctx->frame_size;
int32_t tmp;
mode = estimate_stereo_mode(left, right, n);
switch(mode)
{
case ALAC_CHMODE_LEFT_RIGHT:
s->interlacing_leftweight = 0;
s->interlacing_shift = 0;
break;
case ALAC_CHMODE_LEFT_SIDE:
for(i=0; i<n; i++) {
right[i] = left[i] - right[i];
}
s->interlacing_leftweight = 1;
s->interlacing_shift = 0;
break;
case ALAC_CHMODE_RIGHT_SIDE:
for(i=0; i<n; i++) {
tmp = right[i];
right[i] = left[i] - right[i];
left[i] = tmp + (right[i] >> 31);
}
s->interlacing_leftweight = 1;
s->interlacing_shift = 31;
break;
default:
for(i=0; i<n; i++) {
tmp = left[i];
left[i] = (tmp + right[i]) >> 1;
right[i] = tmp - right[i];
}
s->interlacing_leftweight = 1;
s->interlacing_shift = 1;
break;
}
}
static void alac_linear_predictor(AlacEncodeContext *s, int ch)
{
int i;
LPCContext lpc = s->lpc[ch];
if(lpc.lpc_order == 31) {
s->predictor_buf[0] = s->sample_buf[ch][0];
for(i=1; i<s->avctx->frame_size; i++)
s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
return;
}
// generalised linear predictor
if(lpc.lpc_order > 0) {
int32_t *samples = s->sample_buf[ch];
int32_t *residual = s->predictor_buf;
// generate warm-up samples
residual[0] = samples[0];
for(i=1;i<=lpc.lpc_order;i++)
residual[i] = samples[i] - samples[i-1];
// perform lpc on remaining samples
for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
for (j = 0; j < lpc.lpc_order; j++) {
sum += (samples[lpc.lpc_order-j] - samples[0]) *
lpc.lpc_coeff[j];
}
sum >>= lpc.lpc_quant;
sum += samples[0];
residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
s->write_sample_size);
res_val = residual[i];
if(res_val) {
int index = lpc.lpc_order - 1;
int neg = (res_val < 0);
while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
int val = samples[0] - samples[lpc.lpc_order - index];
int sign = (val ? FFSIGN(val) : 0);
if(neg)
sign*=-1;
lpc.lpc_coeff[index] -= sign;
val *= sign;
res_val -= ((val >> lpc.lpc_quant) *
(lpc.lpc_order - index));
index--;
}
}
samples++;
}
}
}
static void alac_entropy_coder(AlacEncodeContext *s)
{
unsigned int history = s->rc.initial_history;
int sign_modifier = 0, i, k;
int32_t *samples = s->predictor_buf;
for(i=0;i < s->avctx->frame_size;) {
int x;
k = av_log2((history >> 9) + 3);
x = -2*(*samples)-1;
x ^= (x>>31);
samples++;
i++;
encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
history += x * s->rc.history_mult
- ((history * s->rc.history_mult) >> 9);
sign_modifier = 0;
if(x > 0xFFFF)
history = 0xFFFF;
if((history < 128) && (i < s->avctx->frame_size)) {
unsigned int block_size = 0;
k = 7 - av_log2(history) + ((history + 16) >> 6);
while((*samples == 0) && (i < s->avctx->frame_size)) {
samples++;
i++;
block_size++;
}
encode_scalar(s, block_size, k, 16);
sign_modifier = (block_size <= 0xFFFF);
history = 0;
}
}
}
static void write_compressed_frame(AlacEncodeContext *s)
{
int i, j;
if(s->avctx->channels == 2)
alac_stereo_decorrelation(s);
put_bits(&s->pbctx, 8, s->interlacing_shift);
put_bits(&s->pbctx, 8, s->interlacing_leftweight);
for(i=0;i<s->avctx->channels;i++) {
calc_predictor_params(s, i);
put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
put_bits(&s->pbctx, 3, s->rc.rice_modifier);
put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
// predictor coeff. table
for(j=0;j<s->lpc[i].lpc_order;j++) {
put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
}
}
// apply lpc and entropy coding to audio samples
for(i=0;i<s->avctx->channels;i++) {
alac_linear_predictor(s, i);
alac_entropy_coder(s);
}
}
static av_cold int alac_encode_init(AVCodecContext *avctx)
{
AlacEncodeContext *s = avctx->priv_data;
uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
avctx->frame_size = DEFAULT_FRAME_SIZE;
avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
if(avctx->sample_fmt != SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
return -1;
}
// Set default compression level
if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
s->compression_level = 2;
else
s->compression_level = av_clip(avctx->compression_level, 0, 2);
// Initialize default Rice parameters
s->rc.history_mult = 40;
s->rc.initial_history = 10;
s->rc.k_modifier = 14;
s->rc.rice_modifier = 4;
s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3);
s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
AV_WB32(alac_extradata+12, avctx->frame_size);
AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
AV_WB8 (alac_extradata+21, avctx->channels);
AV_WB32(alac_extradata+24, s->max_coded_frame_size);
AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
AV_WB32(alac_extradata+32, avctx->sample_rate);
// Set relevant extradata fields
if(s->compression_level > 0) {
AV_WB8(alac_extradata+18, s->rc.history_mult);
AV_WB8(alac_extradata+19, s->rc.initial_history);
AV_WB8(alac_extradata+20, s->rc.k_modifier);
}
s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
if(avctx->min_prediction_order >= 0) {
if(avctx->min_prediction_order < MIN_LPC_ORDER ||
avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
return -1;
}
s->min_prediction_order = avctx->min_prediction_order;
}
s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
if(avctx->max_prediction_order >= 0) {
if(avctx->max_prediction_order < MIN_LPC_ORDER ||
avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
return -1;
}
s->max_prediction_order = avctx->max_prediction_order;
}
if(s->max_prediction_order < s->min_prediction_order) {
av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
s->min_prediction_order, s->max_prediction_order);
return -1;
}
avctx->extradata = alac_extradata;
avctx->extradata_size = ALAC_EXTRADATA_SIZE;
avctx->coded_frame = avcodec_alloc_frame();
avctx->coded_frame->key_frame = 1;
s->avctx = avctx;
dsputil_init(&s->dspctx, avctx);
return 0;
}
static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
{
AlacEncodeContext *s = avctx->priv_data;
PutBitContext *pb = &s->pbctx;
int i, out_bytes, verbatim_flag = 0;
if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
return -1;
}
if(buf_size < 2*s->max_coded_frame_size) {
av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
return -1;
}
verbatim:
init_put_bits(pb, frame, buf_size);
if((s->compression_level == 0) || verbatim_flag) {
// Verbatim mode
int16_t *samples = data;
write_frame_header(s, 1);
for(i=0; i<avctx->frame_size*avctx->channels; i++) {
put_sbits(pb, 16, *samples++);
}
} else {
init_sample_buffers(s, data);
write_frame_header(s, 0);
write_compressed_frame(s);
}
put_bits(pb, 3, 7);
flush_put_bits(pb);
out_bytes = put_bits_count(pb) >> 3;
if(out_bytes > s->max_coded_frame_size) {
/* frame too large. use verbatim mode */
if(verbatim_flag || (s->compression_level == 0)) {
/* still too large. must be an error. */
av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
return -1;
}
verbatim_flag = 1;
goto verbatim;
}
return out_bytes;
}
static av_cold int alac_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
av_freep(&avctx->coded_frame);
return 0;
}
AVCodec alac_encoder = {
"alac",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_ALAC,
sizeof(AlacEncodeContext),
alac_encode_init,
alac_encode_frame,
alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum SampleFormat[]){ SAMPLE_FMT_S16, SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/allcodecs.c
-385
View File
@@ -1,385 +0,0 @@
/*
* Provides registration of all codecs, parsers and bitstream filters for libavcodec.
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Provides registration of all codecs, parsers and bitstream filters for libavcodec.
*/
#include "avcodec.h"
#define REGISTER_HWACCEL(X,x) { \
extern AVHWAccel x##_hwaccel; \
if(CONFIG_##X##_HWACCEL) av_register_hwaccel(&x##_hwaccel); }
#define REGISTER_ENCODER(X,x) { \
extern AVCodec x##_encoder; \
if(CONFIG_##X##_ENCODER) avcodec_register(&x##_encoder); }
#define REGISTER_DECODER(X,x) { \
extern AVCodec x##_decoder; \
if(CONFIG_##X##_DECODER) avcodec_register(&x##_decoder); }
#define REGISTER_ENCDEC(X,x) REGISTER_ENCODER(X,x); REGISTER_DECODER(X,x)
#define REGISTER_PARSER(X,x) { \
extern AVCodecParser x##_parser; \
if(CONFIG_##X##_PARSER) av_register_codec_parser(&x##_parser); }
#define REGISTER_BSF(X,x) { \
extern AVBitStreamFilter x##_bsf; \
if(CONFIG_##X##_BSF) av_register_bitstream_filter(&x##_bsf); }
void avcodec_register_all(void)
{
static int initialized;
if (initialized)
return;
initialized = 1;
/* hardware accelerators */
REGISTER_HWACCEL (H263_VAAPI, h263_vaapi);
REGISTER_HWACCEL (H264_DXVA2, h264_dxva2);
REGISTER_HWACCEL (H264_VAAPI, h264_vaapi);
REGISTER_HWACCEL (MPEG2_VAAPI, mpeg2_vaapi);
REGISTER_HWACCEL (MPEG4_VAAPI, mpeg4_vaapi);
REGISTER_HWACCEL (VC1_DXVA2, vc1_dxva2);
REGISTER_HWACCEL (VC1_VAAPI, vc1_vaapi);
REGISTER_HWACCEL (WMV3_DXVA2, wmv3_dxva2);
REGISTER_HWACCEL (WMV3_VAAPI, wmv3_vaapi);
/* video codecs */
REGISTER_DECODER (AASC, aasc);
REGISTER_DECODER (AMV, amv);
REGISTER_DECODER (ANM, anm);
REGISTER_ENCDEC (ASV1, asv1);
REGISTER_ENCDEC (ASV2, asv2);
REGISTER_DECODER (AURA, aura);
REGISTER_DECODER (AURA2, aura2);
REGISTER_DECODER (AVS, avs);
REGISTER_DECODER (BETHSOFTVID, bethsoftvid);
REGISTER_DECODER (BFI, bfi);
REGISTER_DECODER (BINK, bink);
REGISTER_ENCDEC (BMP, bmp);
REGISTER_DECODER (C93, c93);
REGISTER_DECODER (CAVS, cavs);
REGISTER_DECODER (CDGRAPHICS, cdgraphics);
REGISTER_DECODER (CINEPAK, cinepak);
REGISTER_DECODER (CLJR, cljr);
REGISTER_DECODER (CSCD, cscd);
REGISTER_DECODER (CYUV, cyuv);
REGISTER_ENCDEC (DNXHD, dnxhd);
REGISTER_DECODER (DPX, dpx);
REGISTER_DECODER (DSICINVIDEO, dsicinvideo);
REGISTER_ENCDEC (DVVIDEO, dvvideo);
REGISTER_DECODER (DXA, dxa);
REGISTER_DECODER (EACMV, eacmv);
REGISTER_DECODER (EAMAD, eamad);
REGISTER_DECODER (EATGQ, eatgq);
REGISTER_DECODER (EATGV, eatgv);
REGISTER_DECODER (EATQI, eatqi);
REGISTER_DECODER (EIGHTBPS, eightbps);
REGISTER_DECODER (EIGHTSVX_EXP, eightsvx_exp);
REGISTER_DECODER (EIGHTSVX_FIB, eightsvx_fib);
REGISTER_DECODER (ESCAPE124, escape124);
REGISTER_ENCDEC (FFV1, ffv1);
REGISTER_ENCDEC (FFVHUFF, ffvhuff);
REGISTER_ENCDEC (FLASHSV, flashsv);
REGISTER_DECODER (FLIC, flic);
REGISTER_ENCDEC (FLV, flv);
REGISTER_DECODER (FOURXM, fourxm);
REGISTER_DECODER (FRAPS, fraps);
REGISTER_DECODER (FRWU, frwu);
REGISTER_ENCDEC (GIF, gif);
REGISTER_ENCDEC (H261, h261);
REGISTER_ENCDEC (H263, h263);
REGISTER_DECODER (H263I, h263i);
REGISTER_ENCODER (H263P, h263p);
REGISTER_DECODER (H264, h264);
REGISTER_DECODER (H264_VDPAU, h264_vdpau);
REGISTER_ENCDEC (HUFFYUV, huffyuv);
REGISTER_DECODER (IDCIN, idcin);
REGISTER_DECODER (IFF_BYTERUN1, iff_byterun1);
REGISTER_DECODER (IFF_ILBM, iff_ilbm);
REGISTER_DECODER (INDEO2, indeo2);
REGISTER_DECODER (INDEO3, indeo3);
REGISTER_DECODER (INDEO5, indeo5);
REGISTER_DECODER (INTERPLAY_VIDEO, interplay_video);
REGISTER_ENCDEC (JPEGLS, jpegls);
REGISTER_DECODER (KGV1, kgv1);
REGISTER_DECODER (KMVC, kmvc);
REGISTER_ENCODER (LJPEG, ljpeg);
REGISTER_DECODER (LOCO, loco);
REGISTER_DECODER (MDEC, mdec);
REGISTER_DECODER (MIMIC, mimic);
REGISTER_ENCDEC (MJPEG, mjpeg);
REGISTER_DECODER (MJPEGB, mjpegb);
REGISTER_DECODER (MMVIDEO, mmvideo);
REGISTER_DECODER (MOTIONPIXELS, motionpixels);
REGISTER_DECODER (MPEG_XVMC, mpeg_xvmc);
REGISTER_ENCDEC (MPEG1VIDEO, mpeg1video);
REGISTER_ENCDEC (MPEG2VIDEO, mpeg2video);
REGISTER_ENCDEC (MPEG4, mpeg4);
REGISTER_DECODER (MPEG4_VDPAU, mpeg4_vdpau);
REGISTER_DECODER (MPEGVIDEO, mpegvideo);
REGISTER_DECODER (MPEG_VDPAU, mpeg_vdpau);
REGISTER_DECODER (MPEG1_VDPAU, mpeg1_vdpau);
REGISTER_ENCDEC (MSMPEG4V1, msmpeg4v1);
REGISTER_ENCDEC (MSMPEG4V2, msmpeg4v2);
REGISTER_ENCDEC (MSMPEG4V3, msmpeg4v3);
REGISTER_DECODER (MSRLE, msrle);
REGISTER_DECODER (MSVIDEO1, msvideo1);
REGISTER_DECODER (MSZH, mszh);
REGISTER_DECODER (NUV, nuv);
REGISTER_ENCDEC (PAM, pam);
REGISTER_ENCDEC (PBM, pbm);
REGISTER_ENCDEC (PCX, pcx);
REGISTER_ENCDEC (PGM, pgm);
REGISTER_ENCDEC (PGMYUV, pgmyuv);
REGISTER_ENCDEC (PNG, png);
REGISTER_ENCDEC (PPM, ppm);
REGISTER_DECODER (PTX, ptx);
REGISTER_DECODER (QDRAW, qdraw);
REGISTER_DECODER (QPEG, qpeg);
REGISTER_ENCDEC (QTRLE, qtrle);
REGISTER_DECODER (R210, r210);
REGISTER_ENCDEC (RAWVIDEO, rawvideo);
REGISTER_DECODER (RL2, rl2);
REGISTER_ENCDEC (ROQ, roq);
REGISTER_DECODER (RPZA, rpza);
REGISTER_ENCDEC (RV10, rv10);
REGISTER_ENCDEC (RV20, rv20);
REGISTER_DECODER (RV30, rv30);
REGISTER_DECODER (RV40, rv40);
REGISTER_ENCDEC (SGI, sgi);
REGISTER_DECODER (SMACKER, smacker);
REGISTER_DECODER (SMC, smc);
REGISTER_ENCDEC (SNOW, snow);
REGISTER_DECODER (SP5X, sp5x);
REGISTER_DECODER (SUNRAST, sunrast);
REGISTER_ENCDEC (SVQ1, svq1);
REGISTER_DECODER (SVQ3, svq3);
REGISTER_ENCDEC (TARGA, targa);
REGISTER_DECODER (THEORA, theora);
REGISTER_DECODER (THP, thp);
REGISTER_DECODER (TIERTEXSEQVIDEO, tiertexseqvideo);
REGISTER_ENCDEC (TIFF, tiff);
REGISTER_DECODER (TMV, tmv);
REGISTER_DECODER (TRUEMOTION1, truemotion1);
REGISTER_DECODER (TRUEMOTION2, truemotion2);
REGISTER_DECODER (TSCC, tscc);
REGISTER_DECODER (TXD, txd);
REGISTER_DECODER (ULTI, ulti);
REGISTER_ENCDEC (V210, v210);
REGISTER_DECODER (V210X, v210x);
REGISTER_DECODER (VB, vb);
REGISTER_DECODER (VC1, vc1);
REGISTER_DECODER (VC1_VDPAU, vc1_vdpau);
REGISTER_DECODER (VCR1, vcr1);
REGISTER_DECODER (VMDVIDEO, vmdvideo);
REGISTER_DECODER (VMNC, vmnc);
REGISTER_DECODER (VP3, vp3);
REGISTER_DECODER (VP5, vp5);
REGISTER_DECODER (VP6, vp6);
REGISTER_DECODER (VP6A, vp6a);
REGISTER_DECODER (VP6F, vp6f);
REGISTER_DECODER (VQA, vqa);
REGISTER_ENCDEC (WMV1, wmv1);
REGISTER_ENCDEC (WMV2, wmv2);
REGISTER_DECODER (WMV3, wmv3);
REGISTER_DECODER (WMV3_VDPAU, wmv3_vdpau);
REGISTER_DECODER (WNV1, wnv1);
REGISTER_DECODER (XAN_WC3, xan_wc3);
REGISTER_DECODER (XL, xl);
REGISTER_DECODER (YOP, yop);
REGISTER_ENCDEC (ZLIB, zlib);
REGISTER_ENCDEC (ZMBV, zmbv);
/* audio codecs */
REGISTER_ENCDEC (AAC, aac);
REGISTER_ENCDEC (AC3, ac3);
REGISTER_ENCDEC (ALAC, alac);
REGISTER_DECODER (ALS, als);
REGISTER_DECODER (AMRNB, amrnb);
REGISTER_DECODER (APE, ape);
REGISTER_DECODER (ATRAC1, atrac1);
REGISTER_DECODER (ATRAC3, atrac3);
REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER (COOK, cook);
REGISTER_DECODER (DCA, dca);
REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
REGISTER_DECODER (EAC3, eac3);
REGISTER_ENCDEC (FLAC, flac);
REGISTER_DECODER (IMC, imc);
REGISTER_DECODER (MACE3, mace3);
REGISTER_DECODER (MACE6, mace6);
REGISTER_DECODER (MLP, mlp);
REGISTER_DECODER (MP1, mp1);
REGISTER_ENCDEC (MP2, mp2);
REGISTER_DECODER (MP3, mp3);
REGISTER_DECODER (MP3ADU, mp3adu);
REGISTER_DECODER (MP3ON4, mp3on4);
REGISTER_DECODER (MPC7, mpc7);
REGISTER_DECODER (MPC8, mpc8);
REGISTER_ENCDEC (NELLYMOSER, nellymoser);
REGISTER_DECODER (QCELP, qcelp);
REGISTER_DECODER (QDM2, qdm2);
REGISTER_DECODER (RA_144, ra_144);
REGISTER_DECODER (RA_288, ra_288);
REGISTER_DECODER (SHORTEN, shorten);
REGISTER_DECODER (SIPR, sipr);
REGISTER_DECODER (SMACKAUD, smackaud);
REGISTER_ENCDEC (SONIC, sonic);
REGISTER_ENCODER (SONIC_LS, sonic_ls);
REGISTER_DECODER (TRUEHD, truehd);
REGISTER_DECODER (TRUESPEECH, truespeech);
REGISTER_DECODER (TTA, tta);
REGISTER_DECODER (TWINVQ, twinvq);
REGISTER_DECODER (VMDAUDIO, vmdaudio);
REGISTER_DECODER (VORBIS, vorbis);
REGISTER_DECODER (WAVPACK, wavpack);
REGISTER_DECODER (WMAPRO, wmapro);
REGISTER_ENCDEC (WMAV1, wmav1);
REGISTER_ENCDEC (WMAV2, wmav2);
REGISTER_DECODER (WMAVOICE, wmavoice);
REGISTER_DECODER (WS_SND1, ws_snd1);
/* PCM codecs */
REGISTER_ENCDEC (PCM_ALAW, pcm_alaw);
REGISTER_DECODER (PCM_BLURAY, pcm_bluray);
REGISTER_DECODER (PCM_DVD, pcm_dvd);
REGISTER_ENCDEC (PCM_F32BE, pcm_f32be);
REGISTER_ENCDEC (PCM_F32LE, pcm_f32le);
REGISTER_ENCDEC (PCM_F64BE, pcm_f64be);
REGISTER_ENCDEC (PCM_F64LE, pcm_f64le);
REGISTER_ENCDEC (PCM_MULAW, pcm_mulaw);
REGISTER_ENCDEC (PCM_S8, pcm_s8);
REGISTER_ENCDEC (PCM_S16BE, pcm_s16be);
REGISTER_ENCDEC (PCM_S16LE, pcm_s16le);
REGISTER_DECODER (PCM_S16LE_PLANAR, pcm_s16le_planar);
REGISTER_ENCDEC (PCM_S24BE, pcm_s24be);
REGISTER_ENCDEC (PCM_S24DAUD, pcm_s24daud);
REGISTER_ENCDEC (PCM_S24LE, pcm_s24le);
REGISTER_ENCDEC (PCM_S32BE, pcm_s32be);
REGISTER_ENCDEC (PCM_S32LE, pcm_s32le);
REGISTER_ENCDEC (PCM_U8, pcm_u8);
REGISTER_ENCDEC (PCM_U16BE, pcm_u16be);
REGISTER_ENCDEC (PCM_U16LE, pcm_u16le);
REGISTER_ENCDEC (PCM_U24BE, pcm_u24be);
REGISTER_ENCDEC (PCM_U24LE, pcm_u24le);
REGISTER_ENCDEC (PCM_U32BE, pcm_u32be);
REGISTER_ENCDEC (PCM_U32LE, pcm_u32le);
REGISTER_ENCDEC (PCM_ZORK , pcm_zork);
/* DPCM codecs */
REGISTER_DECODER (INTERPLAY_DPCM, interplay_dpcm);
REGISTER_ENCDEC (ROQ_DPCM, roq_dpcm);
REGISTER_DECODER (SOL_DPCM, sol_dpcm);
REGISTER_DECODER (XAN_DPCM, xan_dpcm);
/* ADPCM codecs */
REGISTER_DECODER (ADPCM_4XM, adpcm_4xm);
REGISTER_ENCDEC (ADPCM_ADX, adpcm_adx);
REGISTER_DECODER (ADPCM_CT, adpcm_ct);
REGISTER_DECODER (ADPCM_EA, adpcm_ea);
REGISTER_DECODER (ADPCM_EA_MAXIS_XA, adpcm_ea_maxis_xa);
REGISTER_DECODER (ADPCM_EA_R1, adpcm_ea_r1);
REGISTER_DECODER (ADPCM_EA_R2, adpcm_ea_r2);
REGISTER_DECODER (ADPCM_EA_R3, adpcm_ea_r3);
REGISTER_DECODER (ADPCM_EA_XAS, adpcm_ea_xas);
REGISTER_ENCDEC (ADPCM_G726, adpcm_g726);
REGISTER_DECODER (ADPCM_IMA_AMV, adpcm_ima_amv);
REGISTER_DECODER (ADPCM_IMA_DK3, adpcm_ima_dk3);
REGISTER_DECODER (ADPCM_IMA_DK4, adpcm_ima_dk4);
REGISTER_DECODER (ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs);
REGISTER_DECODER (ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead);
REGISTER_DECODER (ADPCM_IMA_ISS, adpcm_ima_iss);
REGISTER_ENCDEC (ADPCM_IMA_QT, adpcm_ima_qt);
REGISTER_DECODER (ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg);
REGISTER_ENCDEC (ADPCM_IMA_WAV, adpcm_ima_wav);
REGISTER_DECODER (ADPCM_IMA_WS, adpcm_ima_ws);
REGISTER_ENCDEC (ADPCM_MS, adpcm_ms);
REGISTER_DECODER (ADPCM_SBPRO_2, adpcm_sbpro_2);
REGISTER_DECODER (ADPCM_SBPRO_3, adpcm_sbpro_3);
REGISTER_DECODER (ADPCM_SBPRO_4, adpcm_sbpro_4);
REGISTER_ENCDEC (ADPCM_SWF, adpcm_swf);
REGISTER_DECODER (ADPCM_THP, adpcm_thp);
REGISTER_DECODER (ADPCM_XA, adpcm_xa);
REGISTER_ENCDEC (ADPCM_YAMAHA, adpcm_yamaha);
/* subtitles */
REGISTER_ENCDEC (DVBSUB, dvbsub);
REGISTER_ENCDEC (DVDSUB, dvdsub);
REGISTER_DECODER (PGSSUB, pgssub);
REGISTER_ENCDEC (XSUB, xsub);
/* external libraries */
REGISTER_ENCDEC (LIBDIRAC, libdirac);
REGISTER_ENCODER (LIBFAAC, libfaac);
REGISTER_DECODER (LIBFAAD, libfaad);
REGISTER_ENCDEC (LIBGSM, libgsm);
REGISTER_ENCDEC (LIBGSM_MS, libgsm_ms);
REGISTER_ENCODER (LIBMP3LAME, libmp3lame);
REGISTER_ENCDEC (LIBOPENCORE_AMRNB, libopencore_amrnb);
REGISTER_DECODER (LIBOPENCORE_AMRWB, libopencore_amrwb);
REGISTER_DECODER (LIBOPENJPEG, libopenjpeg);
REGISTER_ENCDEC (LIBSCHROEDINGER, libschroedinger);
REGISTER_DECODER (LIBSPEEX, libspeex);
REGISTER_ENCODER (LIBTHEORA, libtheora);
REGISTER_ENCODER (LIBVORBIS, libvorbis);
REGISTER_ENCDEC (LIBVPX, libvpx);
REGISTER_ENCODER (LIBX264, libx264);
REGISTER_ENCODER (LIBXVID, libxvid);
/* parsers */
REGISTER_PARSER (AAC, aac);
REGISTER_PARSER (AC3, ac3);
REGISTER_PARSER (CAVSVIDEO, cavsvideo);
REGISTER_PARSER (DCA, dca);
REGISTER_PARSER (DIRAC, dirac);
REGISTER_PARSER (DNXHD, dnxhd);
REGISTER_PARSER (DVBSUB, dvbsub);
REGISTER_PARSER (DVDSUB, dvdsub);
REGISTER_PARSER (H261, h261);
REGISTER_PARSER (H263, h263);
REGISTER_PARSER (H264, h264);
REGISTER_PARSER (MJPEG, mjpeg);
REGISTER_PARSER (MLP, mlp);
REGISTER_PARSER (MPEG4VIDEO, mpeg4video);
REGISTER_PARSER (MPEGAUDIO, mpegaudio);
REGISTER_PARSER (MPEGVIDEO, mpegvideo);
REGISTER_PARSER (PNM, pnm);
REGISTER_PARSER (VC1, vc1);
REGISTER_PARSER (VP3, vp3);
/* bitstream filters */
REGISTER_BSF (AAC_ADTSTOASC, aac_adtstoasc);
REGISTER_BSF (DUMP_EXTRADATA, dump_extradata);
REGISTER_BSF (H264_MP4TOANNEXB, h264_mp4toannexb);
REGISTER_BSF (IMX_DUMP_HEADER, imx_dump_header);
REGISTER_BSF (MJPEGA_DUMP_HEADER, mjpega_dump_header);
REGISTER_BSF (MP3_HEADER_COMPRESS, mp3_header_compress);
REGISTER_BSF (MP3_HEADER_DECOMPRESS, mp3_header_decompress);
REGISTER_BSF (MOV2TEXTSUB, mov2textsub);
REGISTER_BSF (NOISE, noise);
REGISTER_BSF (REMOVE_EXTRADATA, remove_extradata);
REGISTER_BSF (TEXT2MOVSUB, text2movsub);
}
src/add-ons/media/plugins/ffmpeg/libavcodec/alsdec.c
-1636
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File diff suppressed because it is too large Load Diff
src/add-ons/media/plugins/ffmpeg/libavcodec/amrnbdata.h
-1672
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src/add-ons/media/plugins/ffmpeg/libavcodec/amrnbdec.c
-1076
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File diff suppressed because it is too large Load Diff
src/add-ons/media/plugins/ffmpeg/libavcodec/anm.c
-197
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@@ -1,197 +0,0 @@
/*
* Deluxe Paint Animation decoder
* Copyright (c) 2009 Peter Ross
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Deluxe Paint Animation decoder
*/
#include "avcodec.h"
#include "bytestream.h"
typedef struct AnmContext {
AVFrame frame;
int x; ///< x coordinate position
} AnmContext;
static av_cold int decode_init(AVCodecContext *avctx)
{
AnmContext *s = avctx->priv_data;
const uint8_t *buf;
int i;
avctx->pix_fmt = PIX_FMT_PAL8;
if (avctx->extradata_size != 16*8 + 4*256)
return -1;
s->frame.reference = 1;
if (avctx->get_buffer(avctx, &s->frame) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
buf = avctx->extradata + 16*8;
for (i = 0; i < 256; i++)
((uint32_t*)s->frame.data[1])[i] = bytestream_get_le32(&buf);
return 0;
}
/**
* Perform decode operation
* @param dst, dst_end Destination image buffer
* @param buf, buf_end Source buffer (optional, see below)
* @param pixel Fill color (optional, see below)
* @param count Pixel count
* @param x Pointer to x-axis counter
* @param width Image width
* @param linesize Destination image buffer linesize
* @return non-zero if destination buffer is exhausted
*
* a copy operation is achieved when 'buf' is set
* a fill operation is acheived when 'buf' is null and pixel is >= 0
* a skip operation is acheived when 'buf' is null and pixel is < 0
*/
static inline int op(uint8_t **dst, const uint8_t *dst_end,
const uint8_t **buf, const uint8_t *buf_end,
int pixel, int count,
int *x, int width, int linesize)
{
int remaining = width - *x;
while(count > 0) {
int striplen = FFMIN(count, remaining);
if (buf) {
striplen = FFMIN(striplen, buf_end - *buf);
memcpy(*dst, *buf, striplen);
*buf += striplen;
} else if (pixel >= 0)
memset(*dst, pixel, striplen);
*dst += striplen;
remaining -= striplen;
count -= striplen;
if (remaining <= 0) {
*dst += linesize - width;
remaining = width;
}
if (linesize > 0) {
if (*dst >= dst_end) goto exhausted;
} else {
if (*dst <= dst_end) goto exhausted;
}
}
*x = width - remaining;
return 0;
exhausted:
*x = width - remaining;
return 1;
}
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
AnmContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
const int buf_size = avpkt->size;
const uint8_t *buf_end = buf + buf_size;
uint8_t *dst, *dst_end;
int count;
if(avctx->reget_buffer(avctx, &s->frame) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
dst = s->frame.data[0];
dst_end = s->frame.data[0] + s->frame.linesize[0]*avctx->height;
if (buf[0] != 0x42) {
av_log_ask_for_sample(avctx, "unknown record type\n");
return buf_size;
}
if (buf[1]) {
av_log_ask_for_sample(avctx, "padding bytes not supported\n");
return buf_size;
}
buf += 4;
s->x = 0;
do {
/* if statements are ordered by probability */
#define OP(buf, pixel, count) \
op(&dst, dst_end, (buf), buf_end, (pixel), (count), &s->x, avctx->width, s->frame.linesize[0])
int type = bytestream_get_byte(&buf);
count = type & 0x7F;
type >>= 7;
if (count) {
if (OP(type ? NULL : &buf, -1, count)) break;
} else if (!type) {
int pixel;
count = bytestream_get_byte(&buf); /* count==0 gives nop */
pixel = bytestream_get_byte(&buf);
if (OP(NULL, pixel, count)) break;
} else {
int pixel;
type = bytestream_get_le16(&buf);
count = type & 0x3FFF;
type >>= 14;
if (!count) {
if (type == 0)
break; // stop
if (type == 2) {
av_log_ask_for_sample(avctx, "unknown opcode");
return AVERROR_INVALIDDATA;
}
continue;
}
pixel = type == 3 ? bytestream_get_byte(&buf) : -1;
if (type == 1) count += 0x4000;
if (OP(type == 2 ? &buf : NULL, pixel, count)) break;
}
} while (buf + 1 < buf_end);
*data_size = sizeof(AVFrame);
*(AVFrame*)data = s->frame;
return buf_size;
}
static av_cold int decode_end(AVCodecContext *avctx)
{
AnmContext *s = avctx->priv_data;
if (s->frame.data[0])
avctx->release_buffer(avctx, &s->frame);
return 0;
}
AVCodec anm_decoder = {
"anm",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_ANM,
sizeof(AnmContext),
decode_init,
NULL,
decode_end,
decode_frame,
CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Deluxe Paint Animation"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/apedec.c
-891
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@@ -1,891 +0,0 @@
/*
* Monkey's Audio lossless audio decoder
* Copyright (c) 2007 Benjamin Zores <ben@geexbox.org>
* based upon libdemac from Dave Chapman.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define ALT_BITSTREAM_READER_LE
#include "avcodec.h"
#include "dsputil.h"
#include "get_bits.h"
#include "bytestream.h"
/**
* @file
* Monkey's Audio lossless audio decoder
*/
#define BLOCKS_PER_LOOP 4608
#define MAX_CHANNELS 2
#define MAX_BYTESPERSAMPLE 3
#define APE_FRAMECODE_MONO_SILENCE 1
#define APE_FRAMECODE_STEREO_SILENCE 3
#define APE_FRAMECODE_PSEUDO_STEREO 4
#define HISTORY_SIZE 512
#define PREDICTOR_ORDER 8
/** Total size of all predictor histories */
#define PREDICTOR_SIZE 50
#define YDELAYA (18 + PREDICTOR_ORDER*4)
#define YDELAYB (18 + PREDICTOR_ORDER*3)
#define XDELAYA (18 + PREDICTOR_ORDER*2)
#define XDELAYB (18 + PREDICTOR_ORDER)
#define YADAPTCOEFFSA 18
#define XADAPTCOEFFSA 14
#define YADAPTCOEFFSB 10
#define XADAPTCOEFFSB 5
/**
* Possible compression levels
* @{
*/
enum APECompressionLevel {
COMPRESSION_LEVEL_FAST = 1000,
COMPRESSION_LEVEL_NORMAL = 2000,
COMPRESSION_LEVEL_HIGH = 3000,
COMPRESSION_LEVEL_EXTRA_HIGH = 4000,
COMPRESSION_LEVEL_INSANE = 5000
};
/** @} */
#define APE_FILTER_LEVELS 3
/** Filter orders depending on compression level */
static const uint16_t ape_filter_orders[5][APE_FILTER_LEVELS] = {
{ 0, 0, 0 },
{ 16, 0, 0 },
{ 64, 0, 0 },
{ 32, 256, 0 },
{ 16, 256, 1280 }
};
/** Filter fraction bits depending on compression level */
static const uint8_t ape_filter_fracbits[5][APE_FILTER_LEVELS] = {
{ 0, 0, 0 },
{ 11, 0, 0 },
{ 11, 0, 0 },
{ 10, 13, 0 },
{ 11, 13, 15 }
};
/** Filters applied to the decoded data */
typedef struct APEFilter {
int16_t *coeffs; ///< actual coefficients used in filtering
int16_t *adaptcoeffs; ///< adaptive filter coefficients used for correcting of actual filter coefficients
int16_t *historybuffer; ///< filter memory
int16_t *delay; ///< filtered values
int avg;
} APEFilter;
typedef struct APERice {
uint32_t k;
uint32_t ksum;
} APERice;
typedef struct APERangecoder {
uint32_t low; ///< low end of interval
uint32_t range; ///< length of interval
uint32_t help; ///< bytes_to_follow resp. intermediate value
unsigned int buffer; ///< buffer for input/output
} APERangecoder;
/** Filter histories */
typedef struct APEPredictor {
int32_t *buf;
int32_t lastA[2];
int32_t filterA[2];
int32_t filterB[2];
int32_t coeffsA[2][4]; ///< adaption coefficients
int32_t coeffsB[2][5]; ///< adaption coefficients
int32_t historybuffer[HISTORY_SIZE + PREDICTOR_SIZE];
} APEPredictor;
/** Decoder context */
typedef struct APEContext {
AVCodecContext *avctx;
DSPContext dsp;
int channels;
int samples; ///< samples left to decode in current frame
int fileversion; ///< codec version, very important in decoding process
int compression_level; ///< compression levels
int fset; ///< which filter set to use (calculated from compression level)
int flags; ///< global decoder flags
uint32_t CRC; ///< frame CRC
int frameflags; ///< frame flags
int currentframeblocks; ///< samples (per channel) in current frame
int blocksdecoded; ///< count of decoded samples in current frame
APEPredictor predictor; ///< predictor used for final reconstruction
int32_t decoded0[BLOCKS_PER_LOOP]; ///< decoded data for the first channel
int32_t decoded1[BLOCKS_PER_LOOP]; ///< decoded data for the second channel
int16_t* filterbuf[APE_FILTER_LEVELS]; ///< filter memory
APERangecoder rc; ///< rangecoder used to decode actual values
APERice riceX; ///< rice code parameters for the second channel
APERice riceY; ///< rice code parameters for the first channel
APEFilter filters[APE_FILTER_LEVELS][2]; ///< filters used for reconstruction
uint8_t *data; ///< current frame data
uint8_t *data_end; ///< frame data end
const uint8_t *ptr; ///< current position in frame data
const uint8_t *last_ptr; ///< position where last 4608-sample block ended
int error;
} APEContext;
// TODO: dsputilize
static av_cold int ape_decode_init(AVCodecContext * avctx)
{
APEContext *s = avctx->priv_data;
int i;
if (avctx->extradata_size != 6) {
av_log(avctx, AV_LOG_ERROR, "Incorrect extradata\n");
return -1;
}
if (avctx->bits_per_coded_sample != 16) {
av_log(avctx, AV_LOG_ERROR, "Only 16-bit samples are supported\n");
return -1;
}
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo is supported\n");
return -1;
}
s->avctx = avctx;
s->channels = avctx->channels;
s->fileversion = AV_RL16(avctx->extradata);
s->compression_level = AV_RL16(avctx->extradata + 2);
s->flags = AV_RL16(avctx->extradata + 4);
av_log(avctx, AV_LOG_DEBUG, "Compression Level: %d - Flags: %d\n", s->compression_level, s->flags);
if (s->compression_level % 1000 || s->compression_level > COMPRESSION_LEVEL_INSANE) {
av_log(avctx, AV_LOG_ERROR, "Incorrect compression level %d\n", s->compression_level);
return -1;
}
s->fset = s->compression_level / 1000 - 1;
for (i = 0; i < APE_FILTER_LEVELS; i++) {
if (!ape_filter_orders[s->fset][i])
break;
s->filterbuf[i] = av_malloc((ape_filter_orders[s->fset][i] * 3 + HISTORY_SIZE) * 4);
}
dsputil_init(&s->dsp, avctx);
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
return 0;
}
static av_cold int ape_decode_close(AVCodecContext * avctx)
{
APEContext *s = avctx->priv_data;
int i;
for (i = 0; i < APE_FILTER_LEVELS; i++)
av_freep(&s->filterbuf[i]);
av_freep(&s->data);
return 0;
}
/**
* @defgroup rangecoder APE range decoder
* @{
*/
#define CODE_BITS 32
#define TOP_VALUE ((unsigned int)1 << (CODE_BITS-1))
#define SHIFT_BITS (CODE_BITS - 9)
#define EXTRA_BITS ((CODE_BITS-2) % 8 + 1)
#define BOTTOM_VALUE (TOP_VALUE >> 8)
/** Start the decoder */
static inline void range_start_decoding(APEContext * ctx)
{
ctx->rc.buffer = bytestream_get_byte(&ctx->ptr);
ctx->rc.low = ctx->rc.buffer >> (8 - EXTRA_BITS);
ctx->rc.range = (uint32_t) 1 << EXTRA_BITS;
}
/** Perform normalization */
static inline void range_dec_normalize(APEContext * ctx)
{
while (ctx->rc.range <= BOTTOM_VALUE) {
ctx->rc.buffer <<= 8;
if(ctx->ptr < ctx->data_end)
ctx->rc.buffer += *ctx->ptr;
ctx->ptr++;
ctx->rc.low = (ctx->rc.low << 8) | ((ctx->rc.buffer >> 1) & 0xFF);
ctx->rc.range <<= 8;
}
}
/**
* Calculate culmulative frequency for next symbol. Does NO update!
* @param ctx decoder context
* @param tot_f is the total frequency or (code_value)1<<shift
* @return the culmulative frequency
*/
static inline int range_decode_culfreq(APEContext * ctx, int tot_f)
{
range_dec_normalize(ctx);
ctx->rc.help = ctx->rc.range / tot_f;
return ctx->rc.low / ctx->rc.help;
}
/**
* Decode value with given size in bits
* @param ctx decoder context
* @param shift number of bits to decode
*/
static inline int range_decode_culshift(APEContext * ctx, int shift)
{
range_dec_normalize(ctx);
ctx->rc.help = ctx->rc.range >> shift;
return ctx->rc.low / ctx->rc.help;
}
/**
* Update decoding state
* @param ctx decoder context
* @param sy_f the interval length (frequency of the symbol)
* @param lt_f the lower end (frequency sum of < symbols)
*/
static inline void range_decode_update(APEContext * ctx, int sy_f, int lt_f)
{
ctx->rc.low -= ctx->rc.help * lt_f;
ctx->rc.range = ctx->rc.help * sy_f;
}
/** Decode n bits (n <= 16) without modelling */
static inline int range_decode_bits(APEContext * ctx, int n)
{
int sym = range_decode_culshift(ctx, n);
range_decode_update(ctx, 1, sym);
return sym;
}
#define MODEL_ELEMENTS 64
/**
* Fixed probabilities for symbols in Monkey Audio version 3.97
*/
static const uint16_t counts_3970[22] = {
0, 14824, 28224, 39348, 47855, 53994, 58171, 60926,
62682, 63786, 64463, 64878, 65126, 65276, 65365, 65419,
65450, 65469, 65480, 65487, 65491, 65493,
};
/**
* Probability ranges for symbols in Monkey Audio version 3.97
*/
static const uint16_t counts_diff_3970[21] = {
14824, 13400, 11124, 8507, 6139, 4177, 2755, 1756,
1104, 677, 415, 248, 150, 89, 54, 31,
19, 11, 7, 4, 2,
};
/**
* Fixed probabilities for symbols in Monkey Audio version 3.98
*/
static const uint16_t counts_3980[22] = {
0, 19578, 36160, 48417, 56323, 60899, 63265, 64435,
64971, 65232, 65351, 65416, 65447, 65466, 65476, 65482,
65485, 65488, 65490, 65491, 65492, 65493,
};
/**
* Probability ranges for symbols in Monkey Audio version 3.98
*/
static const uint16_t counts_diff_3980[21] = {
19578, 16582, 12257, 7906, 4576, 2366, 1170, 536,
261, 119, 65, 31, 19, 10, 6, 3,
3, 2, 1, 1, 1,
};
/**
* Decode symbol
* @param ctx decoder context
* @param counts probability range start position
* @param counts_diff probability range widths
*/
static inline int range_get_symbol(APEContext * ctx,
const uint16_t counts[],
const uint16_t counts_diff[])
{
int symbol, cf;
cf = range_decode_culshift(ctx, 16);
if(cf > 65492){
symbol= cf - 65535 + 63;
range_decode_update(ctx, 1, cf);
if(cf > 65535)
ctx->error=1;
return symbol;
}
/* figure out the symbol inefficiently; a binary search would be much better */
for (symbol = 0; counts[symbol + 1] <= cf; symbol++);
range_decode_update(ctx, counts_diff[symbol], counts[symbol]);
return symbol;
}
/** @} */ // group rangecoder
static inline void update_rice(APERice *rice, int x)
{
int lim = rice->k ? (1 << (rice->k + 4)) : 0;
rice->ksum += ((x + 1) / 2) - ((rice->ksum + 16) >> 5);
if (rice->ksum < lim)
rice->k--;
else if (rice->ksum >= (1 << (rice->k + 5)))
rice->k++;
}
static inline int ape_decode_value(APEContext * ctx, APERice *rice)
{
int x, overflow;
if (ctx->fileversion < 3990) {
int tmpk;
overflow = range_get_symbol(ctx, counts_3970, counts_diff_3970);
if (overflow == (MODEL_ELEMENTS - 1)) {
tmpk = range_decode_bits(ctx, 5);
overflow = 0;
} else
tmpk = (rice->k < 1) ? 0 : rice->k - 1;
if (tmpk <= 16)
x = range_decode_bits(ctx, tmpk);
else {
x = range_decode_bits(ctx, 16);
x |= (range_decode_bits(ctx, tmpk - 16) << 16);
}
x += overflow << tmpk;
} else {
int base, pivot;
pivot = rice->ksum >> 5;
if (pivot == 0)
pivot = 1;
overflow = range_get_symbol(ctx, counts_3980, counts_diff_3980);
if (overflow == (MODEL_ELEMENTS - 1)) {
overflow = range_decode_bits(ctx, 16) << 16;
overflow |= range_decode_bits(ctx, 16);
}
if (pivot < 0x10000) {
base = range_decode_culfreq(ctx, pivot);
range_decode_update(ctx, 1, base);
} else {
int base_hi = pivot, base_lo;
int bbits = 0;
while (base_hi & ~0xFFFF) {
base_hi >>= 1;
bbits++;
}
base_hi = range_decode_culfreq(ctx, base_hi + 1);
range_decode_update(ctx, 1, base_hi);
base_lo = range_decode_culfreq(ctx, 1 << bbits);
range_decode_update(ctx, 1, base_lo);
base = (base_hi << bbits) + base_lo;
}
x = base + overflow * pivot;
}
update_rice(rice, x);
/* Convert to signed */
if (x & 1)
return (x >> 1) + 1;
else
return -(x >> 1);
}
static void entropy_decode(APEContext * ctx, int blockstodecode, int stereo)
{
int32_t *decoded0 = ctx->decoded0;
int32_t *decoded1 = ctx->decoded1;
ctx->blocksdecoded = blockstodecode;
if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
/* We are pure silence, just memset the output buffer. */
memset(decoded0, 0, blockstodecode * sizeof(int32_t));
memset(decoded1, 0, blockstodecode * sizeof(int32_t));
} else {
while (blockstodecode--) {
*decoded0++ = ape_decode_value(ctx, &ctx->riceY);
if (stereo)
*decoded1++ = ape_decode_value(ctx, &ctx->riceX);
}
}
if (ctx->blocksdecoded == ctx->currentframeblocks)
range_dec_normalize(ctx); /* normalize to use up all bytes */
}
static void init_entropy_decoder(APEContext * ctx)
{
/* Read the CRC */
ctx->CRC = bytestream_get_be32(&ctx->ptr);
/* Read the frame flags if they exist */
ctx->frameflags = 0;
if ((ctx->fileversion > 3820) && (ctx->CRC & 0x80000000)) {
ctx->CRC &= ~0x80000000;
ctx->frameflags = bytestream_get_be32(&ctx->ptr);
}
/* Keep a count of the blocks decoded in this frame */
ctx->blocksdecoded = 0;
/* Initialize the rice structs */
ctx->riceX.k = 10;
ctx->riceX.ksum = (1 << ctx->riceX.k) * 16;
ctx->riceY.k = 10;
ctx->riceY.ksum = (1 << ctx->riceY.k) * 16;
/* The first 8 bits of input are ignored. */
ctx->ptr++;
range_start_decoding(ctx);
}
static const int32_t initial_coeffs[4] = {
360, 317, -109, 98
};
static void init_predictor_decoder(APEContext * ctx)
{
APEPredictor *p = &ctx->predictor;
/* Zero the history buffers */
memset(p->historybuffer, 0, PREDICTOR_SIZE * sizeof(int32_t));
p->buf = p->historybuffer;
/* Initialize and zero the coefficients */
memcpy(p->coeffsA[0], initial_coeffs, sizeof(initial_coeffs));
memcpy(p->coeffsA[1], initial_coeffs, sizeof(initial_coeffs));
memset(p->coeffsB, 0, sizeof(p->coeffsB));
p->filterA[0] = p->filterA[1] = 0;
p->filterB[0] = p->filterB[1] = 0;
p->lastA[0] = p->lastA[1] = 0;
}
/** Get inverse sign of integer (-1 for positive, 1 for negative and 0 for zero) */
static inline int APESIGN(int32_t x) {
return (x < 0) - (x > 0);
}
static av_always_inline int predictor_update_filter(APEPredictor *p, const int decoded, const int filter, const int delayA, const int delayB, const int adaptA, const int adaptB)
{
int32_t predictionA, predictionB, sign;
p->buf[delayA] = p->lastA[filter];
p->buf[adaptA] = APESIGN(p->buf[delayA]);
p->buf[delayA - 1] = p->buf[delayA] - p->buf[delayA - 1];
p->buf[adaptA - 1] = APESIGN(p->buf[delayA - 1]);
predictionA = p->buf[delayA ] * p->coeffsA[filter][0] +
p->buf[delayA - 1] * p->coeffsA[filter][1] +
p->buf[delayA - 2] * p->coeffsA[filter][2] +
p->buf[delayA - 3] * p->coeffsA[filter][3];
/* Apply a scaled first-order filter compression */
p->buf[delayB] = p->filterA[filter ^ 1] - ((p->filterB[filter] * 31) >> 5);
p->buf[adaptB] = APESIGN(p->buf[delayB]);
p->buf[delayB - 1] = p->buf[delayB] - p->buf[delayB - 1];
p->buf[adaptB - 1] = APESIGN(p->buf[delayB - 1]);
p->filterB[filter] = p->filterA[filter ^ 1];
predictionB = p->buf[delayB ] * p->coeffsB[filter][0] +
p->buf[delayB - 1] * p->coeffsB[filter][1] +
p->buf[delayB - 2] * p->coeffsB[filter][2] +
p->buf[delayB - 3] * p->coeffsB[filter][3] +
p->buf[delayB - 4] * p->coeffsB[filter][4];
p->lastA[filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
p->filterA[filter] = p->lastA[filter] + ((p->filterA[filter] * 31) >> 5);
sign = APESIGN(decoded);
p->coeffsA[filter][0] += p->buf[adaptA ] * sign;
p->coeffsA[filter][1] += p->buf[adaptA - 1] * sign;
p->coeffsA[filter][2] += p->buf[adaptA - 2] * sign;
p->coeffsA[filter][3] += p->buf[adaptA - 3] * sign;
p->coeffsB[filter][0] += p->buf[adaptB ] * sign;
p->coeffsB[filter][1] += p->buf[adaptB - 1] * sign;
p->coeffsB[filter][2] += p->buf[adaptB - 2] * sign;
p->coeffsB[filter][3] += p->buf[adaptB - 3] * sign;
p->coeffsB[filter][4] += p->buf[adaptB - 4] * sign;
return p->filterA[filter];
}
static void predictor_decode_stereo(APEContext * ctx, int count)
{
APEPredictor *p = &ctx->predictor;
int32_t *decoded0 = ctx->decoded0;
int32_t *decoded1 = ctx->decoded1;
while (count--) {
/* Predictor Y */
*decoded0 = predictor_update_filter(p, *decoded0, 0, YDELAYA, YDELAYB, YADAPTCOEFFSA, YADAPTCOEFFSB);
decoded0++;
*decoded1 = predictor_update_filter(p, *decoded1, 1, XDELAYA, XDELAYB, XADAPTCOEFFSA, XADAPTCOEFFSB);
decoded1++;
/* Combined */
p->buf++;
/* Have we filled the history buffer? */
if (p->buf == p->historybuffer + HISTORY_SIZE) {
memmove(p->historybuffer, p->buf, PREDICTOR_SIZE * sizeof(int32_t));
p->buf = p->historybuffer;
}
}
}
static void predictor_decode_mono(APEContext * ctx, int count)
{
APEPredictor *p = &ctx->predictor;
int32_t *decoded0 = ctx->decoded0;
int32_t predictionA, currentA, A, sign;
currentA = p->lastA[0];
while (count--) {
A = *decoded0;
p->buf[YDELAYA] = currentA;
p->buf[YDELAYA - 1] = p->buf[YDELAYA] - p->buf[YDELAYA - 1];
predictionA = p->buf[YDELAYA ] * p->coeffsA[0][0] +
p->buf[YDELAYA - 1] * p->coeffsA[0][1] +
p->buf[YDELAYA - 2] * p->coeffsA[0][2] +
p->buf[YDELAYA - 3] * p->coeffsA[0][3];
currentA = A + (predictionA >> 10);
p->buf[YADAPTCOEFFSA] = APESIGN(p->buf[YDELAYA ]);
p->buf[YADAPTCOEFFSA - 1] = APESIGN(p->buf[YDELAYA - 1]);
sign = APESIGN(A);
p->coeffsA[0][0] += p->buf[YADAPTCOEFFSA ] * sign;
p->coeffsA[0][1] += p->buf[YADAPTCOEFFSA - 1] * sign;
p->coeffsA[0][2] += p->buf[YADAPTCOEFFSA - 2] * sign;
p->coeffsA[0][3] += p->buf[YADAPTCOEFFSA - 3] * sign;
p->buf++;
/* Have we filled the history buffer? */
if (p->buf == p->historybuffer + HISTORY_SIZE) {
memmove(p->historybuffer, p->buf, PREDICTOR_SIZE * sizeof(int32_t));
p->buf = p->historybuffer;
}
p->filterA[0] = currentA + ((p->filterA[0] * 31) >> 5);
*(decoded0++) = p->filterA[0];
}
p->lastA[0] = currentA;
}
static void do_init_filter(APEFilter *f, int16_t * buf, int order)
{
f->coeffs = buf;
f->historybuffer = buf + order;
f->delay = f->historybuffer + order * 2;
f->adaptcoeffs = f->historybuffer + order;
memset(f->historybuffer, 0, (order * 2) * sizeof(int16_t));
memset(f->coeffs, 0, order * sizeof(int16_t));
f->avg = 0;
}
static void init_filter(APEContext * ctx, APEFilter *f, int16_t * buf, int order)
{
do_init_filter(&f[0], buf, order);
do_init_filter(&f[1], buf + order * 3 + HISTORY_SIZE, order);
}
static void do_apply_filter(APEContext * ctx, int version, APEFilter *f, int32_t *data, int count, int order, int fracbits)
{
int res;
int absres;
while (count--) {
/* round fixedpoint scalar product */
res = ctx->dsp.scalarproduct_and_madd_int16(f->coeffs, f->delay - order, f->adaptcoeffs - order, order, APESIGN(*data));
res = (res + (1 << (fracbits - 1))) >> fracbits;
res += *data;
*data++ = res;
/* Update the output history */
*f->delay++ = av_clip_int16(res);
if (version < 3980) {
/* Version ??? to < 3.98 files (untested) */
f->adaptcoeffs[0] = (res == 0) ? 0 : ((res >> 28) & 8) - 4;
f->adaptcoeffs[-4] >>= 1;
f->adaptcoeffs[-8] >>= 1;
} else {
/* Version 3.98 and later files */
/* Update the adaption coefficients */
absres = FFABS(res);
if (absres)
*f->adaptcoeffs = ((res & (1<<31)) - (1<<30)) >> (25 + (absres <= f->avg*3) + (absres <= f->avg*4/3));
else
*f->adaptcoeffs = 0;
f->avg += (absres - f->avg) / 16;
f->adaptcoeffs[-1] >>= 1;
f->adaptcoeffs[-2] >>= 1;
f->adaptcoeffs[-8] >>= 1;
}
f->adaptcoeffs++;
/* Have we filled the history buffer? */
if (f->delay == f->historybuffer + HISTORY_SIZE + (order * 2)) {
memmove(f->historybuffer, f->delay - (order * 2),
(order * 2) * sizeof(int16_t));
f->delay = f->historybuffer + order * 2;
f->adaptcoeffs = f->historybuffer + order;
}
}
}
static void apply_filter(APEContext * ctx, APEFilter *f,
int32_t * data0, int32_t * data1,
int count, int order, int fracbits)
{
do_apply_filter(ctx, ctx->fileversion, &f[0], data0, count, order, fracbits);
if (data1)
do_apply_filter(ctx, ctx->fileversion, &f[1], data1, count, order, fracbits);
}
static void ape_apply_filters(APEContext * ctx, int32_t * decoded0,
int32_t * decoded1, int count)
{
int i;
for (i = 0; i < APE_FILTER_LEVELS; i++) {
if (!ape_filter_orders[ctx->fset][i])
break;
apply_filter(ctx, ctx->filters[i], decoded0, decoded1, count, ape_filter_orders[ctx->fset][i], ape_filter_fracbits[ctx->fset][i]);
}
}
static void init_frame_decoder(APEContext * ctx)
{
int i;
init_entropy_decoder(ctx);
init_predictor_decoder(ctx);
for (i = 0; i < APE_FILTER_LEVELS; i++) {
if (!ape_filter_orders[ctx->fset][i])
break;
init_filter(ctx, ctx->filters[i], ctx->filterbuf[i], ape_filter_orders[ctx->fset][i]);
}
}
static void ape_unpack_mono(APEContext * ctx, int count)
{
int32_t left;
int32_t *decoded0 = ctx->decoded0;
int32_t *decoded1 = ctx->decoded1;
if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
entropy_decode(ctx, count, 0);
/* We are pure silence, so we're done. */
av_log(ctx->avctx, AV_LOG_DEBUG, "pure silence mono\n");
return;
}
entropy_decode(ctx, count, 0);
ape_apply_filters(ctx, decoded0, NULL, count);
/* Now apply the predictor decoding */
predictor_decode_mono(ctx, count);
/* Pseudo-stereo - just copy left channel to right channel */
if (ctx->channels == 2) {
while (count--) {
left = *decoded0;
*(decoded1++) = *(decoded0++) = left;
}
}
}
static void ape_unpack_stereo(APEContext * ctx, int count)
{
int32_t left, right;
int32_t *decoded0 = ctx->decoded0;
int32_t *decoded1 = ctx->decoded1;
if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
/* We are pure silence, so we're done. */
av_log(ctx->avctx, AV_LOG_DEBUG, "pure silence stereo\n");
return;
}
entropy_decode(ctx, count, 1);
ape_apply_filters(ctx, decoded0, decoded1, count);
/* Now apply the predictor decoding */
predictor_decode_stereo(ctx, count);
/* Decorrelate and scale to output depth */
while (count--) {
left = *decoded1 - (*decoded0 / 2);
right = left + *decoded0;
*(decoded0++) = left;
*(decoded1++) = right;
}
}
static int ape_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
APEContext *s = avctx->priv_data;
int16_t *samples = data;
int nblocks;
int i, n;
int blockstodecode;
int bytes_used;
if (buf_size == 0 && !s->samples) {
*data_size = 0;
return 0;
}
/* should not happen but who knows */
if (BLOCKS_PER_LOOP * 2 * avctx->channels > *data_size) {
av_log (avctx, AV_LOG_ERROR, "Packet size is too big to be handled in lavc! (max is %d where you have %d)\n", *data_size, s->samples * 2 * avctx->channels);
return -1;
}
if(!s->samples){
s->data = av_realloc(s->data, (buf_size + 3) & ~3);
s->dsp.bswap_buf((uint32_t*)s->data, (const uint32_t*)buf, buf_size >> 2);
s->ptr = s->last_ptr = s->data;
s->data_end = s->data + buf_size;
nblocks = s->samples = bytestream_get_be32(&s->ptr);
n = bytestream_get_be32(&s->ptr);
if(n < 0 || n > 3){
av_log(avctx, AV_LOG_ERROR, "Incorrect offset passed\n");
s->data = NULL;
return -1;
}
s->ptr += n;
s->currentframeblocks = nblocks;
buf += 4;
if (s->samples <= 0) {
*data_size = 0;
return buf_size;
}
memset(s->decoded0, 0, sizeof(s->decoded0));
memset(s->decoded1, 0, sizeof(s->decoded1));
/* Initialize the frame decoder */
init_frame_decoder(s);
}
if (!s->data) {
*data_size = 0;
return buf_size;
}
nblocks = s->samples;
blockstodecode = FFMIN(BLOCKS_PER_LOOP, nblocks);
s->error=0;
if ((s->channels == 1) || (s->frameflags & APE_FRAMECODE_PSEUDO_STEREO))
ape_unpack_mono(s, blockstodecode);
else
ape_unpack_stereo(s, blockstodecode);
emms_c();
if(s->error || s->ptr > s->data_end){
s->samples=0;
av_log(avctx, AV_LOG_ERROR, "Error decoding frame\n");
return -1;
}
for (i = 0; i < blockstodecode; i++) {
*samples++ = s->decoded0[i];
if(s->channels == 2)
*samples++ = s->decoded1[i];
}
s->samples -= blockstodecode;
*data_size = blockstodecode * 2 * s->channels;
bytes_used = s->samples ? s->ptr - s->last_ptr : buf_size;
s->last_ptr = s->ptr;
return bytes_used;
}
AVCodec ape_decoder = {
"ape",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_APE,
sizeof(APEContext),
ape_decode_init,
NULL,
ape_decode_close,
ape_decode_frame,
.capabilities = CODEC_CAP_SUBFRAMES,
.long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/api-example.c
-467
View File
@@ -1,467 +0,0 @@
/*
* copyright (c) 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* avcodec API use example.
*
* Note that this library only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, etc...). See library 'libavformat' for the
* format handling
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#ifdef HAVE_AV_CONFIG_H
#undef HAVE_AV_CONFIG_H
#endif
#include "libavcodec/avcodec.h"
#include "libavutil/mathematics.h"
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_size, i, j, out_size, outbuf_size;
FILE *f;
short *samples;
float t, tincr;
uint8_t *outbuf;
printf("Audio encoding\n");
/* find the MP2 encoder */
codec = avcodec_find_encoder(CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c= avcodec_alloc_context();
/* put sample parameters */
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
/* the codec gives us the frame size, in samples */
frame_size = c->frame_size;
samples = malloc(frame_size * 2 * c->channels);
outbuf_size = 10000;
outbuf = malloc(outbuf_size);
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++) {
for(j=0;j<frame_size;j++) {
samples[2*j] = (int)(sin(t) * 10000);
samples[2*j+1] = samples[2*j];
t += tincr;
}
/* encode the samples */
out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
fwrite(outbuf, 1, out_size, f);
}
fclose(f);
free(outbuf);
free(samples);
avcodec_close(c);
av_free(c);
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int out_size, len;
FILE *f, *outfile;
uint8_t *outbuf;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
av_init_packet(&avpkt);
printf("Audio decoding\n");
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c= avcodec_alloc_context();
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
len = avcodec_decode_audio3(c, (short *)outbuf, &out_size, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (out_size > 0) {
/* if a frame has been decoded, output it */
fwrite(outbuf, 1, out_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
free(outbuf);
avcodec_close(c);
av_free(c);
}
/*
* Video encoding example
*/
static void video_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int i, out_size, size, x, y, outbuf_size;
FILE *f;
AVFrame *picture;
uint8_t *outbuf, *picture_buf;
printf("Video encoding\n");
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c= avcodec_alloc_context();
picture= avcodec_alloc_frame();
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base= (AVRational){1,25};
c->gop_size = 10; /* emit one intra frame every ten frames */
c->max_b_frames=1;
c->pix_fmt = PIX_FMT_YUV420P;
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* alloc image and output buffer */
outbuf_size = 100000;
outbuf = malloc(outbuf_size);
size = c->width * c->height;
picture_buf = malloc((size * 3) / 2); /* size for YUV 420 */
picture->data[0] = picture_buf;
picture->data[1] = picture->data[0] + size;
picture->data[2] = picture->data[1] + size / 4;
picture->linesize[0] = c->width;
picture->linesize[1] = c->width / 2;
picture->linesize[2] = c->width / 2;
/* encode 1 second of video */
for(i=0;i<25;i++) {
fflush(stdout);
/* prepare a dummy image */
/* Y */
for(y=0;y<c->height;y++) {
for(x=0;x<c->width;x++) {
picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for(y=0;y<c->height/2;y++) {
for(x=0;x<c->width/2;x++) {
picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
}
}
/* encode the image */
out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
printf("encoding frame %3d (size=%5d)\n", i, out_size);
fwrite(outbuf, 1, out_size, f);
}
/* get the delayed frames */
for(; out_size; i++) {
fflush(stdout);
out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
printf("write frame %3d (size=%5d)\n", i, out_size);
fwrite(outbuf, 1, out_size, f);
}
/* add sequence end code to have a real mpeg file */
outbuf[0] = 0x00;
outbuf[1] = 0x00;
outbuf[2] = 0x01;
outbuf[3] = 0xb7;
fwrite(outbuf, 1, 4, f);
fclose(f);
free(picture_buf);
free(outbuf);
avcodec_close(c);
av_free(c);
av_free(picture);
printf("\n");
}
/*
* Video decoding example
*/
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f=fopen(filename,"w");
fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
for(i=0;i<ysize;i++)
fwrite(buf + i * wrap,1,xsize,f);
fclose(f);
}
static void video_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame, got_picture, len;
FILE *f;
AVFrame *picture;
uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
char buf[1024];
AVPacket avpkt;
av_init_packet(&avpkt);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
printf("Video decoding\n");
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c= avcodec_alloc_context();
picture= avcodec_alloc_frame();
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
/* the codec gives us the frame size, in samples */
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
frame = 0;
for(;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
break;
/* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
and this is the only method to use them because you cannot
know the compressed data size before analysing it.
BUT some other codecs (msmpeg4, mpeg4) are inherently frame
based, so you must call them with all the data for one
frame exactly. You must also initialize 'width' and
'height' before initializing them. */
/* NOTE2: some codecs allow the raw parameters (frame size,
sample rate) to be changed at any frame. We handle this, so
you should also take care of it */
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
avpkt.data = inbuf;
while (avpkt.size > 0) {
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", frame);
exit(1);
}
if (got_picture) {
printf("saving frame %3d\n", frame);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), outfilename, frame);
pgm_save(picture->data[0], picture->linesize[0],
c->width, c->height, buf);
frame++;
}
avpkt.size -= len;
avpkt.data += len;
}
}
/* some codecs, such as MPEG, transmit the I and P frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (got_picture) {
printf("saving last frame %3d\n", frame);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), outfilename, frame);
pgm_save(picture->data[0], picture->linesize[0],
c->width, c->height, buf);
frame++;
}
fclose(f);
avcodec_close(c);
av_free(c);
av_free(picture);
printf("\n");
}
int main(int argc, char **argv)
{
const char *filename;
/* must be called before using avcodec lib */
avcodec_init();
/* register all the codecs */
avcodec_register_all();
if (argc <= 1) {
audio_encode_example("/tmp/test.mp2");
audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
video_encode_example("/tmp/test.mpg");
filename = "/tmp/test.mpg";
} else {
filename = argv[1];
}
// audio_decode_example("/tmp/test.sw", filename);
video_decode_example("/tmp/test%d.pgm", filename);
return 0;
}
src/add-ons/media/plugins/ffmpeg/libavcodec/asv1.c
-668
View File
@@ -1,668 +0,0 @@
/*
* ASUS V1/V2 codec
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ASUS V1/V2 codec.
*/
#include "avcodec.h"
#include "libavutil/common.h"
#include "put_bits.h"
#include "dsputil.h"
#include "mpeg12data.h"
//#undef NDEBUG
//#include <assert.h>
#define VLC_BITS 6
#define ASV2_LEVEL_VLC_BITS 10
typedef struct ASV1Context{
AVCodecContext *avctx;
DSPContext dsp;
AVFrame picture;
PutBitContext pb;
GetBitContext gb;
ScanTable scantable;
int inv_qscale;
int mb_width;
int mb_height;
int mb_width2;
int mb_height2;
DECLARE_ALIGNED(16, DCTELEM, block)[6][64];
uint16_t intra_matrix[64];
int q_intra_matrix[64];
uint8_t *bitstream_buffer;
unsigned int bitstream_buffer_size;
} ASV1Context;
static const uint8_t scantab[64]={
0x00,0x08,0x01,0x09,0x10,0x18,0x11,0x19,
0x02,0x0A,0x03,0x0B,0x12,0x1A,0x13,0x1B,
0x04,0x0C,0x05,0x0D,0x20,0x28,0x21,0x29,
0x06,0x0E,0x07,0x0F,0x14,0x1C,0x15,0x1D,
0x22,0x2A,0x23,0x2B,0x30,0x38,0x31,0x39,
0x16,0x1E,0x17,0x1F,0x24,0x2C,0x25,0x2D,
0x32,0x3A,0x33,0x3B,0x26,0x2E,0x27,0x2F,
0x34,0x3C,0x35,0x3D,0x36,0x3E,0x37,0x3F,
};
static const uint8_t ccp_tab[17][2]={
{0x2,2}, {0x7,5}, {0xB,5}, {0x3,5},
{0xD,5}, {0x5,5}, {0x9,5}, {0x1,5},
{0xE,5}, {0x6,5}, {0xA,5}, {0x2,5},
{0xC,5}, {0x4,5}, {0x8,5}, {0x3,2},
{0xF,5}, //EOB
};
static const uint8_t level_tab[7][2]={
{3,4}, {3,3}, {3,2}, {0,3}, {2,2}, {2,3}, {2,4}
};
static const uint8_t dc_ccp_tab[8][2]={
{0x1,2}, {0xD,4}, {0xF,4}, {0xC,4},
{0x5,3}, {0xE,4}, {0x4,3}, {0x0,2},
};
static const uint8_t ac_ccp_tab[16][2]={
{0x00,2}, {0x3B,6}, {0x0A,4}, {0x3A,6},
{0x02,3}, {0x39,6}, {0x3C,6}, {0x38,6},
{0x03,3}, {0x3D,6}, {0x08,4}, {0x1F,5},
{0x09,4}, {0x0B,4}, {0x0D,4}, {0x0C,4},
};
static const uint8_t asv2_level_tab[63][2]={
{0x3F,10},{0x2F,10},{0x37,10},{0x27,10},{0x3B,10},{0x2B,10},{0x33,10},{0x23,10},
{0x3D,10},{0x2D,10},{0x35,10},{0x25,10},{0x39,10},{0x29,10},{0x31,10},{0x21,10},
{0x1F, 8},{0x17, 8},{0x1B, 8},{0x13, 8},{0x1D, 8},{0x15, 8},{0x19, 8},{0x11, 8},
{0x0F, 6},{0x0B, 6},{0x0D, 6},{0x09, 6},
{0x07, 4},{0x05, 4},
{0x03, 2},
{0x00, 5},
{0x02, 2},
{0x04, 4},{0x06, 4},
{0x08, 6},{0x0C, 6},{0x0A, 6},{0x0E, 6},
{0x10, 8},{0x18, 8},{0x14, 8},{0x1C, 8},{0x12, 8},{0x1A, 8},{0x16, 8},{0x1E, 8},
{0x20,10},{0x30,10},{0x28,10},{0x38,10},{0x24,10},{0x34,10},{0x2C,10},{0x3C,10},
{0x22,10},{0x32,10},{0x2A,10},{0x3A,10},{0x26,10},{0x36,10},{0x2E,10},{0x3E,10},
};
static VLC ccp_vlc;
static VLC level_vlc;
static VLC dc_ccp_vlc;
static VLC ac_ccp_vlc;
static VLC asv2_level_vlc;
static av_cold void init_vlcs(ASV1Context *a){
static int done = 0;
if (!done) {
done = 1;
INIT_VLC_STATIC(&ccp_vlc, VLC_BITS, 17,
&ccp_tab[0][1], 2, 1,
&ccp_tab[0][0], 2, 1, 64);
INIT_VLC_STATIC(&dc_ccp_vlc, VLC_BITS, 8,
&dc_ccp_tab[0][1], 2, 1,
&dc_ccp_tab[0][0], 2, 1, 64);
INIT_VLC_STATIC(&ac_ccp_vlc, VLC_BITS, 16,
&ac_ccp_tab[0][1], 2, 1,
&ac_ccp_tab[0][0], 2, 1, 64);
INIT_VLC_STATIC(&level_vlc, VLC_BITS, 7,
&level_tab[0][1], 2, 1,
&level_tab[0][0], 2, 1, 64);
INIT_VLC_STATIC(&asv2_level_vlc, ASV2_LEVEL_VLC_BITS, 63,
&asv2_level_tab[0][1], 2, 1,
&asv2_level_tab[0][0], 2, 1, 1024);
}
}
//FIXME write a reversed bitstream reader to avoid the double reverse
static inline int asv2_get_bits(GetBitContext *gb, int n){
return av_reverse[ get_bits(gb, n) << (8-n) ];
}
static inline void asv2_put_bits(PutBitContext *pb, int n, int v){
put_bits(pb, n, av_reverse[ v << (8-n) ]);
}
static inline int asv1_get_level(GetBitContext *gb){
int code= get_vlc2(gb, level_vlc.table, VLC_BITS, 1);
if(code==3) return get_sbits(gb, 8);
else return code - 3;
}
static inline int asv2_get_level(GetBitContext *gb){
int code= get_vlc2(gb, asv2_level_vlc.table, ASV2_LEVEL_VLC_BITS, 1);
if(code==31) return (int8_t)asv2_get_bits(gb, 8);
else return code - 31;
}
static inline void asv1_put_level(PutBitContext *pb, int level){
unsigned int index= level + 3;
if(index <= 6) put_bits(pb, level_tab[index][1], level_tab[index][0]);
else{
put_bits(pb, level_tab[3][1], level_tab[3][0]);
put_sbits(pb, 8, level);
}
}
static inline void asv2_put_level(PutBitContext *pb, int level){
unsigned int index= level + 31;
if(index <= 62) put_bits(pb, asv2_level_tab[index][1], asv2_level_tab[index][0]);
else{
put_bits(pb, asv2_level_tab[31][1], asv2_level_tab[31][0]);
asv2_put_bits(pb, 8, level&0xFF);
}
}
static inline int asv1_decode_block(ASV1Context *a, DCTELEM block[64]){
int i;
block[0]= 8*get_bits(&a->gb, 8);
for(i=0; i<11; i++){
const int ccp= get_vlc2(&a->gb, ccp_vlc.table, VLC_BITS, 1);
if(ccp){
if(ccp == 16) break;
if(ccp < 0 || i>=10){
av_log(a->avctx, AV_LOG_ERROR, "coded coeff pattern damaged\n");
return -1;
}
if(ccp&8) block[a->scantable.permutated[4*i+0]]= (asv1_get_level(&a->gb) * a->intra_matrix[4*i+0])>>4;
if(ccp&4) block[a->scantable.permutated[4*i+1]]= (asv1_get_level(&a->gb) * a->intra_matrix[4*i+1])>>4;
if(ccp&2) block[a->scantable.permutated[4*i+2]]= (asv1_get_level(&a->gb) * a->intra_matrix[4*i+2])>>4;
if(ccp&1) block[a->scantable.permutated[4*i+3]]= (asv1_get_level(&a->gb) * a->intra_matrix[4*i+3])>>4;
}
}
return 0;
}
static inline int asv2_decode_block(ASV1Context *a, DCTELEM block[64]){
int i, count, ccp;
count= asv2_get_bits(&a->gb, 4);
block[0]= 8*asv2_get_bits(&a->gb, 8);
ccp= get_vlc2(&a->gb, dc_ccp_vlc.table, VLC_BITS, 1);
if(ccp){
if(ccp&4) block[a->scantable.permutated[1]]= (asv2_get_level(&a->gb) * a->intra_matrix[1])>>4;
if(ccp&2) block[a->scantable.permutated[2]]= (asv2_get_level(&a->gb) * a->intra_matrix[2])>>4;
if(ccp&1) block[a->scantable.permutated[3]]= (asv2_get_level(&a->gb) * a->intra_matrix[3])>>4;
}
for(i=1; i<count+1; i++){
const int ccp= get_vlc2(&a->gb, ac_ccp_vlc.table, VLC_BITS, 1);
if(ccp){
if(ccp&8) block[a->scantable.permutated[4*i+0]]= (asv2_get_level(&a->gb) * a->intra_matrix[4*i+0])>>4;
if(ccp&4) block[a->scantable.permutated[4*i+1]]= (asv2_get_level(&a->gb) * a->intra_matrix[4*i+1])>>4;
if(ccp&2) block[a->scantable.permutated[4*i+2]]= (asv2_get_level(&a->gb) * a->intra_matrix[4*i+2])>>4;
if(ccp&1) block[a->scantable.permutated[4*i+3]]= (asv2_get_level(&a->gb) * a->intra_matrix[4*i+3])>>4;
}
}
return 0;
}
static inline void asv1_encode_block(ASV1Context *a, DCTELEM block[64]){
int i;
int nc_count=0;
put_bits(&a->pb, 8, (block[0] + 32)>>6);
block[0]= 0;
for(i=0; i<10; i++){
const int index= scantab[4*i];
int ccp=0;
if( (block[index + 0] = (block[index + 0]*a->q_intra_matrix[index + 0] + (1<<15))>>16) ) ccp |= 8;
if( (block[index + 8] = (block[index + 8]*a->q_intra_matrix[index + 8] + (1<<15))>>16) ) ccp |= 4;
if( (block[index + 1] = (block[index + 1]*a->q_intra_matrix[index + 1] + (1<<15))>>16) ) ccp |= 2;
if( (block[index + 9] = (block[index + 9]*a->q_intra_matrix[index + 9] + (1<<15))>>16) ) ccp |= 1;
if(ccp){
for(;nc_count; nc_count--)
put_bits(&a->pb, ccp_tab[0][1], ccp_tab[0][0]);
put_bits(&a->pb, ccp_tab[ccp][1], ccp_tab[ccp][0]);
if(ccp&8) asv1_put_level(&a->pb, block[index + 0]);
if(ccp&4) asv1_put_level(&a->pb, block[index + 8]);
if(ccp&2) asv1_put_level(&a->pb, block[index + 1]);
if(ccp&1) asv1_put_level(&a->pb, block[index + 9]);
}else{
nc_count++;
}
}
put_bits(&a->pb, ccp_tab[16][1], ccp_tab[16][0]);
}
static inline void asv2_encode_block(ASV1Context *a, DCTELEM block[64]){
int i;
int count=0;
for(count=63; count>3; count--){
const int index= scantab[count];
if( (block[index]*a->q_intra_matrix[index] + (1<<15))>>16 )
break;
}
count >>= 2;
asv2_put_bits(&a->pb, 4, count);
asv2_put_bits(&a->pb, 8, (block[0] + 32)>>6);
block[0]= 0;
for(i=0; i<=count; i++){
const int index= scantab[4*i];
int ccp=0;
if( (block[index + 0] = (block[index + 0]*a->q_intra_matrix[index + 0] + (1<<15))>>16) ) ccp |= 8;
if( (block[index + 8] = (block[index + 8]*a->q_intra_matrix[index + 8] + (1<<15))>>16) ) ccp |= 4;
if( (block[index + 1] = (block[index + 1]*a->q_intra_matrix[index + 1] + (1<<15))>>16) ) ccp |= 2;
if( (block[index + 9] = (block[index + 9]*a->q_intra_matrix[index + 9] + (1<<15))>>16) ) ccp |= 1;
assert(i || ccp<8);
if(i) put_bits(&a->pb, ac_ccp_tab[ccp][1], ac_ccp_tab[ccp][0]);
else put_bits(&a->pb, dc_ccp_tab[ccp][1], dc_ccp_tab[ccp][0]);
if(ccp){
if(ccp&8) asv2_put_level(&a->pb, block[index + 0]);
if(ccp&4) asv2_put_level(&a->pb, block[index + 8]);
if(ccp&2) asv2_put_level(&a->pb, block[index + 1]);
if(ccp&1) asv2_put_level(&a->pb, block[index + 9]);
}
}
}
static inline int decode_mb(ASV1Context *a, DCTELEM block[6][64]){
int i;
a->dsp.clear_blocks(block[0]);
if(a->avctx->codec_id == CODEC_ID_ASV1){
for(i=0; i<6; i++){
if( asv1_decode_block(a, block[i]) < 0)
return -1;
}
}else{
for(i=0; i<6; i++){
if( asv2_decode_block(a, block[i]) < 0)
return -1;
}
}
return 0;
}
static inline int encode_mb(ASV1Context *a, DCTELEM block[6][64]){
int i;
if(a->pb.buf_end - a->pb.buf - (put_bits_count(&a->pb)>>3) < 30*16*16*3/2/8){
av_log(a->avctx, AV_LOG_ERROR, "encoded frame too large\n");
return -1;
}
if(a->avctx->codec_id == CODEC_ID_ASV1){
for(i=0; i<6; i++)
asv1_encode_block(a, block[i]);
}else{
for(i=0; i<6; i++)
asv2_encode_block(a, block[i]);
}
return 0;
}
static inline void idct_put(ASV1Context *a, int mb_x, int mb_y){
DCTELEM (*block)[64]= a->block;
int linesize= a->picture.linesize[0];
uint8_t *dest_y = a->picture.data[0] + (mb_y * 16* linesize ) + mb_x * 16;
uint8_t *dest_cb = a->picture.data[1] + (mb_y * 8 * a->picture.linesize[1]) + mb_x * 8;
uint8_t *dest_cr = a->picture.data[2] + (mb_y * 8 * a->picture.linesize[2]) + mb_x * 8;
a->dsp.idct_put(dest_y , linesize, block[0]);
a->dsp.idct_put(dest_y + 8, linesize, block[1]);
a->dsp.idct_put(dest_y + 8*linesize , linesize, block[2]);
a->dsp.idct_put(dest_y + 8*linesize + 8, linesize, block[3]);
if(!(a->avctx->flags&CODEC_FLAG_GRAY)){
a->dsp.idct_put(dest_cb, a->picture.linesize[1], block[4]);
a->dsp.idct_put(dest_cr, a->picture.linesize[2], block[5]);
}
}
static inline void dct_get(ASV1Context *a, int mb_x, int mb_y){
DCTELEM (*block)[64]= a->block;
int linesize= a->picture.linesize[0];
int i;
uint8_t *ptr_y = a->picture.data[0] + (mb_y * 16* linesize ) + mb_x * 16;
uint8_t *ptr_cb = a->picture.data[1] + (mb_y * 8 * a->picture.linesize[1]) + mb_x * 8;
uint8_t *ptr_cr = a->picture.data[2] + (mb_y * 8 * a->picture.linesize[2]) + mb_x * 8;
a->dsp.get_pixels(block[0], ptr_y , linesize);
a->dsp.get_pixels(block[1], ptr_y + 8, linesize);
a->dsp.get_pixels(block[2], ptr_y + 8*linesize , linesize);
a->dsp.get_pixels(block[3], ptr_y + 8*linesize + 8, linesize);
for(i=0; i<4; i++)
a->dsp.fdct(block[i]);
if(!(a->avctx->flags&CODEC_FLAG_GRAY)){
a->dsp.get_pixels(block[4], ptr_cb, a->picture.linesize[1]);
a->dsp.get_pixels(block[5], ptr_cr, a->picture.linesize[2]);
for(i=4; i<6; i++)
a->dsp.fdct(block[i]);
}
}
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ASV1Context * const a = avctx->priv_data;
AVFrame *picture = data;
AVFrame * const p= &a->picture;
int mb_x, mb_y;
if(p->data[0])
avctx->release_buffer(avctx, p);
p->reference= 0;
if(avctx->get_buffer(avctx, p) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
p->pict_type= FF_I_TYPE;
p->key_frame= 1;
av_fast_malloc(&a->bitstream_buffer, &a->bitstream_buffer_size, buf_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!a->bitstream_buffer)
return AVERROR(ENOMEM);
if(avctx->codec_id == CODEC_ID_ASV1)
a->dsp.bswap_buf((uint32_t*)a->bitstream_buffer, (const uint32_t*)buf, buf_size/4);
else{
int i;
for(i=0; i<buf_size; i++)
a->bitstream_buffer[i]= av_reverse[ buf[i] ];
}
init_get_bits(&a->gb, a->bitstream_buffer, buf_size*8);
for(mb_y=0; mb_y<a->mb_height2; mb_y++){
for(mb_x=0; mb_x<a->mb_width2; mb_x++){
if( decode_mb(a, a->block) <0)
return -1;
idct_put(a, mb_x, mb_y);
}
}
if(a->mb_width2 != a->mb_width){
mb_x= a->mb_width2;
for(mb_y=0; mb_y<a->mb_height2; mb_y++){
if( decode_mb(a, a->block) <0)
return -1;
idct_put(a, mb_x, mb_y);
}
}
if(a->mb_height2 != a->mb_height){
mb_y= a->mb_height2;
for(mb_x=0; mb_x<a->mb_width; mb_x++){
if( decode_mb(a, a->block) <0)
return -1;
idct_put(a, mb_x, mb_y);
}
}
#if 0
int i;
printf("%d %d\n", 8*buf_size, get_bits_count(&a->gb));
for(i=get_bits_count(&a->gb); i<8*buf_size; i++){
printf("%d", get_bits1(&a->gb));
}
for(i=0; i<s->avctx->extradata_size; i++){
printf("%c\n", ((uint8_t*)s->avctx->extradata)[i]);
}
#endif
*picture= *(AVFrame*)&a->picture;
*data_size = sizeof(AVPicture);
emms_c();
return (get_bits_count(&a->gb)+31)/32*4;
}
#if CONFIG_ASV1_ENCODER || CONFIG_ASV2_ENCODER
static int encode_frame(AVCodecContext *avctx, unsigned char *buf, int buf_size, void *data){
ASV1Context * const a = avctx->priv_data;
AVFrame *pict = data;
AVFrame * const p= &a->picture;
int size;
int mb_x, mb_y;
init_put_bits(&a->pb, buf, buf_size);
*p = *pict;
p->pict_type= FF_I_TYPE;
p->key_frame= 1;
for(mb_y=0; mb_y<a->mb_height2; mb_y++){
for(mb_x=0; mb_x<a->mb_width2; mb_x++){
dct_get(a, mb_x, mb_y);
encode_mb(a, a->block);
}
}
if(a->mb_width2 != a->mb_width){
mb_x= a->mb_width2;
for(mb_y=0; mb_y<a->mb_height2; mb_y++){
dct_get(a, mb_x, mb_y);
encode_mb(a, a->block);
}
}
if(a->mb_height2 != a->mb_height){
mb_y= a->mb_height2;
for(mb_x=0; mb_x<a->mb_width; mb_x++){
dct_get(a, mb_x, mb_y);
encode_mb(a, a->block);
}
}
emms_c();
align_put_bits(&a->pb);
while(put_bits_count(&a->pb)&31)
put_bits(&a->pb, 8, 0);
size= put_bits_count(&a->pb)/32;
if(avctx->codec_id == CODEC_ID_ASV1)
a->dsp.bswap_buf((uint32_t*)buf, (uint32_t*)buf, size);
else{
int i;
for(i=0; i<4*size; i++)
buf[i]= av_reverse[ buf[i] ];
}
return size*4;
}
#endif /* CONFIG_ASV1_ENCODER || CONFIG_ASV2_ENCODER */
static av_cold void common_init(AVCodecContext *avctx){
ASV1Context * const a = avctx->priv_data;
dsputil_init(&a->dsp, avctx);
a->mb_width = (avctx->width + 15) / 16;
a->mb_height = (avctx->height + 15) / 16;
a->mb_width2 = (avctx->width + 0) / 16;
a->mb_height2 = (avctx->height + 0) / 16;
avctx->coded_frame= &a->picture;
a->avctx= avctx;
}
static av_cold int decode_init(AVCodecContext *avctx){
ASV1Context * const a = avctx->priv_data;
AVFrame *p= &a->picture;
int i;
const int scale= avctx->codec_id == CODEC_ID_ASV1 ? 1 : 2;
common_init(avctx);
init_vlcs(a);
ff_init_scantable(a->dsp.idct_permutation, &a->scantable, scantab);
avctx->pix_fmt= PIX_FMT_YUV420P;
a->inv_qscale= avctx->extradata[0];
if(a->inv_qscale == 0){
av_log(avctx, AV_LOG_ERROR, "illegal qscale 0\n");
if(avctx->codec_id == CODEC_ID_ASV1)
a->inv_qscale= 6;
else
a->inv_qscale= 10;
}
for(i=0; i<64; i++){
int index= scantab[i];
a->intra_matrix[i]= 64*scale*ff_mpeg1_default_intra_matrix[index] / a->inv_qscale;
}
p->qstride= a->mb_width;
p->qscale_table= av_malloc( p->qstride * a->mb_height);
p->quality= (32*scale + a->inv_qscale/2)/a->inv_qscale;
memset(p->qscale_table, p->quality, p->qstride*a->mb_height);
return 0;
}
#if CONFIG_ASV1_ENCODER || CONFIG_ASV2_ENCODER
static av_cold int encode_init(AVCodecContext *avctx){
ASV1Context * const a = avctx->priv_data;
int i;
const int scale= avctx->codec_id == CODEC_ID_ASV1 ? 1 : 2;
common_init(avctx);
if(avctx->global_quality == 0) avctx->global_quality= 4*FF_QUALITY_SCALE;
a->inv_qscale= (32*scale*FF_QUALITY_SCALE + avctx->global_quality/2) / avctx->global_quality;
avctx->extradata= av_mallocz(8);
avctx->extradata_size=8;
((uint32_t*)avctx->extradata)[0]= le2me_32(a->inv_qscale);
((uint32_t*)avctx->extradata)[1]= le2me_32(AV_RL32("ASUS"));
for(i=0; i<64; i++){
int q= 32*scale*ff_mpeg1_default_intra_matrix[i];
a->q_intra_matrix[i]= ((a->inv_qscale<<16) + q/2) / q;
}
return 0;
}
#endif /* CONFIG_ASV1_ENCODER || CONFIG_ASV2_ENCODER */
static av_cold int decode_end(AVCodecContext *avctx){
ASV1Context * const a = avctx->priv_data;
av_freep(&a->bitstream_buffer);
av_freep(&a->picture.qscale_table);
a->bitstream_buffer_size=0;
if(a->picture.data[0])
avctx->release_buffer(avctx, &a->picture);
return 0;
}
AVCodec asv1_decoder = {
"asv1",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_ASV1,
sizeof(ASV1Context),
decode_init,
NULL,
decode_end,
decode_frame,
CODEC_CAP_DR1,
.long_name= NULL_IF_CONFIG_SMALL("ASUS V1"),
};
AVCodec asv2_decoder = {
"asv2",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_ASV2,
sizeof(ASV1Context),
decode_init,
NULL,
decode_end,
decode_frame,
CODEC_CAP_DR1,
.long_name= NULL_IF_CONFIG_SMALL("ASUS V2"),
};
#if CONFIG_ASV1_ENCODER
AVCodec asv1_encoder = {
"asv1",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_ASV1,
sizeof(ASV1Context),
encode_init,
encode_frame,
//encode_end,
.pix_fmts= (const enum PixelFormat[]){PIX_FMT_YUV420P, PIX_FMT_NONE},
.long_name= NULL_IF_CONFIG_SMALL("ASUS V1"),
};
#endif
#if CONFIG_ASV2_ENCODER
AVCodec asv2_encoder = {
"asv2",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_ASV2,
sizeof(ASV1Context),
encode_init,
encode_frame,
//encode_end,
.pix_fmts= (const enum PixelFormat[]){PIX_FMT_YUV420P, PIX_FMT_NONE},
.long_name= NULL_IF_CONFIG_SMALL("ASUS V2"),
};
#endif
src/add-ons/media/plugins/ffmpeg/libavcodec/atrac.c
-120
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@@ -1,120 +0,0 @@
/*
* Atrac common functions
* Copyright (c) 2006-2008 Maxim Poliakovski
* Copyright (c) 2006-2008 Benjamin Larsson
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
*/
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#include "avcodec.h"
#include "dsputil.h"
#include "atrac.h"
float sf_table[64];
float qmf_window[48];
static const float qmf_48tap_half[24] = {
-0.00001461907, -0.00009205479,-0.000056157569,0.00030117269,
0.0002422519, -0.00085293897,-0.0005205574, 0.0020340169,
0.00078333891, -0.0042153862, -0.00075614988, 0.0078402944,
-0.000061169922,-0.01344162, 0.0024626821, 0.021736089,
-0.007801671, -0.034090221, 0.01880949, 0.054326009,
-0.043596379, -0.099384367, 0.13207909, 0.46424159
};
/**
* Generate common tables
*/
void atrac_generate_tables(void)
{
int i;
float s;
/* Generate scale factors */
if (!sf_table[63])
for (i=0 ; i<64 ; i++)
sf_table[i] = pow(2.0, (i - 15) / 3.0);
/* Generate the QMF window. */
if (!qmf_window[47])
for (i=0 ; i<24; i++) {
s = qmf_48tap_half[i] * 2.0;
qmf_window[i] = qmf_window[47 - i] = s;
}
}
/**
* Quadrature mirror synthesis filter.
*
* @param inlo lower part of spectrum
* @param inhi higher part of spectrum
* @param nIn size of spectrum buffer
* @param pOut out buffer
* @param delayBuf delayBuf buffer
* @param temp temp buffer
*/
void atrac_iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
{
int i, j;
float *p1, *p3;
memcpy(temp, delayBuf, 46*sizeof(float));
p3 = temp + 46;
/* loop1 */
for(i=0; i<nIn; i+=2){
p3[2*i+0] = inlo[i ] + inhi[i ];
p3[2*i+1] = inlo[i ] - inhi[i ];
p3[2*i+2] = inlo[i+1] + inhi[i+1];
p3[2*i+3] = inlo[i+1] - inhi[i+1];
}
/* loop2 */
p1 = temp;
for (j = nIn; j != 0; j--) {
float s1 = 0.0;
float s2 = 0.0;
for (i = 0; i < 48; i += 2) {
s1 += p1[i] * qmf_window[i];
s2 += p1[i+1] * qmf_window[i+1];
}
pOut[0] = s2;
pOut[1] = s1;
p1 += 2;
pOut += 2;
}
/* Update the delay buffer. */
memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));
}
src/add-ons/media/plugins/ffmpeg/libavcodec/atrac.h
-38
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@@ -1,38 +0,0 @@
/*
* Atrac common data
* Copyright (c) 2009 Maxim Poliakovski
* Copyright (c) 2009 Benjamin Larsson
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Atrac common header
*/
#ifndef AVCODEC_ATRAC_H
#define AVCODEC_ATRAC_H
extern float sf_table[64];
extern float qmf_window[48];
void atrac_generate_tables(void);
void atrac_iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp);
#endif /* AVCODEC_ATRAC_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/atrac1.c
-377
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@@ -1,377 +0,0 @@
/*
* Atrac 1 compatible decoder
* Copyright (c) 2009 Maxim Poliakovski
* Copyright (c) 2009 Benjamin Larsson
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Atrac 1 compatible decoder.
* This decoder handles raw ATRAC1 data and probably SDDS data.
*/
/* Many thanks to Tim Craig for all the help! */
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "atrac.h"
#include "atrac1data.h"
#define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
#define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
#define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
#define AT1_FRAME_SIZE AT1_SU_SIZE * 2
#define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
#define AT1_MAX_CHANNELS 2
#define AT1_QMF_BANDS 3
#define IDX_LOW_BAND 0
#define IDX_MID_BAND 1
#define IDX_HIGH_BAND 2
/**
* Sound unit struct, one unit is used per channel
*/
typedef struct {
int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
int num_bfus; ///< number of Block Floating Units
float* spectrum[2];
DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
DECLARE_ALIGNED(16, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
DECLARE_ALIGNED(16, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter
} AT1SUCtx;
/**
* The atrac1 context, holds all needed parameters for decoding
*/
typedef struct {
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
DECLARE_ALIGNED(16, float, low)[256];
DECLARE_ALIGNED(16, float, mid)[256];
DECLARE_ALIGNED(16, float, high)[512];
float* bands[3];
DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
} AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */
static const uint16_t samples_per_band[3] = {128, 128, 256};
static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
int rev_spec)
{
FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
int transf_size = 1 << nbits;
if (rev_spec) {
int i;
for (i = 0; i < transf_size / 2; i++)
FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
}
ff_imdct_half(mdct_context, out, spec);
}
static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
{
int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
unsigned int start_pos, ref_pos = 0, pos = 0;
for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
float *prev_buf;
int j;
band_samples = samples_per_band[band_num];
log2_block_count = su->log2_block_count[band_num];
/* number of mdct blocks in the current QMF band: 1 - for long mode */
/* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
num_blocks = 1 << log2_block_count;
if (num_blocks == 1) {
/* mdct block size in samples: 128 (long mode, low & mid bands), */
/* 256 (long mode, high band) and 32 (short mode, all bands) */
block_size = band_samples >> log2_block_count;
/* calc transform size in bits according to the block_size_mode */
nbits = mdct_long_nbits[band_num] - log2_block_count;
if (nbits != 5 && nbits != 7 && nbits != 8)
return -1;
} else {
block_size = 32;
nbits = 5;
}
start_pos = 0;
prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
for (j=0; j < num_blocks; j++) {
at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
/* overlap and window */
q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
&su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
start_pos += block_size;
pos += block_size;
}
if (num_blocks == 1)
memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
ref_pos += band_samples;
}
/* Swap buffers so the mdct overlap works */
FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
return 0;
}
/**
* Parse the block size mode byte
*/
static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
{
int log2_block_count_tmp, i;
for (i = 0; i < 2; i++) {
/* low and mid band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp & 1)
return -1;
log2_block_cnt[i] = 2 - log2_block_count_tmp;
}
/* high band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
return -1;
log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
skip_bits(gb, 2);
return 0;
}
static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
float spec[AT1_SU_SAMPLES])
{
int bits_used, band_num, bfu_num, i;
uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
/* parse the info byte (2nd byte) telling how much BFUs were coded */
su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
/* calc number of consumed bits:
num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
+ info_byte_copy(8bits) + log2_block_count_copy(8bits) */
bits_used = su->num_bfus * 10 + 32 +
bfu_amount_tab2[get_bits(gb, 2)] +
(bfu_amount_tab3[get_bits(gb, 3)] << 1);
/* get word length index (idwl) for each BFU */
for (i = 0; i < su->num_bfus; i++)
idwls[i] = get_bits(gb, 4);
/* get scalefactor index (idsf) for each BFU */
for (i = 0; i < su->num_bfus; i++)
idsfs[i] = get_bits(gb, 6);
/* zero idwl/idsf for empty BFUs */
for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
idwls[i] = idsfs[i] = 0;
/* read in the spectral data and reconstruct MDCT spectrum of this channel */
for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
int pos;
int num_specs = specs_per_bfu[bfu_num];
int word_len = !!idwls[bfu_num] + idwls[bfu_num];
float scale_factor = sf_table[idsfs[bfu_num]];
bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
/* check for bitstream overflow */
if (bits_used > AT1_SU_MAX_BITS)
return -1;
/* get the position of the 1st spec according to the block size mode */
pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
if (word_len) {
float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
for (i = 0; i < num_specs; i++) {
/* read in a quantized spec and convert it to
* signed int and then inverse quantization
*/
spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
}
} else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
memset(&spec[pos], 0, num_specs * sizeof(float));
}
}
}
return 0;
}
static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
{
float temp[256];
float iqmf_temp[512 + 46];
/* combine low and middle bands */
atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
/* delay the signal of the high band by 23 samples */
memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
/* combine (low + middle) and high bands */
atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
}
static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
int ch, ret, i;
GetBitContext gb;
float* samples = data;
if (buf_size < 212 * q->channels) {
av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
return -1;
}
for (ch = 0; ch < q->channels; ch++) {
AT1SUCtx* su = &q->SUs[ch];
init_get_bits(&gb, &buf[212 * ch], 212 * 8);
/* parse block_size_mode, 1st byte */
ret = at1_parse_bsm(&gb, su->log2_block_count);
if (ret < 0)
return ret;
ret = at1_unpack_dequant(&gb, su, q->spec);
if (ret < 0)
return ret;
ret = at1_imdct_block(su, q);
if (ret < 0)
return ret;
at1_subband_synthesis(q, su, q->out_samples[ch]);
}
/* interleave; FIXME, should create/use a DSP function */
if (q->channels == 1) {
/* mono */
memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
} else {
/* stereo */
for (i = 0; i < AT1_SU_SAMPLES; i++) {
samples[i * 2] = q->out_samples[0][i];
samples[i * 2 + 1] = q->out_samples[1][i];
}
}
*data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
return avctx->block_align;
}
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data;
avctx->sample_fmt = SAMPLE_FMT_FLT;
q->channels = avctx->channels;
/* Init the mdct transforms */
ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
ff_init_ff_sine_windows(5);
atrac_generate_tables();
dsputil_init(&q->dsp, avctx);
q->bands[0] = q->low;
q->bands[1] = q->mid;
q->bands[2] = q->high;
/* Prepare the mdct overlap buffers */
q->SUs[0].spectrum[0] = q->SUs[0].spec1;
q->SUs[0].spectrum[1] = q->SUs[0].spec2;
q->SUs[1].spectrum[0] = q->SUs[1].spec1;
q->SUs[1].spectrum[1] = q->SUs[1].spec2;
return 0;
}
static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
AT1Ctx *q = avctx->priv_data;
ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]);
return 0;
}
AVCodec atrac1_decoder = {
.name = "atrac1",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ATRAC1,
.priv_data_size = sizeof(AT1Ctx),
.init = atrac1_decode_init,
.close = atrac1_decode_end,
.decode = atrac1_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/atrac1data.h
-64
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@@ -1,64 +0,0 @@
/*
* Atrac 1 compatible decoder data
* Copyright (c) 2009 Maxim Poliakovski
* Copyright (c) 2009 Benjamin Larsson
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Atrac 1 compatible decoder data
*/
#ifndef AVCODEC_ATRAC1DATA_H
#define AVCODEC_ATRAC1DATA_H
#include <stdint.h>
static const uint8_t bfu_amount_tab1[8] = {20, 28, 32, 36, 40, 44, 48, 52};
static const uint8_t bfu_amount_tab2[4] = { 0, 112, 176, 208};
static const uint8_t bfu_amount_tab3[8] = { 0, 24, 36, 48, 72, 108, 132, 156};
/** number of BFUs in each QMF band */
static const uint8_t bfu_bands_t[4] = {0, 20, 36, 52};
/** number of spectral lines in each BFU
* block floating unit = group of spectral frequencies having the
* same quantization parameters like word length and scale factor
*/
static const uint8_t specs_per_bfu[52] = {
8, 8, 8, 8, 4, 4, 4, 4, 8, 8, 8, 8, 6, 6, 6, 6, 6, 6, 6, 6, // low band
6, 6, 6, 6, 7, 7, 7, 7, 9, 9, 9, 9, 10, 10, 10, 10, // midle band
12, 12, 12, 12, 12, 12, 12, 12, 20, 20, 20, 20, 20, 20, 20, 20 // high band
};
/** start position of each BFU in the MDCT spectrum for the long mode */
static const uint16_t bfu_start_long[52] = {
0, 8, 16, 24, 32, 36, 40, 44, 48, 56, 64, 72, 80, 86, 92, 98, 104, 110, 116, 122,
128, 134, 140, 146, 152, 159, 166, 173, 180, 189, 198, 207, 216, 226, 236, 246,
256, 268, 280, 292, 304, 316, 328, 340, 352, 372, 392, 412, 432, 452, 472, 492,
};
/** start position of each BFU in the MDCT spectrum for the short mode */
static const uint16_t bfu_start_short[52] = {
0, 32, 64, 96, 8, 40, 72, 104, 12, 44, 76, 108, 20, 52, 84, 116, 26, 58, 90, 122,
128, 160, 192, 224, 134, 166, 198, 230, 141, 173, 205, 237, 150, 182, 214, 246,
256, 288, 320, 352, 384, 416, 448, 480, 268, 300, 332, 364, 396, 428, 460, 492
};
#endif /* AVCODEC_ATRAC1DATA_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/atrac3.c
-1026
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src/add-ons/media/plugins/ffmpeg/libavcodec/atrac3data.h
-133
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@@ -1,133 +0,0 @@
/*
* Atrac 3 compatible decoder data
* Copyright (c) 2006-2007 Maxim Poliakovski
* Copyright (c) 2006-2007 Benjamin Larsson
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Atrac 3 AKA RealAudio 8 compatible decoder data
*/
#ifndef AVCODEC_ATRAC3DATA_H
#define AVCODEC_ATRAC3DATA_H
#include <stdint.h>
/* VLC tables */
static const uint8_t huffcode1[9] = {
0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F,
};
static const uint8_t huffbits1[9] = {
1,3,3,4,4,5,5,5,5,
};
static const uint8_t huffcode2[5] = {
0x0,0x4,0x5,0x6,0x7,
};
static const uint8_t huffbits2[5] = {
1,3,3,3,3,
};
static const uint8_t huffcode3[7] = {
0x0,0x4,0x5,0xC,0xD,0xE,0xF,
};
static const uint8_t huffbits3[7] = {
1,3,3,4,4,4,4,
};
static const uint8_t huffcode4[9] = {
0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F,
};
static const uint8_t huffbits4[9] = {
1,3,3,4,4,5,5,5,5,
};
static const uint8_t huffcode5[15] = {
0x0,0x2,0x3,0x8,0x9,0xA,0xB,0x1C,0x1D,0x3C,0x3D,0x3E,0x3F,0xC,0xD,
};
static const uint8_t huffbits5[15] = {
2,3,3,4,4,4,4,5,5,6,6,6,6,4,4
};
static const uint8_t huffcode6[31] = {
0x0,0x2,0x3,0x4,0x5,0x6,0x7,0x14,0x15,0x16,0x17,0x18,0x19,0x34,0x35,
0x36,0x37,0x38,0x39,0x3A,0x3B,0x78,0x79,0x7A,0x7B,0x7C,0x7D,0x7E,0x7F,0x8,0x9,
};
static const uint8_t huffbits6[31] = {
3,4,4,4,4,4,4,5,5,5,5,5,5,6,6,6,6,6,6,6,6,7,7,7,7,7,7,7,7,4,4
};
static const uint8_t huffcode7[63] = {
0x0,0x8,0x9,0xA,0xB,0xC,0xD,0xE,0xF,0x10,0x11,0x24,0x25,0x26,0x27,0x28,
0x29,0x2A,0x2B,0x2C,0x2D,0x2E,0x2F,0x30,0x31,0x32,0x33,0x68,0x69,0x6A,0x6B,0x6C,
0x6D,0x6E,0x6F,0x70,0x71,0x72,0x73,0x74,0x75,0xEC,0xED,0xEE,0xEF,0xF0,0xF1,0xF2,
0xF3,0xF4,0xF5,0xF6,0xF7,0xF8,0xF9,0xFA,0xFB,0xFC,0xFD,0xFE,0xFF,0x2,0x3,
};
static const uint8_t huffbits7[63] = {
3,5,5,5,5,5,5,5,5,5,5,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,4,4
};
static const uint8_t huff_tab_sizes[7] = {
9, 5, 7, 9, 15, 31, 63,
};
static const uint8_t* const huff_codes[7] = {
huffcode1,huffcode2,huffcode3,huffcode4,huffcode5,huffcode6,huffcode7,
};
static const uint8_t* const huff_bits[7] = {
huffbits1,huffbits2,huffbits3,huffbits4,huffbits5,huffbits6,huffbits7,
};
static const uint16_t atrac3_vlc_offs[] = {
0,512,1024,1536,2048,2560,3072,3584,4096
};
/* selector tables */
static const uint8_t CLCLengthTab[8] = {0, 4, 3, 3, 4, 4, 5, 6};
static const int8_t seTab_0[4] = {0, 1, -2, -1};
static const int8_t decTable1[18] = {0,0, 0,1, 0,-1, 1,0, -1,0, 1,1, 1,-1, -1,1, -1,-1};
/* tables for the scalefactor decoding */
static const float iMaxQuant[8] = {
0.0, 1.0/1.5, 1.0/2.5, 1.0/3.5, 1.0/4.5, 1.0/7.5, 1.0/15.5, 1.0/31.5
};
static const uint16_t subbandTab[33] = {
0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224,
256, 288, 320, 352, 384, 416, 448, 480, 512, 576, 640, 704, 768, 896, 1024
};
/* joint stereo related tables */
static const float matrixCoeffs[8] = {0.0, 2.0, 2.0, 2.0, 0.0, 0.0, 1.0, 1.0};
#endif /* AVCODEC_ATRAC3DATA_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/audioconvert.c
-243
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@@ -1,243 +0,0 @@
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio conversion
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/avstring.h"
#include "libavutil/libm.h"
#include "avcodec.h"
#include "audioconvert.h"
typedef struct SampleFmtInfo {
const char *name;
int bits;
} SampleFmtInfo;
/** this table gives more information about formats */
static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = {
[SAMPLE_FMT_U8] = { .name = "u8", .bits = 8 },
[SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 },
[SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 },
[SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 },
[SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64 },
};
const char *avcodec_get_sample_fmt_name(int sample_fmt)
{
if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB)
return NULL;
return sample_fmt_info[sample_fmt].name;
}
enum SampleFormat avcodec_get_sample_fmt(const char* name)
{
int i;
for (i=0; i < SAMPLE_FMT_NB; i++)
if (!strcmp(sample_fmt_info[i].name, name))
return i;
return SAMPLE_FMT_NONE;
}
void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt)
{
/* print header */
if (sample_fmt < 0)
snprintf (buf, buf_size, "name " " depth");
else if (sample_fmt < SAMPLE_FMT_NB) {
SampleFmtInfo info= sample_fmt_info[sample_fmt];
snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits);
}
}
static const char* const channel_names[]={
"FL", "FR", "FC", "LFE", "BL", "BR", "FLC", "FRC",
"BC", "SL", "SR", "TC", "TFL", "TFC", "TFR", "TBL",
"TBC", "TBR",
[29] = "DL",
[30] = "DR",
};
static const char *get_channel_name(int channel_id)
{
if (channel_id<0 || channel_id>=FF_ARRAY_ELEMS(channel_names))
return NULL;
return channel_names[channel_id];
}
int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name)
{
switch(nb_channels) {
case 1: return CH_LAYOUT_MONO;
case 2: return CH_LAYOUT_STEREO;
case 3: return CH_LAYOUT_SURROUND;
case 4: return CH_LAYOUT_QUAD;
case 5: return CH_LAYOUT_5POINT0;
case 6: return CH_LAYOUT_5POINT1;
case 8: return CH_LAYOUT_7POINT1;
default: return 0;
}
}
static const struct {
const char *name;
int nb_channels;
int64_t layout;
} channel_layout_map[] = {
{ "mono", 1, CH_LAYOUT_MONO },
{ "stereo", 2, CH_LAYOUT_STEREO },
{ "4.0", 4, CH_LAYOUT_4POINT0 },
{ "quad", 4, CH_LAYOUT_QUAD },
{ "5.0", 5, CH_LAYOUT_5POINT0 },
{ "5.0", 5, CH_LAYOUT_5POINT0_BACK },
{ "5.1", 6, CH_LAYOUT_5POINT1 },
{ "5.1", 6, CH_LAYOUT_5POINT1_BACK },
{ "5.1+downmix", 8, CH_LAYOUT_5POINT1|CH_LAYOUT_STEREO_DOWNMIX, },
{ "7.1", 8, CH_LAYOUT_7POINT1 },
{ "7.1(wide)", 8, CH_LAYOUT_7POINT1_WIDE },
{ "7.1+downmix", 10, CH_LAYOUT_7POINT1|CH_LAYOUT_STEREO_DOWNMIX, },
{ 0 }
};
void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout)
{
int i;
for (i=0; channel_layout_map[i].name; i++)
if (nb_channels == channel_layout_map[i].nb_channels &&
channel_layout == channel_layout_map[i].layout) {
av_strlcpy(buf, channel_layout_map[i].name, buf_size);
return;
}
snprintf(buf, buf_size, "%d channels", nb_channels);
if (channel_layout) {
int i,ch;
av_strlcat(buf, " (", buf_size);
for(i=0,ch=0; i<64; i++) {
if ((channel_layout & (1L<<i))) {
const char *name = get_channel_name(i);
if (name) {
if (ch>0) av_strlcat(buf, "|", buf_size);
av_strlcat(buf, name, buf_size);
}
ch++;
}
}
av_strlcat(buf, ")", buf_size);
}
}
int avcodec_channel_layout_num_channels(int64_t channel_layout)
{
int count;
uint64_t x = channel_layout;
for (count = 0; x; count++)
x &= x-1; // unset lowest set bit
return count;
}
struct AVAudioConvert {
int in_channels, out_channels;
int fmt_pair;
};
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat in_fmt, int in_channels,
const float *matrix, int flags)
{
AVAudioConvert *ctx;
if (in_channels!=out_channels)
return NULL; /* FIXME: not supported */
ctx = av_malloc(sizeof(AVAudioConvert));
if (!ctx)
return NULL;
ctx->in_channels = in_channels;
ctx->out_channels = out_channels;
ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt;
return ctx;
}
void av_audio_convert_free(AVAudioConvert *ctx)
{
av_free(ctx);
}
int av_audio_convert(AVAudioConvert *ctx,
void * const out[6], const int out_stride[6],
const void * const in[6], const int in_stride[6], int len)
{
int ch;
//FIXME optimize common cases
for(ch=0; ch<ctx->out_channels; ch++){
const int is= in_stride[ch];
const int os= out_stride[ch];
const uint8_t *pi= in[ch];
uint8_t *po= out[ch];
uint8_t *end= po + os*len;
if(!out[ch])
continue;
#define CONV(ofmt, otype, ifmt, expr)\
if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
do{\
*(otype*)po = expr; pi += is; po += os;\
}while(po < end);\
}
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding ?
CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(const uint8_t*)pi)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(const int16_t*)pi)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(const int32_t*)pi)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(const double*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(const double*)pi)
else return -1;
}
return 0;
}
src/add-ons/media/plugins/ffmpeg/libavcodec/audioconvert.h
-115
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@@ -1,115 +0,0 @@
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2008 Peter Ross
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AUDIOCONVERT_H
#define AVCODEC_AUDIOCONVERT_H
/**
* @file
* Audio format conversion routines
*/
#include "avcodec.h"
/**
* Generate string corresponding to the sample format with
* number sample_fmt, or a header if sample_fmt is negative.
*
* @param[in] buf the buffer where to write the string
* @param[in] buf_size the size of buf
* @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
* a negative value to print the corresponding header.
* Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
*/
void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
/**
* @return NULL on error
*/
const char *avcodec_get_sample_fmt_name(int sample_fmt);
/**
* @return SAMPLE_FMT_NONE on error
*/
enum SampleFormat avcodec_get_sample_fmt(const char* name);
/**
* @return NULL on error
*/
const char *avcodec_get_channel_name(int channel_id);
/**
* Return description of channel layout
*/
void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout);
/**
* Guess the channel layout
* @param nb_channels
* @param codec_id Codec identifier, or CODEC_ID_NONE if unknown
* @param fmt_name Format name, or NULL if unknown
* @return Channel layout mask
*/
int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);
/**
* @return the number of channels in the channel layout.
*/
int avcodec_channel_layout_num_channels(int64_t channel_layout);
struct AVAudioConvert;
typedef struct AVAudioConvert AVAudioConvert;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param out_channels Number of output channels
* @param in_fmt Input sample format
* @param in_channels Number of input channels
* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
* @param flags See FF_MM_xx
* @return NULL on error
*/
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat in_fmt, int in_channels,
const float *matrix, int flags);
/**
* Free audio sample format converter context
*/
void av_audio_convert_free(AVAudioConvert *ctx);
/**
* Convert between audio sample formats
* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
* @param[in] out_stride distance between consecutive output samples (measured in bytes)
* @param[in] in array of input buffers for each channel
* @param[in] in_stride distance between consecutive input samples (measured in bytes)
* @param len length of audio frame size (measured in samples)
*/
int av_audio_convert(AVAudioConvert *ctx,
void * const out[6], const int out_stride[6],
const void * const in[6], const int in_stride[6], int len);
#endif /* AVCODEC_AUDIOCONVERT_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/aura.c
-138
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@@ -1,138 +0,0 @@
/*
* Aura 2 decoder
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Aura 2 decoder
*/
#include "avcodec.h"
typedef struct AuraDecodeContext {
AVCodecContext *avctx;
AVFrame frame;
} AuraDecodeContext;
static av_cold int aura_decode_init(AVCodecContext *avctx)
{
AuraDecodeContext *s = avctx->priv_data;
s->avctx = avctx;
/* width needs to be divisible by 4 for this codec to work */
if (avctx->width & 0x3)
return -1;
avctx->pix_fmt = PIX_FMT_YUV422P;
return 0;
}
static int aura_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *pkt)
{
AuraDecodeContext *s=avctx->priv_data;
uint8_t *Y, *U, *V;
uint8_t val;
int x, y;
const uint8_t *buf = pkt->data;
/* prediction error tables (make it clear that they are signed values) */
const int8_t *delta_table = (const int8_t*)buf + 16;
if (pkt->size != 48 + avctx->height * avctx->width) {
av_log(avctx, AV_LOG_ERROR, "got a buffer with %d bytes when %d were expected\n",
pkt->size, 48 + avctx->height * avctx->width);
return -1;
}
/* pixel data starts 48 bytes in, after 3x16-byte tables */
buf += 48;
if(s->frame.data[0])
avctx->release_buffer(avctx, &s->frame);
s->frame.buffer_hints = FF_BUFFER_HINTS_VALID;
s->frame.reference = 0;
if(avctx->get_buffer(avctx, &s->frame) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
Y = s->frame.data[0];
U = s->frame.data[1];
V = s->frame.data[2];
/* iterate through each line in the height */
for (y = 0; y < avctx->height; y++) {
/* reset predictors */
val = *buf++;
U[0] = val & 0xF0;
Y[0] = val << 4;
val = *buf++;
V[0] = val & 0xF0;
Y[1] = Y[0] + delta_table[val & 0xF];
Y += 2; U++; V++;
/* iterate through the remaining pixel groups (4 pixels/group) */
for (x = 1; x < (avctx->width >> 1); x++) {
val = *buf++;
U[0] = U[-1] + delta_table[val >> 4];
Y[0] = Y[-1] + delta_table[val & 0xF];
val = *buf++;
V[0] = V[-1] + delta_table[val >> 4];
Y[1] = Y[ 0] + delta_table[val & 0xF];
Y += 2; U++; V++;
}
Y += s->frame.linesize[0] - avctx->width;
U += s->frame.linesize[1] - (avctx->width >> 1);
V += s->frame.linesize[2] - (avctx->width >> 1);
}
*data_size=sizeof(AVFrame);
*(AVFrame*)data= s->frame;
return pkt->size;
}
static av_cold int aura_decode_end(AVCodecContext *avctx)
{
AuraDecodeContext *s = avctx->priv_data;
if (s->frame.data[0])
avctx->release_buffer(avctx, &s->frame);
return 0;
}
AVCodec aura2_decoder = {
"aura2",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_AURA2,
sizeof(AuraDecodeContext),
aura_decode_init,
NULL,
aura_decode_end,
aura_decode_frame,
CODEC_CAP_DR1,
NULL,
.long_name = NULL_IF_CONFIG_SMALL("Auravision Aura 2"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/avcodec.h
-3968
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src/add-ons/media/plugins/ffmpeg/libavcodec/avfft.c
-142
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@@ -1,142 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/mem.h"
#include "avfft.h"
#include "fft.h"
/* FFT */
FFTContext *av_fft_init(int nbits, int inverse)
{
FFTContext *s = av_malloc(sizeof(*s));
if (s)
ff_fft_init(s, nbits, inverse);
return s;
}
void av_fft_permute(FFTContext *s, FFTComplex *z)
{
s->fft_permute(s, z);
}
void av_fft_calc(FFTContext *s, FFTComplex *z)
{
s->fft_calc(s, z);
}
void av_fft_end(FFTContext *s)
{
if (s) {
ff_fft_end(s);
av_free(s);
}
}
#if CONFIG_MDCT
FFTContext *av_mdct_init(int nbits, int inverse, double scale)
{
FFTContext *s = av_malloc(sizeof(*s));
if (s)
ff_mdct_init(s, nbits, inverse, scale);
return s;
}
void av_imdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input)
{
s->imdct_calc(s, output, input);
}
void av_imdct_half(FFTContext *s, FFTSample *output, const FFTSample *input)
{
s->imdct_half(s, output, input);
}
void av_mdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input)
{
s->mdct_calc(s, output, input);
}
void av_mdct_end(FFTContext *s)
{
if (s) {
ff_mdct_end(s);
av_free(s);
}
}
#endif /* CONFIG_MDCT */
#if CONFIG_RDFT
RDFTContext *av_rdft_init(int nbits, enum RDFTransformType trans)
{
RDFTContext *s = av_malloc(sizeof(*s));
if (s)
ff_rdft_init(s, nbits, trans);
return s;
}
void av_rdft_calc(RDFTContext *s, FFTSample *data)
{
ff_rdft_calc(s, data);
}
void av_rdft_end(RDFTContext *s)
{
if (s) {
ff_rdft_end(s);
av_free(s);
}
}
#endif /* CONFIG_RDFT */
#if CONFIG_DCT
DCTContext *av_dct_init(int nbits, enum DCTTransformType inverse)
{
DCTContext *s = av_malloc(sizeof(*s));
if (s)
ff_dct_init(s, nbits, inverse);
return s;
}
void av_dct_calc(DCTContext *s, FFTSample *data)
{
ff_dct_calc(s, data);
}
void av_dct_end(DCTContext *s)
{
if (s) {
ff_dct_end(s);
av_free(s);
}
}
#endif /* CONFIG_DCT */
src/add-ons/media/plugins/ffmpeg/libavcodec/avfft.h
-99
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@@ -1,99 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AVFFT_H
#define AVCODEC_AVFFT_H
typedef float FFTSample;
typedef struct FFTComplex {
FFTSample re, im;
} FFTComplex;
typedef struct FFTContext FFTContext;
/**
* Set up a complex FFT.
* @param nbits log2 of the length of the input array
* @param inverse if 0 perform the forward transform, if 1 perform the inverse
*/
FFTContext *av_fft_init(int nbits, int inverse);
/**
* Do the permutation needed BEFORE calling ff_fft_calc().
*/
void av_fft_permute(FFTContext *s, FFTComplex *z);
/**
* Do a complex FFT with the parameters defined in av_fft_init(). The
* input data must be permuted before. No 1.0/sqrt(n) normalization is done.
*/
void av_fft_calc(FFTContext *s, FFTComplex *z);
void av_fft_end(FFTContext *s);
FFTContext *av_mdct_init(int nbits, int inverse, double scale);
void av_imdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input);
void av_imdct_half(FFTContext *s, FFTSample *output, const FFTSample *input);
void av_mdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input);
void av_mdct_end(FFTContext *s);
/* Real Discrete Fourier Transform */
enum RDFTransformType {
DFT_R2C,
IDFT_C2R,
IDFT_R2C,
DFT_C2R,
};
typedef struct RDFTContext RDFTContext;
/**
* Set up a real FFT.
* @param nbits log2 of the length of the input array
* @param trans the type of transform
*/
RDFTContext *av_rdft_init(int nbits, enum RDFTransformType trans);
void av_rdft_calc(RDFTContext *s, FFTSample *data);
void av_rdft_end(RDFTContext *s);
/* Discrete Cosine Transform */
typedef struct DCTContext DCTContext;
enum DCTTransformType {
DCT_II = 0,
DCT_III,
DCT_I,
DST_I,
};
/**
* Sets up DCT.
* @param nbits size of the input array:
* (1 << nbits) for DCT-II, DCT-III and DST-I
* (1 << nbits) + 1 for DCT-I
*
* @note the first element of the input of DST-I is ignored
*/
DCTContext *av_dct_init(int nbits, enum DCTTransformType type);
void av_dct_calc(DCTContext *s, FFTSample *data);
void av_dct_end (DCTContext *s);
#endif /* AVCODEC_AVFFT_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/avpacket.c
-99
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@@ -1,99 +0,0 @@
/*
* AVPacket functions for libavcodec
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
void av_destruct_packet_nofree(AVPacket *pkt)
{
pkt->data = NULL; pkt->size = 0;
}
void av_destruct_packet(AVPacket *pkt)
{
av_free(pkt->data);
pkt->data = NULL; pkt->size = 0;
}
void av_init_packet(AVPacket *pkt)
{
pkt->pts = AV_NOPTS_VALUE;
pkt->dts = AV_NOPTS_VALUE;
pkt->pos = -1;
pkt->duration = 0;
pkt->convergence_duration = 0;
pkt->flags = 0;
pkt->stream_index = 0;
pkt->destruct= NULL;
}
int av_new_packet(AVPacket *pkt, int size)
{
uint8_t *data= NULL;
if((unsigned)size < (unsigned)size + FF_INPUT_BUFFER_PADDING_SIZE)
data = av_malloc(size + FF_INPUT_BUFFER_PADDING_SIZE);
if (data){
memset(data + size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
}else
size=0;
av_init_packet(pkt);
pkt->data = data;
pkt->size = size;
pkt->destruct = av_destruct_packet;
if(!data)
return AVERROR(ENOMEM);
return 0;
}
void av_shrink_packet(AVPacket *pkt, int size)
{
if (pkt->size <= size) return;
pkt->size = size;
memset(pkt->data + size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
}
int av_dup_packet(AVPacket *pkt)
{
if (((pkt->destruct == av_destruct_packet_nofree) || (pkt->destruct == NULL)) && pkt->data) {
uint8_t *data;
/* We duplicate the packet and don't forget to add the padding again. */
if((unsigned)pkt->size > (unsigned)pkt->size + FF_INPUT_BUFFER_PADDING_SIZE)
return AVERROR(ENOMEM);
data = av_malloc(pkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!data) {
return AVERROR(ENOMEM);
}
memcpy(data, pkt->data, pkt->size);
memset(data + pkt->size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
pkt->data = data;
pkt->destruct = av_destruct_packet;
}
return 0;
}
void av_free_packet(AVPacket *pkt)
{
if (pkt) {
if (pkt->destruct) pkt->destruct(pkt);
pkt->data = NULL; pkt->size = 0;
}
}
src/add-ons/media/plugins/ffmpeg/libavcodec/avs.c
-164
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@@ -1,164 +0,0 @@
/*
* AVS video decoder.
* Copyright (c) 2006 Aurelien Jacobs <aurel@gnuage.org>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "get_bits.h"
typedef struct {
AVFrame picture;
} AvsContext;
typedef enum {
AVS_VIDEO = 0x01,
AVS_AUDIO = 0x02,
AVS_PALETTE = 0x03,
AVS_GAME_DATA = 0x04,
} AvsBlockType;
typedef enum {
AVS_I_FRAME = 0x00,
AVS_P_FRAME_3X3 = 0x01,
AVS_P_FRAME_2X2 = 0x02,
AVS_P_FRAME_2X3 = 0x03,
} AvsVideoSubType;
static int
avs_decode_frame(AVCodecContext * avctx,
void *data, int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AvsContext *const avs = avctx->priv_data;
AVFrame *picture = data;
AVFrame *const p = (AVFrame *) & avs->picture;
const uint8_t *table, *vect;
uint8_t *out;
int i, j, x, y, stride, vect_w = 3, vect_h = 3;
AvsVideoSubType sub_type;
AvsBlockType type;
GetBitContext change_map;
if (avctx->reget_buffer(avctx, p)) {
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
return -1;
}
p->reference = 1;
p->pict_type = FF_P_TYPE;
p->key_frame = 0;
out = avs->picture.data[0];
stride = avs->picture.linesize[0];
sub_type = buf[0];
type = buf[1];
buf += 4;
if (type == AVS_PALETTE) {
int first, last;
uint32_t *pal = (uint32_t *) avs->picture.data[1];
first = AV_RL16(buf);
last = first + AV_RL16(buf + 2);
buf += 4;
for (i=first; i<last; i++, buf+=3)
pal[i] = (buf[0] << 18) | (buf[1] << 10) | (buf[2] << 2);
sub_type = buf[0];
type = buf[1];
buf += 4;
}
if (type != AVS_VIDEO)
return -1;
switch (sub_type) {
case AVS_I_FRAME:
p->pict_type = FF_I_TYPE;
p->key_frame = 1;
case AVS_P_FRAME_3X3:
vect_w = 3;
vect_h = 3;
break;
case AVS_P_FRAME_2X2:
vect_w = 2;
vect_h = 2;
break;
case AVS_P_FRAME_2X3:
vect_w = 2;
vect_h = 3;
break;
default:
return -1;
}
table = buf + (256 * vect_w * vect_h);
if (sub_type != AVS_I_FRAME) {
int map_size = ((318 / vect_w + 7) / 8) * (198 / vect_h);
init_get_bits(&change_map, table, map_size);
table += map_size;
}
for (y=0; y<198; y+=vect_h) {
for (x=0; x<318; x+=vect_w) {
if (sub_type == AVS_I_FRAME || get_bits1(&change_map)) {
vect = &buf[*table++ * (vect_w * vect_h)];
for (j=0; j<vect_w; j++) {
out[(y + 0) * stride + x + j] = vect[(0 * vect_w) + j];
out[(y + 1) * stride + x + j] = vect[(1 * vect_w) + j];
if (vect_h == 3)
out[(y + 2) * stride + x + j] =
vect[(2 * vect_w) + j];
}
}
}
if (sub_type != AVS_I_FRAME)
align_get_bits(&change_map);
}
*picture = *(AVFrame *) & avs->picture;
*data_size = sizeof(AVPicture);
return buf_size;
}
static av_cold int avs_decode_init(AVCodecContext * avctx)
{
avctx->pix_fmt = PIX_FMT_PAL8;
return 0;
}
AVCodec avs_decoder = {
"avs",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_AVS,
sizeof(AvsContext),
avs_decode_init,
NULL,
NULL,
avs_decode_frame,
CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("AVS (Audio Video Standard) video"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/beosthread.c
-183
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@@ -1,183 +0,0 @@
/*
* Copyright (c) 2004 François Revol <revol@free.fr>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
//#define DEBUG
#include "avcodec.h"
#include <OS.h>
typedef struct ThreadContext{
AVCodecContext *avctx;
thread_id thread;
sem_id work_sem;
sem_id done_sem;
int (*func)(AVCodecContext *c, void *arg);
void *arg;
int ret;
}ThreadContext;
// it's odd Be never patented that :D
struct benaphore {
vint32 atom;
sem_id sem;
};
static inline int lock_ben(struct benaphore *ben)
{
if (atomic_add(&ben->atom, 1) > 0)
return acquire_sem(ben->sem);
return B_OK;
}
static inline int unlock_ben(struct benaphore *ben)
{
if (atomic_add(&ben->atom, -1) > 1)
return release_sem(ben->sem);
return B_OK;
}
static struct benaphore av_thread_lib_ben;
static int32 ff_thread_func(void *v){
ThreadContext *c= v;
for(;;){
//printf("thread_func %X enter wait\n", (int)v); fflush(stdout);
acquire_sem(c->work_sem);
//printf("thread_func %X after wait (func=%X)\n", (int)v, (int)c->func); fflush(stdout);
if(c->func)
c->ret= c->func(c->avctx, c->arg);
else
return 0;
//printf("thread_func %X signal complete\n", (int)v); fflush(stdout);
release_sem(c->done_sem);
}
return B_OK;
}
/**
* Free what has been allocated by avcodec_thread_init().
* Must be called after decoding has finished, especially do not call while avcodec_thread_execute() is running.
*/
void avcodec_thread_free(AVCodecContext *s){
ThreadContext *c= s->thread_opaque;
int i;
int32 ret;
for(i=0; i<s->thread_count; i++){
c[i].func= NULL;
release_sem(c[i].work_sem);
wait_for_thread(c[i].thread, &ret);
if(c[i].work_sem > B_OK) delete_sem(c[i].work_sem);
if(c[i].done_sem > B_OK) delete_sem(c[i].done_sem);
}
av_freep(&s->thread_opaque);
}
static int avcodec_thread_execute(AVCodecContext *s, int (*func)(AVCodecContext *c2, void *arg2),void *arg, int *ret, int count, int size){
ThreadContext *c= s->thread_opaque;
int i;
assert(s == c->avctx);
assert(count <= s->thread_count);
/* note, we can be certain that this is not called with the same AVCodecContext by different threads at the same time */
for(i=0; i<count; i++){
c[i].arg= (char*)arg + i*size;
c[i].func= func;
c[i].ret= 12345;
release_sem(c[i].work_sem);
}
for(i=0; i<count; i++){
acquire_sem(c[i].done_sem);
c[i].func= NULL;
if(ret) ret[i]= c[i].ret;
}
return 0;
}
int avcodec_thread_init(AVCodecContext *s, int thread_count){
int i;
ThreadContext *c;
s->thread_count= thread_count;
if (thread_count <= 1)
return 0;
assert(!s->thread_opaque);
c= av_mallocz(sizeof(ThreadContext)*thread_count);
s->thread_opaque= c;
for(i=0; i<thread_count; i++){
//printf("init semaphors %d\n", i); fflush(stdout);
c[i].avctx= s;
if((c[i].work_sem = create_sem(0, "ff work sem")) < B_OK)
goto fail;
if((c[i].done_sem = create_sem(0, "ff done sem")) < B_OK)
goto fail;
//printf("create thread %d\n", i); fflush(stdout);
c[i].thread = spawn_thread(ff_thread_func, "libavcodec thread", B_LOW_PRIORITY, &c[i] );
if( c[i].thread < B_OK ) goto fail;
resume_thread(c[i].thread );
}
//printf("init done\n"); fflush(stdout);
s->execute= avcodec_thread_execute;
return 0;
fail:
avcodec_thread_free(s);
return -1;
}
/* provide a mean to serialize calls to avcodec_*() for thread safety. */
int avcodec_thread_lock_lib(void)
{
return lock_ben(&av_thread_lib_ben);
}
int avcodec_thread_unlock_lib(void)
{
return unlock_ben(&av_thread_lib_ben);
}
/* our versions of _init and _fini (which are called by those actually from crt.o) */
void initialize_after(void)
{
av_thread_lib_ben.atom = 0;
av_thread_lib_ben.sem = create_sem(0, "libavcodec benaphore");
}
void uninitialize_before(void)
{
delete_sem(av_thread_lib_ben.sem);
}
src/add-ons/media/plugins/ffmpeg/libavcodec/bethsoftvideo.c
-143
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@@ -1,143 +0,0 @@
/*
* Bethesda VID video decoder
* Copyright (C) 2007 Nicholas Tung
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* @brief Bethesda Softworks VID Video Decoder
* @author Nicholas Tung [ntung (at. ntung com] (2007-03)
* @sa http://wiki.multimedia.cx/index.php?title=Bethsoft_VID
* @sa http://www.svatopluk.com/andux/docs/dfvid.html
*/
#include "libavutil/common.h"
#include "dsputil.h"
#include "bethsoftvideo.h"
#include "bytestream.h"
typedef struct BethsoftvidContext {
AVFrame frame;
} BethsoftvidContext;
static av_cold int bethsoftvid_decode_init(AVCodecContext *avctx)
{
BethsoftvidContext *vid = avctx->priv_data;
vid->frame.reference = 1;
vid->frame.buffer_hints = FF_BUFFER_HINTS_VALID |
FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
avctx->pix_fmt = PIX_FMT_PAL8;
return 0;
}
static void set_palette(AVFrame * frame, const uint8_t * palette_buffer)
{
uint32_t * palette = (uint32_t *)frame->data[1];
int a;
for(a = 0; a < 256; a++){
palette[a] = AV_RB24(&palette_buffer[a * 3]) * 4;
}
frame->palette_has_changed = 1;
}
static int bethsoftvid_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
BethsoftvidContext * vid = avctx->priv_data;
char block_type;
uint8_t * dst;
uint8_t * frame_end;
int remaining = avctx->width; // number of bytes remaining on a line
const int wrap_to_next_line = vid->frame.linesize[0] - avctx->width;
int code;
int yoffset;
if (avctx->reget_buffer(avctx, &vid->frame)) {
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
return -1;
}
dst = vid->frame.data[0];
frame_end = vid->frame.data[0] + vid->frame.linesize[0] * avctx->height;
switch(block_type = *buf++){
case PALETTE_BLOCK:
set_palette(&vid->frame, buf);
return 0;
case VIDEO_YOFF_P_FRAME:
yoffset = bytestream_get_le16(&buf);
if(yoffset >= avctx->height)
return -1;
dst += vid->frame.linesize[0] * yoffset;
}
// main code
while((code = *buf++)){
int length = code & 0x7f;
// copy any bytes starting at the current position, and ending at the frame width
while(length > remaining){
if(code < 0x80)
bytestream_get_buffer(&buf, dst, remaining);
else if(block_type == VIDEO_I_FRAME)
memset(dst, buf[0], remaining);
length -= remaining; // decrement the number of bytes to be copied
dst += remaining + wrap_to_next_line; // skip over extra bytes at end of frame
remaining = avctx->width;
if(dst == frame_end)
goto end;
}
// copy any remaining bytes after / if line overflows
if(code < 0x80)
bytestream_get_buffer(&buf, dst, length);
else if(block_type == VIDEO_I_FRAME)
memset(dst, *buf++, length);
remaining -= length;
dst += length;
}
end:
*data_size = sizeof(AVFrame);
*(AVFrame*)data = vid->frame;
return buf_size;
}
static av_cold int bethsoftvid_decode_end(AVCodecContext *avctx)
{
BethsoftvidContext * vid = avctx->priv_data;
if(vid->frame.data[0])
avctx->release_buffer(avctx, &vid->frame);
return 0;
}
AVCodec bethsoftvid_decoder = {
.name = "bethsoftvid",
.type = AVMEDIA_TYPE_VIDEO,
.id = CODEC_ID_BETHSOFTVID,
.priv_data_size = sizeof(BethsoftvidContext),
.init = bethsoftvid_decode_init,
.close = bethsoftvid_decode_end,
.decode = bethsoftvid_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Bethesda VID video"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/bethsoftvideo.h
-36
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@@ -1,36 +0,0 @@
/*
* Bethesda VID video decoder
* Copyright (C) 2007 Nicholas Tung
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_BETHSOFTVIDEO_H
#define AVCODEC_BETHSOFTVIDEO_H
enum BethsoftVidBlockType
{
PALETTE_BLOCK = 0x02,
FIRST_AUDIO_BLOCK = 0x7c,
AUDIO_BLOCK = 0x7d,
VIDEO_I_FRAME = 0x03,
VIDEO_P_FRAME = 0x01,
VIDEO_YOFF_P_FRAME = 0x04,
EOF_BLOCK = 0x14,
};
#endif /* AVCODEC_BETHSOFTVIDEO_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/bfi.c
-184
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@@ -1,184 +0,0 @@
/*
* Brute Force & Ignorance (BFI) video decoder
* Copyright (c) 2008 Sisir Koppaka
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* @brief Brute Force & Ignorance (.bfi) video decoder
* @author Sisir Koppaka ( sisir.koppaka at gmail dot com )
* @sa http://wiki.multimedia.cx/index.php?title=BFI
*/
#include "libavutil/common.h"
#include "avcodec.h"
#include "bytestream.h"
typedef struct BFIContext {
AVCodecContext *avctx;
AVFrame frame;
uint8_t *dst;
} BFIContext;
static av_cold int bfi_decode_init(AVCodecContext * avctx)
{
BFIContext *bfi = avctx->priv_data;
avctx->pix_fmt = PIX_FMT_PAL8;
bfi->dst = av_mallocz(avctx->width * avctx->height);
return 0;
}
static int bfi_decode_frame(AVCodecContext * avctx, void *data,
int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
BFIContext *bfi = avctx->priv_data;
uint8_t *dst = bfi->dst;
uint8_t *src, *dst_offset, colour1, colour2;
uint8_t *frame_end = bfi->dst + avctx->width * avctx->height;
uint32_t *pal;
int i, j, height = avctx->height;
if (bfi->frame.data[0])
avctx->release_buffer(avctx, &bfi->frame);
bfi->frame.reference = 1;
if (avctx->get_buffer(avctx, &bfi->frame) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
/* Set frame parameters and palette, if necessary */
if (!avctx->frame_number) {
bfi->frame.pict_type = FF_I_TYPE;
bfi->frame.key_frame = 1;
/* Setting the palette */
if(avctx->extradata_size>768) {
av_log(NULL, AV_LOG_ERROR, "Palette is too large.\n");
return -1;
}
pal = (uint32_t *) bfi->frame.data[1];
for (i = 0; i < avctx->extradata_size / 3; i++) {
int shift = 16;
*pal = 0;
for (j = 0; j < 3; j++, shift -= 8)
*pal +=
((avctx->extradata[i * 3 + j] << 2) |
(avctx->extradata[i * 3 + j] >> 4)) << shift;
pal++;
}
bfi->frame.palette_has_changed = 1;
} else {
bfi->frame.pict_type = FF_P_TYPE;
bfi->frame.key_frame = 0;
}
buf += 4; //Unpacked size, not required.
while (dst != frame_end) {
static const uint8_t lentab[4]={0,2,0,1};
unsigned int byte = *buf++, av_uninit(offset);
unsigned int code = byte >> 6;
unsigned int length = byte & ~0xC0;
/* Get length and offset(if required) */
if (length == 0) {
if (code == 1) {
length = bytestream_get_byte(&buf);
offset = bytestream_get_le16(&buf);
} else {
length = bytestream_get_le16(&buf);
if (code == 2 && length == 0)
break;
}
} else {
if (code == 1)
offset = bytestream_get_byte(&buf);
}
/* Do boundary check */
if (dst + (length<<lentab[code]) > frame_end)
break;
switch (code) {
case 0: //Normal Chain
bytestream_get_buffer(&buf, dst, length);
dst += length;
break;
case 1: //Back Chain
dst_offset = dst - offset;
length *= 4; //Convert dwords to bytes.
if (dst_offset < bfi->dst)
break;
while (length--)
*dst++ = *dst_offset++;
break;
case 2: //Skip Chain
dst += length;
break;
case 3: //Fill Chain
colour1 = bytestream_get_byte(&buf);
colour2 = bytestream_get_byte(&buf);
while (length--) {
*dst++ = colour1;
*dst++ = colour2;
}
break;
}
}
src = bfi->dst;
dst = bfi->frame.data[0];
while (height--) {
memcpy(dst, src, avctx->width);
src += avctx->width;
dst += bfi->frame.linesize[0];
}
*data_size = sizeof(AVFrame);
*(AVFrame *) data = bfi->frame;
return buf_size;
}
static av_cold int bfi_decode_close(AVCodecContext * avctx)
{
BFIContext *bfi = avctx->priv_data;
if (bfi->frame.data[0])
avctx->release_buffer(avctx, &bfi->frame);
av_free(bfi->dst);
return 0;
}
AVCodec bfi_decoder = {
.name = "bfi",
.type = AVMEDIA_TYPE_VIDEO,
.id = CODEC_ID_BFI,
.priv_data_size = sizeof(BFIContext),
.init = bfi_decode_init,
.close = bfi_decode_close,
.decode = bfi_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Brute Force & Ignorance"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/bgmc.c
-567
View File
@@ -1,567 +0,0 @@
/*
* Block Gilbert-Moore decoder
* Copyright (c) 2010 Thilo Borgmann <thilo.borgmann _at_ googlemail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Block Gilbert-Moore decoder as used by MPEG-4 ALS
* @author Thilo Borgmann <thilo.borgmann _at_ googlemail.com>
*/
#include "bgmc.h"
#define FREQ_BITS 14 // bits used by frequency counters
#define VALUE_BITS 18 // bits used to represent the values
#define TOP_VALUE ((1 << VALUE_BITS) - 1) // maximum value
#define FIRST_QTR (TOP_VALUE / 4 + 1) // first quarter of values maximum value
#define HALF (2 * FIRST_QTR) // first half of values maximum value
#define THIRD_QTR (3 * FIRST_QTR) // third quarter of values maximum value
#define LUT_BITS (FREQ_BITS - 8) // number of bits used to index lookup tables
#define LUT_SIZE (1 << LUT_BITS) // size of the lookup tables
#define LUT_BUFF 4 // number of buffered lookup tables
/** Cumulative frequency tables for block Gilbert-Moore coding.
*/
static const uint16_t cf_tables_1[3][129] = {
{
16384, 16066, 15748, 15431, 15114, 14799, 14485, 14173, 13861, 13552,
13243, 12939, 12635, 12336, 12038, 11745, 11452, 11161, 10870, 10586,
10303, 10027, 9751, 9483, 9215, 8953, 8692, 8440, 8189, 7946,
7704, 7472, 7240, 7008, 6776, 6554, 6333, 6122, 5912, 5711,
5512, 5320, 5128, 4947, 4766, 4595, 4425, 4264, 4104, 3946,
3788, 3640, 3493, 3355, 3218, 3090, 2963, 2842, 2721, 2609,
2498, 2395, 2292, 2196, 2100, 2004, 1908, 1820, 1732, 1651,
1570, 1497, 1424, 1355, 1287, 1223, 1161, 1100, 1044, 988,
938, 888, 839, 790, 746, 702, 662, 623, 588, 553,
520, 488, 459, 431, 405, 380, 357, 334, 311, 288,
268, 248, 230, 213, 197, 182, 168, 154, 142, 130,
119, 108, 99, 90, 81, 72, 64, 56, 49, 42,
36, 30, 25, 20, 15, 11, 7, 3, 0
},
{
16384, 16080, 15776, 15473, 15170, 14868, 14567, 14268, 13970, 13674,
13378, 13086, 12794, 12505, 12218, 11936, 11654, 11373, 11092, 10818,
10544, 10276, 10008, 9749, 9490, 9236, 8982, 8737, 8492, 8256,
8020, 7792, 7564, 7336, 7108, 6888, 6669, 6459, 6249, 6050,
5852, 5660, 5468, 5286, 5104, 4931, 4760, 4598, 4436, 4275,
4115, 3965, 3816, 3674, 3534, 3403, 3272, 3147, 3023, 2907,
2792, 2684, 2577, 2476, 2375, 2274, 2173, 2079, 1986, 1897,
1810, 1724, 1645, 1567, 1493, 1419, 1351, 1284, 1222, 1161,
1105, 1050, 995, 941, 891, 842, 797, 753, 713, 673,
636, 599, 566, 533, 503, 473, 446, 419, 392, 365,
340, 316, 294, 272, 253, 234, 216, 199, 184, 169,
155, 142, 130, 118, 106, 95, 85, 75, 66, 57,
49, 41, 34, 27, 21, 15, 10, 5, 0
},
{
16384, 16092, 15801, 15510, 15219, 14930, 14641, 14355, 14069, 13785,
13501, 13219, 12938, 12661, 12384, 12112, 11841, 11571, 11301, 11037,
10773, 10514, 10256, 10005, 9754, 9508, 9263, 9025, 8787, 8557,
8327, 8103, 7879, 7655, 7431, 7215, 7000, 6792, 6585, 6387,
6190, 5998, 5807, 5625, 5445, 5272, 5100, 4937, 4774, 4613,
4452, 4301, 4150, 4007, 3865, 3731, 3597, 3469, 3341, 3218,
3099, 2981, 2869, 2758, 2652, 2546, 2440, 2334, 2234, 2134,
2041, 1949, 1864, 1779, 1699, 1620, 1547, 1474, 1407, 1340,
1278, 1217, 1157, 1097, 1043, 989, 940, 891, 846, 801,
759, 718, 680, 643, 609, 575, 543, 511, 479, 447,
418, 389, 363, 337, 314, 291, 270, 249, 230, 212,
195, 179, 164, 149, 135, 121, 108, 96, 85, 74,
64, 54, 45, 36, 28, 20, 13, 6, 0
}
};
static const uint16_t cf_tables_2[8][193] = {
{
16384, 16104, 15825, 15546, 15268, 14991, 14714, 14439, 14164, 13891,
13620, 13350, 13081, 12815, 12549, 12287, 12025, 11765, 11505, 11250,
10996, 10746, 10497, 10254, 10011, 9772, 9534, 9303, 9072, 8848,
8624, 8406, 8188, 7970, 7752, 7539, 7327, 7123, 6919, 6724,
6529, 6339, 6150, 5970, 5790, 5618, 5446, 5282, 5119, 4957,
4795, 4642, 4490, 4345, 4201, 4065, 3929, 3798, 3669, 3547,
3425, 3310, 3196, 3086, 2976, 2866, 2756, 2650, 2545, 2447,
2350, 2260, 2170, 2085, 2000, 1921, 1843, 1770, 1698, 1632,
1566, 1501, 1436, 1376, 1316, 1261, 1207, 1157, 1108, 1061,
1015, 973, 931, 893, 855, 819, 783, 747, 711, 677,
644, 614, 584, 557, 530, 505, 480, 458, 436, 416,
396, 378, 360, 343, 326, 310, 295, 281, 267, 255,
243, 232, 221, 211, 201, 192, 183, 174, 166, 158,
150, 142, 134, 126, 119, 112, 106, 100, 95, 90,
85, 80, 76, 72, 69, 66, 63, 60, 57, 54,
51, 48, 46, 44, 42, 40, 38, 36, 34, 33,
32, 31, 30, 29, 28, 27, 26, 25, 24, 23,
22, 21, 20, 19, 18, 17, 16, 15, 14, 13,
12, 11, 10, 9, 8, 7, 6, 5, 4, 3,
2, 1, 0
},
{
16384, 16116, 15849, 15582, 15316, 15050, 14785, 14521, 14257, 13995,
13734, 13476, 13218, 12963, 12708, 12457, 12206, 11956, 11706, 11460,
11215, 10975, 10735, 10500, 10265, 10034, 9803, 9579, 9355, 9136,
8917, 8703, 8489, 8275, 8061, 7853, 7645, 7444, 7244, 7051,
6858, 6671, 6484, 6305, 6127, 5956, 5785, 5622, 5459, 5298,
5137, 4983, 4830, 4684, 4539, 4401, 4263, 4131, 3999, 3874,
3750, 3632, 3515, 3401, 3287, 3173, 3059, 2949, 2840, 2737,
2635, 2539, 2444, 2354, 2264, 2181, 2098, 2020, 1943, 1872,
1801, 1731, 1661, 1596, 1532, 1472, 1412, 1357, 1303, 1251,
1200, 1153, 1106, 1063, 1020, 979, 938, 897, 856, 818,
780, 746, 712, 681, 650, 621, 592, 566, 540, 517,
494, 473, 452, 431, 410, 391, 373, 356, 340, 325,
310, 296, 282, 270, 258, 247, 236, 225, 214, 203,
192, 182, 172, 162, 153, 144, 136, 128, 121, 114,
108, 102, 97, 92, 87, 82, 77, 73, 69, 65,
62, 59, 56, 53, 50, 47, 45, 43, 41, 39,
37, 35, 33, 31, 29, 27, 26, 25, 24, 23,
22, 21, 20, 19, 18, 17, 16, 15, 14, 13,
12, 11, 10, 9, 8, 7, 6, 5, 4, 3,
2, 1, 0
},
{
16384, 16128, 15872, 15617, 15362, 15107, 14853, 14600, 14347, 14096,
13846, 13597, 13350, 13105, 12860, 12618, 12376, 12135, 11894, 11657,
11421, 11189, 10957, 10730, 10503, 10279, 10056, 9838, 9620, 9407,
9195, 8987, 8779, 8571, 8363, 8159, 7955, 7758, 7561, 7371,
7182, 6997, 6812, 6635, 6459, 6289, 6120, 5957, 5795, 5634,
5473, 5319, 5165, 5018, 4871, 4732, 4593, 4458, 4324, 4197,
4071, 3951, 3831, 3714, 3597, 3480, 3363, 3250, 3138, 3032,
2927, 2828, 2729, 2635, 2541, 2453, 2366, 2284, 2202, 2126,
2050, 1975, 1900, 1830, 1761, 1697, 1633, 1574, 1515, 1459,
1403, 1351, 1300, 1252, 1205, 1160, 1115, 1070, 1025, 982,
939, 899, 860, 824, 789, 756, 723, 693, 663, 636,
609, 584, 559, 535, 511, 489, 467, 447, 427, 409,
391, 374, 358, 343, 328, 313, 300, 287, 274, 261,
248, 235, 223, 211, 200, 189, 179, 169, 160, 151,
143, 135, 128, 121, 115, 109, 103, 97, 92, 87,
82, 77, 73, 69, 65, 61, 58, 55, 52, 49,
46, 43, 40, 37, 35, 33, 31, 29, 27, 25,
23, 21, 20, 19, 18, 17, 16, 15, 14, 13,
12, 11, 10, 9, 8, 7, 6, 5, 4, 3,
2, 1, 0
},
{
16384, 16139, 15894, 15649, 15405, 15162, 14919, 14677, 14435, 14195,
13955, 13717, 13479, 13243, 13008, 12775, 12542, 12310, 12079, 11851,
11623, 11399, 11176, 10956, 10737, 10521, 10305, 10094, 9883, 9677,
9471, 9268, 9065, 8862, 8659, 8459, 8260, 8067, 7874, 7688,
7502, 7321, 7140, 6965, 6790, 6621, 6452, 6290, 6128, 5968,
5808, 5655, 5503, 5356, 5209, 5069, 4929, 4794, 4660, 4532,
4404, 4282, 4160, 4041, 3922, 3803, 3684, 3568, 3452, 3343,
3234, 3131, 3029, 2931, 2833, 2741, 2649, 2563, 2477, 2396,
2316, 2236, 2157, 2083, 2009, 1940, 1871, 1807, 1743, 1683,
1623, 1567, 1511, 1459, 1407, 1357, 1307, 1257, 1207, 1159,
1111, 1067, 1023, 983, 943, 905, 868, 834, 800, 769,
738, 709, 681, 653, 625, 600, 575, 552, 529, 508,
487, 466, 447, 428, 410, 392, 376, 360, 344, 328,
313, 298, 283, 268, 255, 242, 230, 218, 207, 196,
186, 176, 167, 158, 150, 142, 135, 128, 121, 114,
108, 102, 97, 92, 87, 82, 78, 74, 70, 66,
62, 58, 54, 50, 47, 44, 41, 38, 35, 32,
30, 28, 26, 24, 22, 20, 18, 16, 14, 13,
12, 11, 10, 9, 8, 7, 6, 5, 4, 3,
2, 1, 0
},
{
16384, 16149, 15915, 15681, 15447, 15214, 14981, 14749, 14517, 14286,
14055, 13827, 13599, 13373, 13147, 12923, 12699, 12476, 12253, 12034,
11815, 11599, 11383, 11171, 10959, 10750, 10541, 10337, 10133, 9933,
9733, 9536, 9339, 9142, 8945, 8751, 8557, 8369, 8181, 7998,
7816, 7638, 7460, 7288, 7116, 6950, 6785, 6625, 6465, 6306,
6147, 5995, 5843, 5697, 5551, 5411, 5271, 5135, 5000, 4871,
4742, 4618, 4495, 4374, 4253, 4132, 4011, 3893, 3775, 3663,
3552, 3446, 3340, 3239, 3138, 3043, 2948, 2858, 2768, 2684,
2600, 2516, 2433, 2355, 2278, 2205, 2133, 2065, 1997, 1932,
1867, 1807, 1747, 1690, 1634, 1580, 1526, 1472, 1418, 1366,
1314, 1266, 1218, 1174, 1130, 1088, 1047, 1009, 971, 936,
901, 868, 836, 804, 772, 743, 714, 685, 658, 631,
606, 582, 559, 536, 515, 494, 475, 456, 437, 418,
399, 380, 362, 344, 328, 312, 297, 283, 270, 257,
245, 233, 222, 211, 201, 191, 181, 172, 163, 155,
147, 139, 132, 125, 119, 113, 107, 101, 96, 91,
86, 81, 76, 71, 66, 62, 58, 54, 50, 46,
43, 40, 37, 34, 31, 28, 26, 24, 22, 20,
18, 16, 14, 12, 10, 8, 6, 5, 4, 3,
2, 1, 0
},
{
16384, 16159, 15934, 15709, 15485, 15261, 15038, 14816, 14594, 14373,
14152, 13933, 13714, 13497, 13280, 13065, 12850, 12636, 12422, 12211,
12000, 11791, 11583, 11378, 11173, 10971, 10769, 10571, 10373, 10179,
9985, 9793, 9601, 9409, 9217, 9029, 8842, 8658, 8475, 8297,
8120, 7946, 7773, 7604, 7435, 7271, 7108, 6950, 6792, 6634,
6477, 6326, 6175, 6029, 5883, 5742, 5602, 5466, 5330, 5199,
5068, 4943, 4818, 4696, 4574, 4452, 4330, 4211, 4093, 3979,
3866, 3759, 3652, 3549, 3446, 3348, 3250, 3157, 3065, 2977,
2889, 2802, 2716, 2634, 2553, 2476, 2399, 2326, 2254, 2185,
2117, 2052, 1987, 1926, 1866, 1808, 1750, 1692, 1634, 1578,
1522, 1470, 1418, 1369, 1321, 1275, 1229, 1187, 1145, 1105,
1066, 1027, 991, 955, 919, 883, 850, 817, 786, 756,
728, 700, 674, 648, 624, 600, 578, 556, 534, 512,
490, 468, 447, 426, 407, 388, 371, 354, 338, 322,
307, 293, 280, 267, 255, 243, 231, 219, 209, 199,
189, 179, 170, 161, 153, 145, 138, 131, 124, 117,
111, 105, 99, 93, 87, 81, 76, 71, 66, 61,
57, 53, 49, 45, 42, 39, 36, 33, 30, 27,
24, 21, 19, 17, 15, 13, 11, 9, 7, 5,
3, 1, 0
},
{
16384, 16169, 15954, 15739, 15524, 15310, 15096, 14883, 14670, 14458,
14246, 14035, 13824, 13614, 13405, 13198, 12991, 12785, 12579, 12376,
12173, 11972, 11772, 11574, 11377, 11182, 10987, 10795, 10603, 10414,
10226, 10040, 9854, 9668, 9482, 9299, 9116, 8937, 8759, 8585,
8411, 8241, 8071, 7906, 7741, 7580, 7419, 7263, 7107, 6952,
6797, 6647, 6497, 6353, 6209, 6070, 5931, 5796, 5661, 5531,
5401, 5275, 5150, 5027, 4904, 4781, 4658, 4538, 4419, 4304,
4190, 4081, 3972, 3867, 3762, 3662, 3562, 3467, 3372, 3281,
3191, 3101, 3012, 2928, 2844, 2764, 2684, 2608, 2533, 2460,
2387, 2318, 2250, 2185, 2121, 2059, 1997, 1935, 1873, 1813,
1754, 1698, 1642, 1588, 1535, 1483, 1433, 1384, 1338, 1292,
1249, 1206, 1165, 1125, 1085, 1045, 1008, 971, 937, 903,
871, 840, 810, 780, 752, 724, 698, 672, 647, 622,
597, 572, 548, 524, 502, 480, 460, 440, 421, 403,
386, 369, 353, 337, 323, 309, 295, 281, 268, 255,
243, 231, 220, 209, 199, 189, 180, 171, 163, 155,
147, 139, 131, 123, 116, 109, 102, 95, 89, 83,
77, 72, 67, 62, 57, 52, 48, 44, 40, 36,
32, 28, 25, 22, 19, 16, 13, 10, 8, 6,
4, 2, 0
},
{
16384, 16177, 15970, 15764, 15558, 15353, 15148, 14944, 14740, 14537,
14334, 14132, 13930, 13729, 13529, 13330, 13131, 12933, 12735, 12539,
12343, 12150, 11957, 11766, 11576, 11388, 11200, 11015, 10830, 10647,
10465, 10285, 10105, 9925, 9745, 9568, 9391, 9218, 9045, 8876,
8707, 8541, 8375, 8213, 8051, 7894, 7737, 7583, 7429, 7277,
7125, 6977, 6830, 6687, 6544, 6406, 6268, 6133, 5998, 5868,
5738, 5612, 5487, 5364, 5241, 5118, 4995, 4875, 4755, 4640,
4525, 4414, 4304, 4198, 4092, 3990, 3888, 3790, 3693, 3600,
3507, 3415, 3323, 3235, 3147, 3064, 2981, 2902, 2823, 2746,
2670, 2594, 2522, 2450, 2382, 2314, 2248, 2182, 2116, 2050,
1987, 1924, 1864, 1804, 1748, 1692, 1638, 1585, 1534, 1484,
1437, 1390, 1346, 1302, 1258, 1215, 1174, 1133, 1095, 1057,
1021, 986, 952, 918, 887, 856, 827, 798, 770, 742,
714, 686, 659, 632, 607, 582, 559, 536, 514, 492,
472, 452, 433, 415, 398, 381, 364, 348, 333, 318,
304, 290, 277, 264, 252, 240, 229, 218, 208, 198,
188, 178, 168, 158, 149, 140, 132, 124, 116, 108,
101, 94, 87, 81, 75, 69, 64, 59, 54, 49,
44, 39, 35, 31, 27, 23, 19, 15, 12, 9,
6, 3, 0
}
};
static const uint16_t cf_tables_3[5][257] = {
{
16384, 16187, 15990, 15793, 15597, 15401, 15205, 15009, 14813, 14618,
14423, 14230, 14037, 13845, 13653, 13463, 13273, 13083, 12894, 12706,
12518, 12332, 12146, 11962, 11778, 11597, 11416, 11237, 11059, 10882,
10706, 10532, 10358, 10184, 10010, 9838, 9666, 9497, 9328, 9163,
8999, 8837, 8675, 8517, 8359, 8205, 8051, 7901, 7751, 7602,
7453, 7308, 7163, 7022, 6882, 6745, 6609, 6476, 6343, 6214,
6085, 5960, 5835, 5712, 5589, 5466, 5343, 5223, 5103, 4987,
4872, 4761, 4650, 4542, 4435, 4332, 4229, 4130, 4031, 3936,
3841, 3747, 3653, 3563, 3473, 3387, 3302, 3220, 3138, 3059,
2980, 2905, 2830, 2759, 2688, 2619, 2550, 2481, 2412, 2345,
2278, 2215, 2152, 2092, 2032, 1974, 1917, 1863, 1809, 1758,
1707, 1659, 1611, 1564, 1517, 1473, 1429, 1387, 1346, 1307,
1268, 1230, 1193, 1158, 1123, 1090, 1058, 1026, 994, 962,
930, 899, 869, 841, 813, 786, 760, 735, 710, 687,
664, 643, 622, 602, 582, 562, 543, 525, 507, 490,
473, 457, 442, 427, 412, 398, 385, 373, 361, 349,
337, 325, 313, 301, 290, 279, 269, 259, 249, 240,
231, 222, 214, 206, 199, 192, 185, 178, 171, 165,
159, 153, 148, 143, 138, 133, 128, 123, 119, 115,
111, 107, 103, 99, 95, 91, 87, 83, 80, 77,
74, 71, 68, 65, 63, 61, 59, 57, 55, 53,
51, 49, 47, 45, 43, 41, 40, 39, 38, 37,
36, 35, 34, 33, 32, 31, 30, 29, 28, 27,
26, 25, 24, 23, 22, 21, 20, 19, 18, 17,
16, 15, 14, 13, 12, 11, 10, 9, 8, 7,
6, 5, 4, 3, 2, 1, 0
},
{
16384, 16195, 16006, 15817, 15629, 15441, 15253, 15065, 14878, 14692,
14506, 14321, 14136, 13952, 13768, 13585, 13402, 13219, 13037, 12857,
12677, 12499, 12321, 12144, 11967, 11792, 11617, 11444, 11271, 11100,
10930, 10762, 10594, 10426, 10258, 10091, 9925, 9761, 9598, 9438,
9278, 9120, 8963, 8809, 8655, 8504, 8354, 8207, 8060, 7914,
7769, 7627, 7485, 7347, 7209, 7074, 6939, 6807, 6676, 6548,
6420, 6296, 6172, 6050, 5928, 5806, 5684, 5564, 5444, 5328,
5212, 5100, 4988, 4879, 4771, 4667, 4563, 4462, 4362, 4265,
4169, 4073, 3978, 3886, 3795, 3707, 3619, 3535, 3451, 3369,
3288, 3210, 3133, 3059, 2985, 2913, 2841, 2769, 2697, 2627,
2557, 2490, 2424, 2360, 2297, 2237, 2177, 2119, 2062, 2007,
1953, 1901, 1849, 1798, 1748, 1700, 1652, 1607, 1562, 1519,
1476, 1435, 1394, 1355, 1317, 1281, 1245, 1210, 1175, 1140,
1105, 1071, 1037, 1005, 973, 943, 913, 885, 857, 830,
804, 779, 754, 731, 708, 685, 663, 642, 621, 601,
581, 563, 545, 528, 511, 495, 479, 463, 448, 433,
419, 405, 391, 377, 364, 351, 338, 326, 314, 302,
291, 280, 270, 260, 251, 242, 234, 226, 218, 210,
202, 195, 188, 181, 174, 168, 162, 156, 150, 144,
139, 134, 129, 124, 119, 114, 109, 104, 100, 96,
92, 88, 84, 80, 77, 74, 71, 68, 65, 62,
59, 56, 54, 52, 50, 48, 46, 44, 42, 40,
38, 36, 34, 33, 32, 31, 30, 29, 28, 27,
26, 25, 24, 23, 22, 21, 20, 19, 18, 17,
16, 15, 14, 13, 12, 11, 10, 9, 8, 7,
6, 5, 4, 3, 2, 1, 0
},
{
16384, 16203, 16022, 15842, 15662, 15482, 15302, 15122, 14942, 14763,
14584, 14406, 14228, 14051, 13874, 13698, 13522, 13347, 13172, 12998,
12824, 12652, 12480, 12310, 12140, 11971, 11803, 11637, 11471, 11307,
11143, 10980, 10817, 10654, 10491, 10330, 10169, 10011, 9853, 9697,
9542, 9389, 9236, 9086, 8936, 8789, 8642, 8498, 8355, 8212,
8070, 7931, 7792, 7656, 7520, 7388, 7256, 7126, 6996, 6870,
6744, 6621, 6498, 6377, 6256, 6135, 6014, 5895, 5776, 5660,
5545, 5433, 5321, 5212, 5104, 4999, 4895, 4793, 4692, 4594,
4496, 4400, 4304, 4211, 4118, 4028, 3939, 3853, 3767, 3684,
3601, 3521, 3441, 3364, 3287, 3212, 3137, 3062, 2987, 2915,
2843, 2773, 2704, 2638, 2572, 2508, 2445, 2384, 2324, 2266,
2208, 2153, 2098, 2044, 1990, 1939, 1888, 1839, 1791, 1745,
1699, 1655, 1611, 1569, 1527, 1487, 1448, 1409, 1370, 1331,
1292, 1255, 1218, 1183, 1148, 1115, 1082, 1051, 1020, 990,
960, 932, 904, 878, 852, 826, 801, 777, 753, 731,
709, 687, 666, 645, 625, 605, 586, 567, 550, 533,
516, 499, 482, 465, 449, 433, 418, 403, 389, 375,
362, 349, 337, 325, 314, 303, 293, 283, 273, 263,
254, 245, 236, 227, 219, 211, 204, 197, 190, 183,
177, 171, 165, 159, 153, 147, 141, 135, 130, 125,
120, 115, 110, 105, 101, 97, 93, 89, 85, 81,
77, 74, 71, 68, 65, 62, 59, 56, 53, 51,
49, 47, 45, 43, 41, 39, 37, 35, 33, 31,
29, 27, 25, 23, 22, 21, 20, 19, 18, 17,
16, 15, 14, 13, 12, 11, 10, 9, 8, 7,
6, 5, 4, 3, 2, 1, 0
},
{
16384, 16210, 16036, 15863, 15690, 15517, 15344, 15172, 15000, 14828,
14656, 14485, 14314, 14145, 13976, 13808, 13640, 13472, 13304, 13137,
12970, 12804, 12639, 12475, 12312, 12149, 11987, 11827, 11667, 11508,
11349, 11192, 11035, 10878, 10721, 10565, 10410, 10257, 10104, 9953,
9802, 9654, 9506, 9359, 9213, 9070, 8927, 8787, 8647, 8508,
8369, 8233, 8097, 7964, 7831, 7700, 7570, 7442, 7315, 7190,
7065, 6943, 6821, 6701, 6581, 6461, 6341, 6223, 6105, 5990,
5876, 5764, 5653, 5545, 5437, 5331, 5226, 5124, 5022, 4924,
4826, 4729, 4632, 4538, 4444, 4353, 4262, 4174, 4087, 4002,
3917, 3835, 3753, 3674, 3595, 3518, 3441, 3364, 3287, 3212,
3138, 3066, 2995, 2926, 2858, 2792, 2726, 2662, 2599, 2538,
2478, 2420, 2362, 2305, 2249, 2195, 2141, 2089, 2037, 1988,
1939, 1891, 1844, 1799, 1754, 1711, 1668, 1626, 1584, 1542,
1500, 1459, 1418, 1380, 1342, 1305, 1269, 1234, 1199, 1166,
1133, 1102, 1071, 1041, 1012, 983, 954, 926, 899, 872,
847, 822, 798, 774, 751, 728, 707, 686, 666, 646,
627, 608, 589, 570, 552, 534, 517, 500, 484, 468,
453, 438, 424, 410, 397, 384, 372, 360, 348, 336,
325, 314, 303, 293, 283, 273, 264, 255, 246, 237,
229, 221, 213, 205, 197, 189, 181, 174, 167, 160,
154, 148, 142, 136, 131, 126, 121, 116, 111, 106,
101, 97, 93, 89, 85, 81, 77, 73, 70, 67,
64, 61, 58, 55, 52, 49, 46, 43, 40, 37,
35, 33, 31, 29, 27, 25, 23, 21, 19, 17,
16, 15, 14, 13, 12, 11, 10, 9, 8, 7,
6, 5, 4, 3, 2, 1, 0
},
{
16384, 16218, 16052, 15886, 15720, 15554, 15389, 15224, 15059, 14895,
14731, 14567, 14403, 14240, 14077, 13915, 13753, 13591, 13429, 13269,
13109, 12950, 12791, 12633, 12476, 12320, 12164, 12009, 11854, 11701,
11548, 11396, 11244, 11092, 10940, 10790, 10640, 10492, 10344, 10198,
10052, 9908, 9764, 9622, 9481, 9342, 9203, 9066, 8929, 8793,
8657, 8524, 8391, 8261, 8131, 8003, 7875, 7749, 7624, 7502,
7380, 7260, 7140, 7022, 6904, 6786, 6668, 6551, 6435, 6322,
6209, 6099, 5989, 5881, 5773, 5668, 5563, 5461, 5359, 5260,
5161, 5063, 4965, 4871, 4777, 4686, 4595, 4506, 4417, 4331,
4245, 4162, 4079, 3999, 3919, 3841, 3763, 3685, 3607, 3530,
3454, 3380, 3307, 3236, 3166, 3097, 3029, 2963, 2897, 2834,
2771, 2710, 2650, 2591, 2532, 2475, 2418, 2363, 2309, 2257,
2205, 2155, 2105, 2057, 2009, 1963, 1918, 1873, 1828, 1783,
1738, 1694, 1650, 1607, 1565, 1524, 1484, 1445, 1407, 1369,
1333, 1297, 1263, 1229, 1197, 1165, 1134, 1103, 1073, 1043,
1015, 987, 960, 933, 907, 882, 858, 834, 811, 788,
766, 744, 722, 700, 679, 658, 638, 618, 599, 581,
563, 545, 528, 511, 495, 480, 465, 451, 437, 423,
410, 397, 384, 372, 360, 348, 337, 326, 315, 305,
295, 285, 275, 265, 255, 245, 236, 227, 219, 211,
203, 195, 188, 181, 174, 167, 161, 155, 149, 143,
137, 131, 126, 121, 116, 111, 106, 101, 97, 93,
89, 85, 81, 77, 73, 69, 65, 61, 58, 55,
52, 49, 46, 43, 40, 37, 34, 32, 30, 28,
26, 24, 22, 20, 18, 16, 14, 12, 10, 8,
6, 5, 4, 3, 2, 1, 0
}
};
static const uint16_t * const cf_table[16] = {
cf_tables_1[0], cf_tables_1[1], cf_tables_1[2], cf_tables_2[0],
cf_tables_2[1], cf_tables_2[2], cf_tables_2[3], cf_tables_2[4],
cf_tables_2[5], cf_tables_2[6], cf_tables_2[7], cf_tables_3[0],
cf_tables_3[1], cf_tables_3[2], cf_tables_3[3], cf_tables_3[4]
};
/** Initializes a given lookup table using a given delta
*/
static void bgmc_lut_fillp(uint8_t *lut, unsigned int *lut_status,
unsigned int delta)
{
unsigned int sx, i;
for (sx = 0; sx < 16; sx++)
for (i = 0; i < LUT_SIZE; i++) {
unsigned int target = (i + 1) << (FREQ_BITS - LUT_BITS);
unsigned int symbol = 1 << delta;
while (cf_table[sx][symbol] > target)
symbol += 1 << delta;
*lut++ = symbol >> delta;
}
*lut_status = delta;
}
/** Retunes the index of a suitable lookup table for a given delta
*/
static uint8_t* bgmc_lut_getp(uint8_t *lut, unsigned int *lut_status,
unsigned int delta)
{
unsigned int i = av_clip(delta, 0, LUT_BUFF - 1);
lut += (i * LUT_SIZE) << 4;
if (lut_status[i] != delta)
bgmc_lut_fillp(lut, &lut_status[i], delta);
return lut;
}
/** Initializes the lookup table arrays
*/
int ff_bgmc_init(AVCodecContext *avctx, uint8_t **cf_lut, unsigned int **cf_lut_status)
{
*cf_lut = av_malloc(sizeof(*cf_lut ) * LUT_BUFF * 16 * LUT_SIZE);
*cf_lut_status = av_malloc(sizeof(*cf_lut_status) * LUT_BUFF);
if (!cf_lut || !cf_lut_status) {
ff_bgmc_end(cf_lut, cf_lut_status);
av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
return AVERROR(ENOMEM);
}
return 0;
}
/** Releases the lookup table arrays
*/
void ff_bgmc_end(uint8_t **cf_lut, unsigned int **cf_lut_status)
{
av_freep(cf_lut);
av_freep(cf_lut_status);
}
/** Initializes decoding and reads the first value
*/
void ff_bgmc_decode_init(GetBitContext *gb,
unsigned int *h, unsigned int *l, unsigned int *v)
{
*h = TOP_VALUE;
*l = 0;
*v = get_bits_long(gb, VALUE_BITS);
}
/** Finish decoding
*/
void ff_bgmc_decode_end(GetBitContext *gb)
{
skip_bits_long(gb, -(VALUE_BITS - 2));
}
/** Reads and decodes a block Gilbert-Moore coded symbol
*/
void ff_bgmc_decode(GetBitContext *gb, unsigned int num, int32_t *dst,
unsigned int delta, unsigned int sx,
unsigned int *h, unsigned int *l, unsigned int *v,
uint8_t *cf_lut, unsigned int *cf_lut_status)
{
unsigned int i;
uint8_t *lut = bgmc_lut_getp(cf_lut, cf_lut_status, delta);
// read current state
unsigned int high = *h;
unsigned int low = *l;
unsigned int value = *v;
lut += sx * LUT_SIZE;
// decode num samples
for (i = 0; i < num; i++) {
unsigned int range = high - low + 1;
unsigned int target = (((value - low + 1) << FREQ_BITS) - 1) / range;
unsigned int symbol = lut[target >> (FREQ_BITS - LUT_BITS)] << delta;
while (cf_table[sx][symbol] > target)
symbol += 1 << delta;
symbol = (symbol >> delta) - 1;
high = low + ((range * cf_table[sx][(symbol ) << delta] - (1 << FREQ_BITS)) >> FREQ_BITS);
low = low + ((range * cf_table[sx][(symbol + 1) << delta] ) >> FREQ_BITS);
while (1) {
if (high >= HALF) {
if (low >= HALF) {
value -= HALF;
low -= HALF;
high -= HALF;
} else if (low >= FIRST_QTR && high < THIRD_QTR) {
value -= FIRST_QTR;
low -= FIRST_QTR;
high -= FIRST_QTR;
} else break;
}
low *= 2;
high = 2 * high + 1;
value = 2 * value + get_bits1(gb);
}
*dst++ = symbol;
}
// save current state
*h = high;
*l = low;
*v = value;
}
src/add-ons/media/plugins/ffmpeg/libavcodec/bgmc.h
-56
View File
@@ -1,56 +0,0 @@
/*
* Block Gilbert-Moore decoder
* Copyright (c) 2010 Thilo Borgmann <thilo.borgmann _at_ googlemail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Block Gilbert-Moore decoder header
* @author Thilo Borgmann <thilo.borgmann _at_ googlemail.com>
*/
#ifndef AVCODEC_BGMC_H
#define AVCODEC_BGMC_H
#include "avcodec.h"
#include "get_bits.h"
int ff_bgmc_init(AVCodecContext *avctx, uint8_t **cf_lut, unsigned int **cf_lut_status);
void ff_bgmc_end(uint8_t **cf_lut, unsigned int **cf_lut_status);
void ff_bgmc_decode_init(GetBitContext *gb,
unsigned int *h, unsigned int *l, unsigned int *v);
void ff_bgmc_decode_end(GetBitContext *gb);
void ff_bgmc_decode(GetBitContext *gb, unsigned int num, int32_t *dst,
unsigned int delta, unsigned int sx,
unsigned int *h, unsigned int *l, unsigned int *v,
uint8_t *cf_lut, unsigned int *cf_lut_status);
#endif /* AVCODEC_BGMC_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/bink.c
-1012
View File
File diff suppressed because it is too large Load Diff
src/add-ons/media/plugins/ffmpeg/libavcodec/binkaudio.c
-311
View File
@@ -1,311 +0,0 @@
/*
* Bink Audio decoder
* Copyright (c) 2007-2010 Peter Ross (pross@xvid.org)
* Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Bink Audio decoder
*
* Technical details here:
* http://wiki.multimedia.cx/index.php?title=Bink_Audio
*/
#include "avcodec.h"
#define ALT_BITSTREAM_READER_LE
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
extern const uint16_t ff_wma_critical_freqs[25];
#define MAX_CHANNELS 2
#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
typedef struct {
AVCodecContext *avctx;
GetBitContext gb;
DSPContext dsp;
int first;
int channels;
int frame_len; ///< transform size (samples)
int overlap_len; ///< overlap size (samples)
int block_size;
int num_bands;
unsigned int *bands;
float root;
DECLARE_ALIGNED(16, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
union {
RDFTContext rdft;
DCTContext dct;
} trans;
} BinkAudioContext;
static av_cold int decode_init(AVCodecContext *avctx)
{
BinkAudioContext *s = avctx->priv_data;
int sample_rate = avctx->sample_rate;
int sample_rate_half;
int i;
int frame_len_bits;
s->avctx = avctx;
dsputil_init(&s->dsp, avctx);
/* determine frame length */
if (avctx->sample_rate < 22050) {
frame_len_bits = 9;
} else if (avctx->sample_rate < 44100) {
frame_len_bits = 10;
} else {
frame_len_bits = 11;
}
s->frame_len = 1 << frame_len_bits;
if (s->channels > MAX_CHANNELS) {
av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
return -1;
}
if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
// audio is already interleaved for the RDFT format variant
sample_rate *= avctx->channels;
s->frame_len *= avctx->channels;
s->channels = 1;
if (avctx->channels == 2)
frame_len_bits++;
} else {
s->channels = avctx->channels;
}
s->overlap_len = s->frame_len / 16;
s->block_size = (s->frame_len - s->overlap_len) * s->channels;
sample_rate_half = (sample_rate + 1) / 2;
s->root = 2.0 / sqrt(s->frame_len);
/* calculate number of bands */
for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
break;
s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
if (!s->bands)
return AVERROR(ENOMEM);
/* populate bands data */
s->bands[0] = 1;
for (i = 1; i < s->num_bands; i++)
s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half;
s->bands[s->num_bands] = s->frame_len / 2;
s->first = 1;
avctx->sample_fmt = SAMPLE_FMT_S16;
for (i = 0; i < s->channels; i++)
s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
else
return -1;
return 0;
}
static float get_float(GetBitContext *gb)
{
int power = get_bits(gb, 5);
float f = ldexpf(get_bits_long(gb, 23), power - 23);
if (get_bits1(gb))
f = -f;
return f;
}
static const uint8_t rle_length_tab[16] = {
2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
};
/**
* Decode Bink Audio block
* @param[out] out Output buffer (must contain s->block_size elements)
*/
static void decode_block(BinkAudioContext *s, short *out, int use_dct)
{
int ch, i, j, k;
float q, quant[25];
int width, coeff;
GetBitContext *gb = &s->gb;
if (use_dct)
skip_bits(gb, 2);
for (ch = 0; ch < s->channels; ch++) {
FFTSample *coeffs = s->coeffs_ptr[ch];
q = 0.0f;
coeffs[0] = get_float(gb) * s->root;
coeffs[1] = get_float(gb) * s->root;
for (i = 0; i < s->num_bands; i++) {
/* constant is result of 0.066399999/log10(M_E) */
int value = get_bits(gb, 8);
quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
}
// find band (k)
for (k = 0; s->bands[k] < 1; k++) {
q = quant[k];
}
// parse coefficients
i = 2;
while (i < s->frame_len) {
if (get_bits1(gb)) {
j = i + rle_length_tab[get_bits(gb, 4)] * 8;
} else {
j = i + 8;
}
j = FFMIN(j, s->frame_len);
width = get_bits(gb, 4);
if (width == 0) {
memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
i = j;
while (s->bands[k] * 2 < i)
q = quant[k++];
} else {
while (i < j) {
if (s->bands[k] * 2 == i)
q = quant[k++];
coeff = get_bits(gb, width);
if (coeff) {
if (get_bits1(gb))
coeffs[i] = -q * coeff;
else
coeffs[i] = q * coeff;
} else {
coeffs[i] = 0.0f;
}
i++;
}
}
}
if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
coeffs[0] /= 0.5;
ff_dct_calc (&s->trans.dct, coeffs);
s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
}
else if (CONFIG_BINKAUDIO_RDFT_DECODER)
ff_rdft_calc(&s->trans.rdft, coeffs);
}
if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
for (i = 0; i < s->channels; i++)
for (j = 0; j < s->frame_len; j++)
s->coeffs_ptr[i][j] = 385.0 + s->coeffs_ptr[i][j]*(1.0/32767.0);
}
s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels);
if (!s->first) {
int count = s->overlap_len * s->channels;
int shift = av_log2(count);
for (i = 0; i < count; i++) {
out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
}
}
memcpy(s->previous, out + s->block_size,
s->overlap_len * s->channels * sizeof(*out));
s->first = 0;
}
static av_cold int decode_end(AVCodecContext *avctx)
{
BinkAudioContext * s = avctx->priv_data;
av_freep(&s->bands);
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_end(&s->trans.rdft);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
ff_dct_end(&s->trans.dct);
return 0;
}
static void get_bits_align32(GetBitContext *s)
{
int n = (-get_bits_count(s)) & 31;
if (n) skip_bits(s, n);
}
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
BinkAudioContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
short *samples = data;
short *samples_end = (short*)((uint8_t*)data + *data_size);
int reported_size;
GetBitContext *gb = &s->gb;
init_get_bits(gb, buf, buf_size * 8);
reported_size = get_bits_long(gb, 32);
while (get_bits_count(gb) / 8 < buf_size &&
samples + s->block_size <= samples_end) {
decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
samples += s->block_size;
get_bits_align32(gb);
}
*data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
return buf_size;
}
AVCodec binkaudio_rdft_decoder = {
"binkaudio_rdft",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_BINKAUDIO_RDFT,
sizeof(BinkAudioContext),
decode_init,
NULL,
decode_end,
decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
};
AVCodec binkaudio_dct_decoder = {
"binkaudio_dct",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_BINKAUDIO_DCT,
sizeof(BinkAudioContext),
decode_init,
NULL,
decode_end,
decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
};
src/add-ons/media/plugins/ffmpeg/libavcodec/binkdata.h
-614
View File
@@ -1,614 +0,0 @@
/*
* Bink video decoder
* Copyright (C) 2009 Kostya Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_BINKDATA_H
#define AVCODEC_BINKDATA_H
#include <stdint.h>
/** Bink DCT and residue 8x8 block scan order */
static const uint8_t bink_scan[64] = {
0, 1, 8, 9, 2, 3, 10, 11,
4, 5, 12, 13, 6, 7, 14, 15,
20, 21, 28, 29, 22, 23, 30, 31,
16, 17, 24, 25, 32, 33, 40, 41,
34, 35, 42, 43, 48, 49, 56, 57,
50, 51, 58, 59, 18, 19, 26, 27,
36, 37, 44, 45, 38, 39, 46, 47,
52, 53, 60, 61, 54, 55, 62, 63
};
static const uint8_t bink_tree_bits[16][16] = {
{
0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07,
0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, 0x0F,
},
{
0x00, 0x01, 0x03, 0x05, 0x07, 0x09, 0x0B, 0x0D,
0x0F, 0x13, 0x15, 0x17, 0x19, 0x1B, 0x1D, 0x1F,
},
{
0x00, 0x02, 0x01, 0x09, 0x05, 0x15, 0x0D, 0x1D,
0x03, 0x13, 0x0B, 0x1B, 0x07, 0x17, 0x0F, 0x1F,
},
{
0x00, 0x02, 0x06, 0x01, 0x09, 0x05, 0x0D, 0x1D,
0x03, 0x13, 0x0B, 0x1B, 0x07, 0x17, 0x0F, 0x1F,
},
{
0x00, 0x04, 0x02, 0x06, 0x01, 0x09, 0x05, 0x0D,
0x03, 0x13, 0x0B, 0x1B, 0x07, 0x17, 0x0F, 0x1F,
},
{
0x00, 0x04, 0x02, 0x0A, 0x06, 0x0E, 0x01, 0x09,
0x05, 0x0D, 0x03, 0x0B, 0x07, 0x17, 0x0F, 0x1F,
},
{
0x00, 0x02, 0x0A, 0x06, 0x0E, 0x01, 0x09, 0x05,
0x0D, 0x03, 0x0B, 0x1B, 0x07, 0x17, 0x0F, 0x1F,
},
{
0x00, 0x01, 0x05, 0x03, 0x13, 0x0B, 0x1B, 0x3B,
0x07, 0x27, 0x17, 0x37, 0x0F, 0x2F, 0x1F, 0x3F,
},
{
0x00, 0x01, 0x03, 0x13, 0x0B, 0x2B, 0x1B, 0x3B,
0x07, 0x27, 0x17, 0x37, 0x0F, 0x2F, 0x1F, 0x3F,
},
{
0x00, 0x01, 0x05, 0x0D, 0x03, 0x13, 0x0B, 0x1B,
0x07, 0x27, 0x17, 0x37, 0x0F, 0x2F, 0x1F, 0x3F,
},
{
0x00, 0x02, 0x01, 0x05, 0x0D, 0x03, 0x13, 0x0B,
0x1B, 0x07, 0x17, 0x37, 0x0F, 0x2F, 0x1F, 0x3F,
},
{
0x00, 0x01, 0x09, 0x05, 0x0D, 0x03, 0x13, 0x0B,
0x1B, 0x07, 0x17, 0x37, 0x0F, 0x2F, 0x1F, 0x3F,
},
{
0x00, 0x02, 0x01, 0x03, 0x13, 0x0B, 0x1B, 0x3B,
0x07, 0x27, 0x17, 0x37, 0x0F, 0x2F, 0x1F, 0x3F,
},
{
0x00, 0x01, 0x05, 0x03, 0x07, 0x27, 0x17, 0x37,
0x0F, 0x4F, 0x2F, 0x6F, 0x1F, 0x5F, 0x3F, 0x7F,
},
{
0x00, 0x01, 0x05, 0x03, 0x07, 0x17, 0x37, 0x77,
0x0F, 0x4F, 0x2F, 0x6F, 0x1F, 0x5F, 0x3F, 0x7F,
},
{
0x00, 0x02, 0x01, 0x05, 0x03, 0x07, 0x27, 0x17,
0x37, 0x0F, 0x2F, 0x6F, 0x1F, 0x5F, 0x3F, 0x7F,
},
};
static const uint8_t bink_tree_lens[16][16] = {
{ 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4 },
{ 1, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5 },
{ 2, 2, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5 },
{ 2, 3, 3, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5 },
{ 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5 },
{ 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5 },
{ 2, 4, 4, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5 },
{ 1, 3, 3, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6 },
{ 1, 2, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6 },
{ 1, 3, 4, 4, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6 },
{ 2, 2, 3, 4, 4, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6 },
{ 1, 4, 4, 4, 4, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6 },
{ 2, 2, 2, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6 },
{ 1, 3, 3, 3, 6, 6, 6, 6, 7, 7, 7, 7, 7, 7, 7, 7 },
{ 1, 3, 3, 3, 5, 6, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7 },
{ 2, 2, 3, 3, 3, 6, 6, 6, 6, 6, 7, 7, 7, 7, 7, 7 },
};
static const uint8_t bink_patterns[16][64] = {
{
0x00, 0x08, 0x10, 0x18, 0x20, 0x28, 0x30, 0x38,
0x39, 0x31, 0x29, 0x21, 0x19, 0x11, 0x09, 0x01,
0x02, 0x0A, 0x12, 0x1A, 0x22, 0x2A, 0x32, 0x3A,
0x3B, 0x33, 0x2B, 0x23, 0x1B, 0x13, 0x0B, 0x03,
0x04, 0x0C, 0x14, 0x1C, 0x24, 0x2C, 0x34, 0x3C,
0x3D, 0x35, 0x2D, 0x25, 0x1D, 0x15, 0x0D, 0x05,
0x06, 0x0E, 0x16, 0x1E, 0x26, 0x2E, 0x36, 0x3E,
0x3F, 0x37, 0x2F, 0x27, 0x1F, 0x17, 0x0F, 0x07,
},
{
0x3B, 0x3A, 0x39, 0x38, 0x30, 0x31, 0x32, 0x33,
0x2B, 0x2A, 0x29, 0x28, 0x20, 0x21, 0x22, 0x23,
0x1B, 0x1A, 0x19, 0x18, 0x10, 0x11, 0x12, 0x13,
0x0B, 0x0A, 0x09, 0x08, 0x00, 0x01, 0x02, 0x03,
0x04, 0x05, 0x06, 0x07, 0x0F, 0x0E, 0x0D, 0x0C,
0x14, 0x15, 0x16, 0x17, 0x1F, 0x1E, 0x1D, 0x1C,
0x24, 0x25, 0x26, 0x27, 0x2F, 0x2E, 0x2D, 0x2C,
0x34, 0x35, 0x36, 0x37, 0x3F, 0x3E, 0x3D, 0x3C,
},
{
0x19, 0x11, 0x12, 0x1A, 0x1B, 0x13, 0x0B, 0x03,
0x02, 0x0A, 0x09, 0x01, 0x00, 0x08, 0x10, 0x18,
0x20, 0x28, 0x30, 0x38, 0x39, 0x31, 0x29, 0x2A,
0x32, 0x3A, 0x3B, 0x33, 0x2B, 0x23, 0x22, 0x21,
0x1D, 0x15, 0x16, 0x1E, 0x1F, 0x17, 0x0F, 0x07,
0x06, 0x0E, 0x0D, 0x05, 0x04, 0x0C, 0x14, 0x1C,
0x24, 0x2C, 0x34, 0x3C, 0x3D, 0x35, 0x2D, 0x2E,
0x36, 0x3E, 0x3F, 0x37, 0x2F, 0x27, 0x26, 0x25,
},
{
0x03, 0x0B, 0x02, 0x0A, 0x01, 0x09, 0x00, 0x08,
0x10, 0x18, 0x11, 0x19, 0x12, 0x1A, 0x13, 0x1B,
0x23, 0x2B, 0x22, 0x2A, 0x21, 0x29, 0x20, 0x28,
0x30, 0x38, 0x31, 0x39, 0x32, 0x3A, 0x33, 0x3B,
0x3C, 0x34, 0x3D, 0x35, 0x3E, 0x36, 0x3F, 0x37,
0x2F, 0x27, 0x2E, 0x26, 0x2D, 0x25, 0x2C, 0x24,
0x1C, 0x14, 0x1D, 0x15, 0x1E, 0x16, 0x1F, 0x17,
0x0F, 0x07, 0x0E, 0x06, 0x0D, 0x05, 0x0C, 0x04,
},
{
0x18, 0x19, 0x10, 0x11, 0x08, 0x09, 0x00, 0x01,
0x02, 0x03, 0x0A, 0x0B, 0x12, 0x13, 0x1A, 0x1B,
0x1C, 0x1D, 0x14, 0x15, 0x0C, 0x0D, 0x04, 0x05,
0x06, 0x07, 0x0E, 0x0F, 0x16, 0x17, 0x1E, 0x1F,
0x27, 0x26, 0x2F, 0x2E, 0x37, 0x36, 0x3F, 0x3E,
0x3D, 0x3C, 0x35, 0x34, 0x2D, 0x2C, 0x25, 0x24,
0x23, 0x22, 0x2B, 0x2A, 0x33, 0x32, 0x3B, 0x3A,
0x39, 0x38, 0x31, 0x30, 0x29, 0x28, 0x21, 0x20,
},
{
0x00, 0x01, 0x02, 0x03, 0x08, 0x09, 0x0A, 0x0B,
0x10, 0x11, 0x12, 0x13, 0x18, 0x19, 0x1A, 0x1B,
0x20, 0x21, 0x22, 0x23, 0x28, 0x29, 0x2A, 0x2B,
0x30, 0x31, 0x32, 0x33, 0x38, 0x39, 0x3A, 0x3B,
0x04, 0x05, 0x06, 0x07, 0x0C, 0x0D, 0x0E, 0x0F,
0x14, 0x15, 0x16, 0x17, 0x1C, 0x1D, 0x1E, 0x1F,
0x24, 0x25, 0x26, 0x27, 0x2C, 0x2D, 0x2E, 0x2F,
0x34, 0x35, 0x36, 0x37, 0x3C, 0x3D, 0x3E, 0x3F,
},
{
0x06, 0x07, 0x0F, 0x0E, 0x0D, 0x05, 0x0C, 0x04,
0x03, 0x0B, 0x02, 0x0A, 0x09, 0x01, 0x00, 0x08,
0x10, 0x18, 0x11, 0x19, 0x12, 0x1A, 0x13, 0x1B,
0x14, 0x1C, 0x15, 0x1D, 0x16, 0x1E, 0x17, 0x1F,
0x27, 0x2F, 0x26, 0x2E, 0x25, 0x2D, 0x24, 0x2C,
0x23, 0x2B, 0x22, 0x2A, 0x21, 0x29, 0x20, 0x28,
0x31, 0x30, 0x38, 0x39, 0x3A, 0x32, 0x3B, 0x33,
0x3C, 0x34, 0x3D, 0x35, 0x36, 0x37, 0x3F, 0x3E,
},
{
0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07,
0x0F, 0x0E, 0x0D, 0x0C, 0x0B, 0x0A, 0x09, 0x08,
0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17,
0x1F, 0x1E, 0x1D, 0x1C, 0x1B, 0x1A, 0x19, 0x18,
0x20, 0x21, 0x22, 0x23, 0x24, 0x25, 0x26, 0x27,
0x2F, 0x2E, 0x2D, 0x2C, 0x2B, 0x2A, 0x29, 0x28,
0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37,
0x3F, 0x3E, 0x3D, 0x3C, 0x3B, 0x3A, 0x39, 0x38,
},
{
0x00, 0x08, 0x09, 0x01, 0x02, 0x03, 0x0B, 0x0A,
0x12, 0x13, 0x1B, 0x1A, 0x19, 0x11, 0x10, 0x18,
0x20, 0x28, 0x29, 0x21, 0x22, 0x23, 0x2B, 0x2A,
0x32, 0x31, 0x30, 0x38, 0x39, 0x3A, 0x3B, 0x33,
0x34, 0x3C, 0x3D, 0x3E, 0x3F, 0x37, 0x36, 0x35,
0x2D, 0x2C, 0x24, 0x25, 0x26, 0x2E, 0x2F, 0x27,
0x1F, 0x17, 0x16, 0x1E, 0x1D, 0x1C, 0x14, 0x15,
0x0D, 0x0C, 0x04, 0x05, 0x06, 0x0E, 0x0F, 0x07,
},
{
0x18, 0x19, 0x10, 0x11, 0x08, 0x09, 0x00, 0x01,
0x02, 0x03, 0x0A, 0x0B, 0x12, 0x13, 0x1A, 0x1B,
0x1C, 0x1D, 0x14, 0x15, 0x0C, 0x0D, 0x04, 0x05,
0x06, 0x07, 0x0E, 0x0F, 0x16, 0x17, 0x1E, 0x1F,
0x26, 0x27, 0x2E, 0x2F, 0x36, 0x37, 0x3E, 0x3F,
0x3C, 0x3D, 0x34, 0x35, 0x2C, 0x2D, 0x24, 0x25,
0x22, 0x23, 0x2A, 0x2B, 0x32, 0x33, 0x3A, 0x3B,
0x38, 0x39, 0x30, 0x31, 0x28, 0x29, 0x20, 0x21,
},
{
0x00, 0x08, 0x01, 0x09, 0x02, 0x0A, 0x03, 0x0B,
0x13, 0x1B, 0x12, 0x1A, 0x11, 0x19, 0x10, 0x18,
0x20, 0x28, 0x21, 0x29, 0x22, 0x2A, 0x23, 0x2B,
0x33, 0x3B, 0x32, 0x3A, 0x31, 0x39, 0x30, 0x38,
0x3C, 0x34, 0x3D, 0x35, 0x3E, 0x36, 0x3F, 0x37,
0x2F, 0x27, 0x2E, 0x26, 0x2D, 0x25, 0x2C, 0x24,
0x1F, 0x17, 0x1E, 0x16, 0x1D, 0x15, 0x1C, 0x14,
0x0C, 0x04, 0x0D, 0x05, 0x0E, 0x06, 0x0F, 0x07,
},
{
0x00, 0x08, 0x10, 0x18, 0x19, 0x1A, 0x1B, 0x13,
0x0B, 0x03, 0x02, 0x01, 0x09, 0x11, 0x12, 0x0A,
0x04, 0x0C, 0x14, 0x1C, 0x1D, 0x1E, 0x1F, 0x17,
0x0F, 0x07, 0x06, 0x05, 0x0D, 0x15, 0x16, 0x0E,
0x24, 0x2C, 0x34, 0x3C, 0x3D, 0x3E, 0x3F, 0x37,
0x2F, 0x27, 0x26, 0x25, 0x2D, 0x35, 0x36, 0x2E,
0x20, 0x28, 0x30, 0x38, 0x39, 0x3A, 0x3B, 0x33,
0x2B, 0x23, 0x22, 0x21, 0x29, 0x31, 0x32, 0x2A,
},
{
0x00, 0x08, 0x09, 0x01, 0x02, 0x03, 0x0B, 0x0A,
0x13, 0x1B, 0x1A, 0x12, 0x11, 0x10, 0x18, 0x19,
0x21, 0x20, 0x28, 0x29, 0x2A, 0x22, 0x23, 0x2B,
0x33, 0x3B, 0x3A, 0x32, 0x31, 0x39, 0x38, 0x30,
0x34, 0x3C, 0x3D, 0x35, 0x36, 0x3E, 0x3F, 0x37,
0x2F, 0x27, 0x26, 0x2E, 0x2D, 0x2C, 0x24, 0x25,
0x1D, 0x1C, 0x14, 0x15, 0x16, 0x1E, 0x1F, 0x17,
0x0E, 0x0F, 0x07, 0x06, 0x05, 0x0D, 0x0C, 0x04,
},
{
0x18, 0x10, 0x08, 0x00, 0x01, 0x02, 0x03, 0x0B,
0x13, 0x1B, 0x1A, 0x19, 0x11, 0x0A, 0x09, 0x12,
0x1C, 0x14, 0x0C, 0x04, 0x05, 0x06, 0x07, 0x0F,
0x17, 0x1F, 0x1E, 0x1D, 0x15, 0x0E, 0x0D, 0x16,
0x3C, 0x34, 0x2C, 0x24, 0x25, 0x26, 0x27, 0x2F,
0x37, 0x3F, 0x3E, 0x3D, 0x35, 0x2E, 0x2D, 0x36,
0x38, 0x30, 0x28, 0x20, 0x21, 0x22, 0x23, 0x2B,
0x33, 0x3B, 0x3A, 0x39, 0x31, 0x2A, 0x29, 0x32,
},
{
0x00, 0x08, 0x09, 0x01, 0x02, 0x0A, 0x12, 0x11,
0x10, 0x18, 0x19, 0x1A, 0x1B, 0x13, 0x0B, 0x03,
0x07, 0x06, 0x0E, 0x0F, 0x17, 0x16, 0x15, 0x0D,
0x05, 0x04, 0x0C, 0x14, 0x1C, 0x1D, 0x1E, 0x1F,
0x3F, 0x3E, 0x36, 0x37, 0x2F, 0x2E, 0x2D, 0x35,
0x3D, 0x3C, 0x34, 0x2C, 0x24, 0x25, 0x26, 0x27,
0x38, 0x30, 0x31, 0x39, 0x3A, 0x32, 0x2A, 0x29,
0x28, 0x20, 0x21, 0x22, 0x23, 0x2B, 0x33, 0x3B,
},
{
0x00, 0x01, 0x08, 0x09, 0x10, 0x11, 0x18, 0x19,
0x20, 0x21, 0x28, 0x29, 0x30, 0x31, 0x38, 0x39,
0x3A, 0x3B, 0x32, 0x33, 0x2A, 0x2B, 0x22, 0x23,
0x1A, 0x1B, 0x12, 0x13, 0x0A, 0x0B, 0x02, 0x03,
0x04, 0x05, 0x0C, 0x0D, 0x14, 0x15, 0x1C, 0x1D,
0x24, 0x25, 0x2C, 0x2D, 0x34, 0x35, 0x3C, 0x3D,
0x3E, 0x3F, 0x36, 0x37, 0x2E, 0x2F, 0x26, 0x27,
0x1E, 0x1F, 0x16, 0x17, 0x0E, 0x0F, 0x06, 0x07,
}
};
static const uint32_t bink_intra_quant[16][64] = {
{
0x010000, 0x016315, 0x01E83D, 0x02A535, 0x014E7B, 0x016577, 0x02F1E6, 0x02724C,
0x010000, 0x00EEDA, 0x024102, 0x017F9B, 0x00BE80, 0x00611E, 0x01083C, 0x00A552,
0x021F88, 0x01DC53, 0x027FAD, 0x01F697, 0x014819, 0x00A743, 0x015A31, 0x009688,
0x02346F, 0x030EE5, 0x01FBFA, 0x02C096, 0x01D000, 0x028396, 0x019247, 0x01F9AA,
0x02346F, 0x01FBFA, 0x01DC53, 0x0231B8, 0x012F12, 0x01E06C, 0x00CB10, 0x0119A8,
0x01C48C, 0x019748, 0x014E86, 0x0122AF, 0x02C628, 0x027F20, 0x0297B5, 0x023F32,
0x025000, 0x01AB6B, 0x01D122, 0x0159B3, 0x012669, 0x008D43, 0x00EE1F, 0x0075ED,
0x01490C, 0x010288, 0x00F735, 0x00EF51, 0x00E0F1, 0x0072AD, 0x00A4D8, 0x006517,
},
{
0x015555, 0x01D971, 0x028AFC, 0x0386F1, 0x01BDF9, 0x01DC9F, 0x03ED33, 0x034311,
0x015555, 0x013E78, 0x030158, 0x01FF7A, 0x00FE00, 0x00817D, 0x01604F, 0x00DC6D,
0x02D4B5, 0x027B19, 0x0354E7, 0x029E1F, 0x01B577, 0x00DF04, 0x01CD96, 0x00C8B6,
0x02F095, 0x0413DC, 0x02A54E, 0x03AB73, 0x026AAB, 0x035A1E, 0x02185E, 0x02A238,
0x02F095, 0x02A54E, 0x027B19, 0x02ECF5, 0x019418, 0x028090, 0x010EC0, 0x01778A,
0x025B66, 0x021F0B, 0x01BE09, 0x018394, 0x03B2E0, 0x03542A, 0x0374F1, 0x02FEEE,
0x031555, 0x0239E4, 0x026C2D, 0x01CCEE, 0x01888C, 0x00BC59, 0x013D7E, 0x009D3C,
0x01B6BB, 0x0158B5, 0x01499C, 0x013F17, 0x012BEC, 0x0098E6, 0x00DBCB, 0x0086C9,
},
{
0x01AAAB, 0x024FCE, 0x032DBB, 0x0468AD, 0x022D78, 0x0253C7, 0x04E87F, 0x0413D5,
0x01AAAB, 0x018E16, 0x03C1AE, 0x027F58, 0x013D80, 0x00A1DC, 0x01B863, 0x011388,
0x0389E2, 0x0319DF, 0x042A21, 0x0345A7, 0x0222D4, 0x0116C5, 0x0240FC, 0x00FAE3,
0x03ACBA, 0x0518D3, 0x034EA1, 0x04964F, 0x030555, 0x0430A5, 0x029E76, 0x034AC5,
0x03ACBA, 0x034EA1, 0x0319DF, 0x03A833, 0x01F91E, 0x0320B4, 0x015270, 0x01D56D,
0x02F23F, 0x02A6CE, 0x022D8B, 0x01E479, 0x049F98, 0x042935, 0x04522D, 0x03BEA9,
0x03DAAB, 0x02C85D, 0x030738, 0x02402A, 0x01EAAF, 0x00EB6F, 0x018CDE, 0x00C48A,
0x022469, 0x01AEE2, 0x019C02, 0x018EDD, 0x0176E7, 0x00BF20, 0x0112BE, 0x00A87B,
},
{
0x020000, 0x02C62A, 0x03D07A, 0x054A69, 0x029CF6, 0x02CAEF, 0x05E3CC, 0x04E499,
0x020000, 0x01DDB4, 0x048204, 0x02FF36, 0x017D01, 0x00C23C, 0x021077, 0x014AA3,
0x043F0F, 0x03B8A6, 0x04FF5A, 0x03ED2E, 0x029032, 0x014E86, 0x02B461, 0x012D11,
0x0468DF, 0x061DCA, 0x03F7F5, 0x05812C, 0x03A000, 0x05072C, 0x03248D, 0x03F353,
0x0468DF, 0x03F7F5, 0x03B8A6, 0x046370, 0x025E24, 0x03C0D8, 0x019620, 0x02334F,
0x038919, 0x032E91, 0x029D0D, 0x02455E, 0x058C50, 0x04FE3F, 0x052F69, 0x047E65,
0x04A000, 0x0356D6, 0x03A243, 0x02B365, 0x024CD2, 0x011A85, 0x01DC3E, 0x00EBD9,
0x029218, 0x020510, 0x01EE69, 0x01DEA2, 0x01C1E2, 0x00E559, 0x0149B0, 0x00CA2D,
},
{
0x02AAAB, 0x03B2E3, 0x0515F8, 0x070DE2, 0x037BF2, 0x03B93E, 0x07DA65, 0x068621,
0x02AAAB, 0x027CF0, 0x0602B1, 0x03FEF3, 0x01FC01, 0x0102FA, 0x02C09F, 0x01B8DA,
0x05A96A, 0x04F632, 0x06A9CE, 0x053C3E, 0x036AED, 0x01BE09, 0x039B2D, 0x01916B,
0x05E129, 0x0827B8, 0x054A9C, 0x0756E5, 0x04D555, 0x06B43B, 0x0430BC, 0x05446F,
0x05E129, 0x054A9C, 0x04F632, 0x05D9EB, 0x032830, 0x050121, 0x021D80, 0x02EF14,
0x04B6CC, 0x043E16, 0x037C11, 0x030728, 0x0765C0, 0x06A855, 0x06E9E2, 0x05FDDB,
0x062AAB, 0x0473C8, 0x04D85A, 0x0399DC, 0x031118, 0x0178B2, 0x027AFD, 0x013A77,
0x036D76, 0x02B16A, 0x029337, 0x027E2E, 0x0257D8, 0x0131CC, 0x01B796, 0x010D91,
},
{
0x038000, 0x04DACA, 0x06ACD5, 0x094238, 0x0492AE, 0x04E322, 0x0A4EA5, 0x08900C,
0x038000, 0x0343FB, 0x07E388, 0x053E9F, 0x029AC1, 0x0153E8, 0x039CD0, 0x02429E,
0x076E5B, 0x068322, 0x08BEDE, 0x06DF11, 0x047C57, 0x02496B, 0x04BBAB, 0x020EDD,
0x07B786, 0x0AB421, 0x06F1ED, 0x09A20D, 0x065800, 0x08CC8E, 0x057FF7, 0x06E9D2,
0x07B786, 0x06F1ED, 0x068322, 0x07AE04, 0x0424BF, 0x06917B, 0x02C6B8, 0x03D9CB,
0x062FEB, 0x05917D, 0x0492D7, 0x03F964, 0x09B58C, 0x08BCEF, 0x0912F8, 0x07DD30,
0x081800, 0x05D7F7, 0x065BF6, 0x04B9F1, 0x040670, 0x01EE69, 0x03416C, 0x019CBC,
0x047FAA, 0x0388DC, 0x036138, 0x03459C, 0x03134C, 0x01915C, 0x0240F5, 0x0161CF,
},
{
0x040000, 0x058C54, 0x07A0F4, 0x0A94D3, 0x0539EC, 0x0595DD, 0x0BC798, 0x09C932,
0x040000, 0x03BB68, 0x090409, 0x05FE6D, 0x02FA01, 0x018477, 0x0420EE, 0x029547,
0x087E1F, 0x07714C, 0x09FEB5, 0x07DA5D, 0x052064, 0x029D0D, 0x0568C3, 0x025A21,
0x08D1BE, 0x0C3B94, 0x07EFEA, 0x0B0258, 0x074000, 0x0A0E59, 0x06491A, 0x07E6A7,
0x08D1BE, 0x07EFEA, 0x07714C, 0x08C6E0, 0x04BC48, 0x0781B1, 0x032C3F, 0x04669F,
0x071232, 0x065D22, 0x053A1A, 0x048ABC, 0x0B18A0, 0x09FC7F, 0x0A5ED3, 0x08FCC9,
0x094000, 0x06ADAC, 0x074487, 0x0566CA, 0x0499A5, 0x02350B, 0x03B87B, 0x01D7B3,
0x052430, 0x040A20, 0x03DCD3, 0x03BD45, 0x0383C5, 0x01CAB3, 0x029361, 0x01945A,
},
{
0x050000, 0x06EF69, 0x098931, 0x0D3A07, 0x068867, 0x06FB55, 0x0EB97E, 0x0C3B7E,
0x050000, 0x04AA42, 0x0B450B, 0x077E08, 0x03B881, 0x01E595, 0x05292A, 0x033A99,
0x0A9DA7, 0x094D9F, 0x0C7E62, 0x09D0F4, 0x06687D, 0x034450, 0x06C2F4, 0x02F0AA,
0x0B062D, 0x0F4A78, 0x09EBE4, 0x0DC2EE, 0x091000, 0x0C91EF, 0x07DB61, 0x09E050,
0x0B062D, 0x09EBE4, 0x094D9F, 0x0AF898, 0x05EB59, 0x09621D, 0x03F74F, 0x058046,
0x08D6BE, 0x07F46A, 0x0688A0, 0x05AD6B, 0x0DDEC8, 0x0C7B9F, 0x0CF687, 0x0B3BFB,
0x0B9000, 0x085917, 0x0915A8, 0x06C07D, 0x05C00E, 0x02C24D, 0x04A69A, 0x024D9F,
0x066D3C, 0x050CA7, 0x04D407, 0x04AC96, 0x0464B6, 0x023D5F, 0x033839, 0x01F971,
},
{
0x060000, 0x08527E, 0x0B716E, 0x0FDF3C, 0x07D6E1, 0x0860CC, 0x11AB63, 0x0EADCB,
0x060000, 0x05991C, 0x0D860D, 0x08FDA3, 0x047702, 0x0246B3, 0x063165, 0x03DFEA,
0x0CBD2E, 0x0B29F1, 0x0EFE0F, 0x0BC78B, 0x07B096, 0x03EB93, 0x081D24, 0x038732,
0x0D3A9C, 0x12595D, 0x0BE7DF, 0x108384, 0x0AE000, 0x0F1585, 0x096DA8, 0x0BD9FA,
0x0D3A9C, 0x0BE7DF, 0x0B29F1, 0x0D2A50, 0x071A6B, 0x0B4289, 0x04C25F, 0x0699EE,
0x0A9B4A, 0x098BB2, 0x07D727, 0x06D01A, 0x10A4F0, 0x0EFABE, 0x0F8E3C, 0x0D7B2E,
0x0DE000, 0x0A0482, 0x0AE6CA, 0x081A2F, 0x06E677, 0x034F90, 0x0594B9, 0x02C38C,
0x07B649, 0x060F2F, 0x05CB3C, 0x059BE7, 0x0545A7, 0x02B00C, 0x03DD11, 0x025E87,
},
{
0x080000, 0x0B18A8, 0x0F41E8, 0x1529A5, 0x0A73D7, 0x0B2BBB, 0x178F2F, 0x139264,
0x080000, 0x0776CF, 0x120812, 0x0BFCD9, 0x05F402, 0x0308EF, 0x0841DC, 0x052A8E,
0x10FC3E, 0x0EE297, 0x13FD69, 0x0FB4B9, 0x0A40C8, 0x053A1A, 0x0AD186, 0x04B442,
0x11A37B, 0x187727, 0x0FDFD4, 0x1604B0, 0x0E8000, 0x141CB1, 0x0C9235, 0x0FCD4D,
0x11A37B, 0x0FDFD4, 0x0EE297, 0x118DC0, 0x09788F, 0x0F0362, 0x06587F, 0x08CD3D,
0x0E2463, 0x0CBA43, 0x0A7434, 0x091577, 0x163140, 0x13F8FE, 0x14BDA5, 0x11F992,
0x128000, 0x0D5B58, 0x0E890D, 0x0ACD94, 0x093349, 0x046A15, 0x0770F7, 0x03AF65,
0x0A4861, 0x08143F, 0x07B9A6, 0x077A89, 0x070789, 0x039565, 0x0526C2, 0x0328B4,
},
{
0x0C0000, 0x10A4FD, 0x16E2DB, 0x1FBE78, 0x0FADC3, 0x10C198, 0x2356C7, 0x1D5B96,
0x0C0000, 0x0B3237, 0x1B0C1A, 0x11FB46, 0x08EE03, 0x048D66, 0x0C62CA, 0x07BFD5,
0x197A5D, 0x1653E3, 0x1DFC1E, 0x178F16, 0x0F612C, 0x07D727, 0x103A49, 0x070E64,
0x1A7539, 0x24B2BB, 0x17CFBD, 0x210709, 0x15C000, 0x1E2B0A, 0x12DB4F, 0x17B3F4,
0x1A7539, 0x17CFBD, 0x1653E3, 0x1A54A0, 0x0E34D7, 0x168513, 0x0984BE, 0x0D33DC,
0x153695, 0x131765, 0x0FAE4E, 0x0DA033, 0x2149E1, 0x1DF57D, 0x1F1C78, 0x1AF65B,
0x1BC000, 0x140904, 0x15CD94, 0x10345E, 0x0DCCEE, 0x069F20, 0x0B2972, 0x058718,
0x0F6C91, 0x0C1E5E, 0x0B9678, 0x0B37CE, 0x0A8B4E, 0x056018, 0x07BA22, 0x04BD0E,
},
{
0x110000, 0x179466, 0x206C0C, 0x2CF87F, 0x16362A, 0x17BCED, 0x321044, 0x299714,
0x110000, 0x0FDC79, 0x265125, 0x19794E, 0x0CA685, 0x0672FB, 0x118BF4, 0x0AFA6D,
0x241804, 0x1FA181, 0x2A7A80, 0x21600A, 0x15C9A9, 0x0B1B77, 0x16FD3C, 0x09FF0D,
0x257B66, 0x33FD33, 0x21BBA2, 0x2EC9F7, 0x1ED000, 0x2ABCF9, 0x1AB6B0, 0x219444,
0x257B66, 0x21BBA2, 0x1FA181, 0x254D38, 0x142030, 0x1FE730, 0x0D7C0E, 0x12B423,
0x1E0D52, 0x1B0BCF, 0x1636EE, 0x134D9E, 0x2F28A9, 0x2A711B, 0x2C12FF, 0x263256,
0x275000, 0x1C621B, 0x1EE33C, 0x16F4DB, 0x138CFB, 0x09616E, 0x0FD00C, 0x07D4B7,
0x15D9CE, 0x112B06, 0x106A80, 0x0FE464, 0x0EF004, 0x079D77, 0x0AF25B, 0x06B67F,
},
{
0x160000, 0x1E83CF, 0x29F53D, 0x3A3286, 0x1CBE90, 0x1EB842, 0x40C9C2, 0x35D293,
0x160000, 0x1486BA, 0x319630, 0x20F756, 0x105F06, 0x085891, 0x16B51E, 0x0E3506,
0x2EB5AA, 0x28EF20, 0x36F8E1, 0x2B30FE, 0x1C3225, 0x0E5FC7, 0x1DC030, 0x0CEFB7,
0x308193, 0x4347AC, 0x2BA786, 0x3C8CE5, 0x27E000, 0x374EE7, 0x229212, 0x2B7494,
0x308193, 0x2BA786, 0x28EF20, 0x3045D0, 0x1A0B89, 0x29494D, 0x11735D, 0x183469,
0x26E410, 0x230039, 0x1CBF8F, 0x18FB09, 0x3D0771, 0x36ECBA, 0x390986, 0x316E52,
0x32E000, 0x24BB33, 0x27F8E4, 0x1DB557, 0x194D09, 0x0C23BB, 0x1476A6, 0x0A2256,
0x1C470A, 0x1637AD, 0x153E87, 0x1490FA, 0x1354B9, 0x09DAD6, 0x0E2A94, 0x08AFF0,
},
{
0x1C0000, 0x26D64D, 0x3566AA, 0x4A11C2, 0x249572, 0x27190E, 0x527525, 0x44805E,
0x1C0000, 0x1A1FD6, 0x3F1C3E, 0x29F4F9, 0x14D607, 0x0A9F44, 0x1CE683, 0x1214F0,
0x3B72D9, 0x341911, 0x45F6F0, 0x36F889, 0x23E2BB, 0x124B5B, 0x25DD54, 0x1076E9,
0x3DBC30, 0x55A109, 0x378F64, 0x4D1069, 0x32C000, 0x46646C, 0x2BFFB9, 0x374E8E,
0x3DBC30, 0x378F64, 0x341911, 0x3D7020, 0x2125F5, 0x348BD6, 0x1635BC, 0x1ECE57,
0x317F5B, 0x2C8BEB, 0x2496B6, 0x1FCB22, 0x4DAC61, 0x45E778, 0x4897C2, 0x3EE97F,
0x40C000, 0x2EBFB5, 0x32DFAE, 0x25CF86, 0x203380, 0x0F734B, 0x1A0B5F, 0x0CE5E2,
0x23FD53, 0x1C46DC, 0x1B09C4, 0x1A2CE1, 0x189A60, 0x0C8AE2, 0x1207A5, 0x0B0E77,
},
{
0x220000, 0x2F28CC, 0x40D818, 0x59F0FE, 0x2C6C53, 0x2F79DA, 0x642089, 0x532E29,
0x220000, 0x1FB8F1, 0x4CA24B, 0x32F29C, 0x194D09, 0x0CE5F7, 0x2317E8, 0x15F4DB,
0x483007, 0x3F4303, 0x54F4FF, 0x42C014, 0x2B9351, 0x1636EE, 0x2DFA79, 0x13FE1A,
0x4AF6CC, 0x67FA67, 0x437743, 0x5D93EE, 0x3DA000, 0x5579F1, 0x356D61, 0x432888,
0x4AF6CC, 0x437743, 0x3F4303, 0x4A9A70, 0x284060, 0x3FCE60, 0x1AF81B, 0x256845,
0x3C1AA5, 0x36179D, 0x2C6DDD, 0x269B3C, 0x5E5152, 0x54E237, 0x5825FE, 0x4C64AD,
0x4EA000, 0x38C437, 0x3DC678, 0x2DE9B5, 0x2719F7, 0x12C2DB, 0x1FA018, 0x0FA96E,
0x2BB39B, 0x22560C, 0x20D500, 0x1FC8C8, 0x1DE007, 0x0F3AEE, 0x15E4B7, 0x0D6CFE,
},
{
0x2C0000, 0x3D079E, 0x53EA79, 0x74650C, 0x397D20, 0x3D7083, 0x819383, 0x6BA525,
0x2C0000, 0x290D75, 0x632C61, 0x41EEAC, 0x20BE0C, 0x10B121, 0x2D6A3B, 0x1C6A0C,
0x5D6B54, 0x51DE40, 0x6DF1C2, 0x5661FB, 0x38644B, 0x1CBF8F, 0x3B8060, 0x19DF6D,
0x610326, 0x868F57, 0x574F0B, 0x7919CA, 0x4FC000, 0x6E9DCE, 0x452423, 0x56E928,
0x610326, 0x574F0B, 0x51DE40, 0x608BA0, 0x341713, 0x52929A, 0x22E6BA, 0x3068D2,
0x4DC821, 0x460071, 0x397F1E, 0x31F611, 0x7A0EE2, 0x6DD974, 0x72130C, 0x62DCA3,
0x65C000, 0x497665, 0x4FF1C9, 0x3B6AAE, 0x329A12, 0x184776, 0x28ED4D, 0x1444AC,
0x388E14, 0x2C6F5A, 0x2A7D0F, 0x2921F4, 0x26A973, 0x13B5AD, 0x1C5528, 0x115FDF,
},
};
static const uint32_t bink_inter_quant[16][64] = {
{
0x010000, 0x017946, 0x01A5A9, 0x0248DC, 0x016363, 0x0152A7, 0x0243EC, 0x0209EA,
0x012000, 0x00E248, 0x01BBDA, 0x015CBC, 0x00A486, 0x0053E0, 0x00F036, 0x008095,
0x01B701, 0x016959, 0x01B0B9, 0x0153FD, 0x00F8E7, 0x007EE4, 0x00EA30, 0x007763,
0x01B701, 0x0260EB, 0x019DE9, 0x023E1B, 0x017000, 0x01FE6E, 0x012DB5, 0x01A27B,
0x01E0D1, 0x01B0B9, 0x018A33, 0x01718D, 0x00D87A, 0x014449, 0x007B9A, 0x00AB71,
0x013178, 0x0112EA, 0x00AD08, 0x009BB9, 0x023D97, 0x020437, 0x021CCC, 0x01E6B4,
0x018000, 0x012DB5, 0x0146D9, 0x0100CE, 0x00CFD2, 0x006E5C, 0x00B0E4, 0x005A2D,
0x00E9CC, 0x00B7B1, 0x00846F, 0x006B85, 0x008337, 0x0042E5, 0x004A10, 0x002831,
},
{
0x015555, 0x01F708, 0x023237, 0x030BD0, 0x01D9D9, 0x01C389, 0x03053B, 0x02B7E3,
0x018000, 0x012DB5, 0x024FCE, 0x01D0FA, 0x00DB5D, 0x006FD5, 0x014048, 0x00AB71,
0x024957, 0x01E1CC, 0x0240F7, 0x01C551, 0x014BDE, 0x00A92F, 0x013840, 0x009F2F,
0x024957, 0x032BE4, 0x0227E1, 0x02FD7A, 0x01EAAB, 0x02A893, 0x019247, 0x022DF9,
0x028116, 0x0240F7, 0x020D99, 0x01ECBC, 0x0120A3, 0x01B061, 0x00A4CE, 0x00E497,
0x01974B, 0x016E8E, 0x00E6B5, 0x00CFA2, 0x02FCC9, 0x02B04A, 0x02D110, 0x0288F1,
0x020000, 0x019247, 0x01B3CC, 0x015668, 0x011518, 0x009325, 0x00EBDA, 0x00783D,
0x0137BB, 0x00F4ED, 0x00B093, 0x008F5C, 0x00AEF4, 0x005931, 0x0062BF, 0x003597,
},
{
0x01AAAB, 0x0274CB, 0x02BEC4, 0x03CEC4, 0x02504F, 0x02346C, 0x03C689, 0x0365DC,
0x01E000, 0x017922, 0x02E3C1, 0x024539, 0x011235, 0x008BCA, 0x01905A, 0x00D64D,
0x02DBAD, 0x025A40, 0x02D134, 0x0236A5, 0x019ED6, 0x00D37B, 0x018650, 0x00C6FB,
0x02DBAD, 0x03F6DD, 0x02B1D9, 0x03BCD8, 0x026555, 0x0352B8, 0x01F6D8, 0x02B977,
0x03215C, 0x02D134, 0x029100, 0x0267EB, 0x0168CC, 0x021C7A, 0x00CE01, 0x011DBD,
0x01FD1E, 0x01CA31, 0x012062, 0x01038A, 0x03BBFB, 0x035C5C, 0x038554, 0x032B2D,
0x028000, 0x01F6D8, 0x0220C0, 0x01AC02, 0x015A5E, 0x00B7EF, 0x0126D1, 0x00964C,
0x0185A9, 0x013228, 0x00DCB8, 0x00B333, 0x00DAB2, 0x006F7D, 0x007B6F, 0x0042FC,
},
{
0x020000, 0x02F28D, 0x034B52, 0x0491B8, 0x02C6C5, 0x02A54E, 0x0487D8, 0x0413D5,
0x024000, 0x01C48F, 0x0377B5, 0x02B977, 0x01490C, 0x00A7BF, 0x01E06C, 0x01012A,
0x036E03, 0x02D2B3, 0x036172, 0x02A7FA, 0x01F1CE, 0x00FDC7, 0x01D460, 0x00EEC7,
0x036E03, 0x04C1D6, 0x033BD1, 0x047C37, 0x02E000, 0x03FCDD, 0x025B6A, 0x0344F5,
0x03C1A1, 0x036172, 0x031466, 0x02E31B, 0x01B0F5, 0x028892, 0x00F735, 0x0156E2,
0x0262F1, 0x0225D5, 0x015A10, 0x013772, 0x047B2D, 0x04086E, 0x043998, 0x03CD69,
0x030000, 0x025B6A, 0x028DB3, 0x02019B, 0x019FA3, 0x00DCB8, 0x0161C7, 0x00B45B,
0x01D398, 0x016F63, 0x0108DD, 0x00D70A, 0x01066F, 0x0085C9, 0x00941F, 0x005062,
},
{
0x02AAAB, 0x03EE11, 0x04646D, 0x0617A0, 0x03B3B2, 0x038713, 0x060A75, 0x056FC6,
0x030000, 0x025B6A, 0x049F9B, 0x03A1F4, 0x01B6BB, 0x00DFAA, 0x028090, 0x0156E2,
0x0492AE, 0x03C399, 0x0481ED, 0x038AA2, 0x0297BD, 0x01525F, 0x027080, 0x013E5E,
0x0492AE, 0x0657C8, 0x044FC1, 0x05FAF4, 0x03D555, 0x055126, 0x03248D, 0x045BF2,
0x05022D, 0x0481ED, 0x041B33, 0x03D979, 0x024147, 0x0360C3, 0x01499C, 0x01C92E,
0x032E96, 0x02DD1C, 0x01CD6A, 0x019F43, 0x05F991, 0x056093, 0x05A220, 0x0511E1,
0x040000, 0x03248D, 0x036799, 0x02ACCF, 0x022A2F, 0x01264B, 0x01D7B5, 0x00F079,
0x026F75, 0x01E9D9, 0x016127, 0x011EB8, 0x015DE9, 0x00B262, 0x00C57F, 0x006B2D,
},
{
0x038000, 0x052876, 0x05C3CF, 0x07FF02, 0x04DBD9, 0x04A148, 0x07EDBA, 0x0722B4,
0x03F000, 0x0317FB, 0x06117C, 0x04C491, 0x023FD5, 0x01258F, 0x0348BD, 0x01C209,
0x060085, 0x04F0B9, 0x05EA87, 0x04A5F5, 0x036728, 0x01BC1C, 0x0333A8, 0x01A1DB,
0x060085, 0x085336, 0x05A8AE, 0x07D960, 0x050800, 0x06FA82, 0x041FF9, 0x05B8AE,
0x0692DA, 0x05EA87, 0x0563B2, 0x050D6E, 0x02F5AD, 0x046F00, 0x01B09C, 0x02580C,
0x042D25, 0x03C235, 0x025D9B, 0x022108, 0x07D78F, 0x070EC1, 0x0764CA, 0x06A777,
0x054000, 0x041FF9, 0x0477F9, 0x0382D0, 0x02D75E, 0x018242, 0x026B1D, 0x013B9F,
0x03324A, 0x0282ED, 0x01CF83, 0x017851, 0x01CB42, 0x00EA21, 0x010336, 0x008CAC,
},
{
0x040000, 0x05E519, 0x0696A4, 0x092370, 0x058D8A, 0x054A9C, 0x090FB0, 0x0827AA,
0x048000, 0x03891F, 0x06EF69, 0x0572EE, 0x029218, 0x014F7E, 0x03C0D8, 0x020254,
0x06DC05, 0x05A565, 0x06C2E4, 0x054FF3, 0x03E39B, 0x01FB8E, 0x03A8C0, 0x01DD8D,
0x06DC05, 0x0983AC, 0x0677A2, 0x08F86E, 0x05C000, 0x07F9B9, 0x04B6D4, 0x0689EB,
0x078343, 0x06C2E4, 0x0628CC, 0x05C635, 0x0361EA, 0x051124, 0x01EE69, 0x02ADC5,
0x04C5E1, 0x044BAA, 0x02B41F, 0x026EE5, 0x08F65A, 0x0810DD, 0x087330, 0x079AD1,
0x060000, 0x04B6D4, 0x051B65, 0x040337, 0x033F47, 0x01B970, 0x02C38F, 0x0168B6,
0x03A730, 0x02DEC6, 0x0211BA, 0x01AE14, 0x020CDD, 0x010B93, 0x01283E, 0x00A0C4,
},
{
0x050000, 0x075E60, 0x083C4D, 0x0B6C4C, 0x06F0ED, 0x069D43, 0x0B539C, 0x0A3194,
0x05A000, 0x046B67, 0x08AB44, 0x06CFAA, 0x03369E, 0x01A35E, 0x04B10F, 0x0282E8,
0x089307, 0x070EBF, 0x08739C, 0x06A3F0, 0x04DC82, 0x027A72, 0x0492F0, 0x0254F0,
0x089307, 0x0BE497, 0x08158B, 0x0B3689, 0x073000, 0x09F827, 0x05E489, 0x082C66,
0x096413, 0x08739C, 0x07B2FF, 0x0737C2, 0x043A64, 0x06556D, 0x026A04, 0x035936,
0x05F75A, 0x055E94, 0x036127, 0x030A9E, 0x0B33F1, 0x0A1514, 0x0A8FFC, 0x098186,
0x078000, 0x05E489, 0x06623F, 0x050405, 0x040F19, 0x0227CC, 0x037473, 0x01C2E3,
0x0490FC, 0x039677, 0x029629, 0x021999, 0x029015, 0x014E78, 0x01724E, 0x00C8F5,
},
{
0x060000, 0x08D7A6, 0x09E1F6, 0x0DB528, 0x085450, 0x07EFEA, 0x0D9788, 0x0C3B7E,
0x06C000, 0x054DAE, 0x0A671E, 0x082C66, 0x03DB24, 0x01F73E, 0x05A145, 0x03037D,
0x0A4A08, 0x087818, 0x0A2455, 0x07F7ED, 0x05D569, 0x02F955, 0x057D20, 0x02CC54,
0x0A4A08, 0x0E4582, 0x09B373, 0x0D74A5, 0x08A000, 0x0BF696, 0x07123E, 0x09CEE0,
0x0B44E4, 0x0A2455, 0x093D32, 0x08A950, 0x0512DF, 0x0799B6, 0x02E59E, 0x0404A7,
0x0728D2, 0x06717F, 0x040E2F, 0x03A657, 0x0D7187, 0x0C194B, 0x0CACC8, 0x0B683A,
0x090000, 0x07123E, 0x07A918, 0x0604D2, 0x04DEEA, 0x029629, 0x042556, 0x021D11,
0x057AC8, 0x044E28, 0x031A97, 0x02851E, 0x03134C, 0x01915C, 0x01BC5D, 0x00F126,
},
{
0x080000, 0x0BCA33, 0x0D2D48, 0x1246E0, 0x0B1B15, 0x0A9538, 0x121F5F, 0x104F53,
0x090000, 0x07123E, 0x0DDED2, 0x0AE5DD, 0x052430, 0x029EFD, 0x0781B1, 0x0404A7,
0x0DB80B, 0x0B4ACB, 0x0D85C7, 0x0A9FE7, 0x07C736, 0x03F71D, 0x075180, 0x03BB1A,
0x0DB80B, 0x130757, 0x0CEF44, 0x11F0DC, 0x0B8000, 0x0FF372, 0x096DA8, 0x0D13D6,
0x0F0686, 0x0D85C7, 0x0C5198, 0x0B8C6A, 0x06C3D4, 0x0A2248, 0x03DCD3, 0x055B8A,
0x098BC3, 0x089754, 0x05683E, 0x04DDC9, 0x11ECB4, 0x1021B9, 0x10E661, 0x0F35A3,
0x0C0000, 0x096DA8, 0x0A36CB, 0x08066E, 0x067E8E, 0x0372E1, 0x05871E, 0x02D16B,
0x074E60, 0x05BD8B, 0x042374, 0x035C28, 0x0419BB, 0x021726, 0x02507C, 0x014188,
},
{
0x0C0000, 0x11AF4C, 0x13C3EC, 0x1B6A50, 0x10A89F, 0x0FDFD4, 0x1B2F0F, 0x1876FD,
0x0D8000, 0x0A9B5D, 0x14CE3C, 0x1058CB, 0x07B649, 0x03EE7B, 0x0B4289, 0x0606FB,
0x149410, 0x10F030, 0x1448AB, 0x0FEFDA, 0x0BAAD2, 0x05F2AB, 0x0AFA40, 0x0598A7,
0x149410, 0x1C8B03, 0x1366E6, 0x1AE949, 0x114000, 0x17ED2B, 0x0E247C, 0x139DC1,
0x1689C8, 0x1448AB, 0x127A63, 0x11529F, 0x0A25BE, 0x0F336D, 0x05CB3C, 0x08094E,
0x0E51A4, 0x0CE2FE, 0x081C5D, 0x074CAE, 0x1AE30E, 0x183296, 0x195991, 0x16D074,
0x120000, 0x0E247C, 0x0F5230, 0x0C09A5, 0x09BDD5, 0x052C51, 0x084AAC, 0x043A21,
0x0AF590, 0x089C51, 0x06352E, 0x050A3B, 0x062698, 0x0322B9, 0x0378BA, 0x01E24D,
},
{
0x110000, 0x190DAC, 0x1C0039, 0x26D69C, 0x17998C, 0x167D16, 0x2682AB, 0x22A891,
0x132000, 0x0F06C3, 0x1D797F, 0x172876, 0x0AECE7, 0x0591D9, 0x0FF398, 0x0889E3,
0x1D2717, 0x17FEEF, 0x1CBC47, 0x1693CA, 0x108754, 0x086D1D, 0x0F8D30, 0x07ED98,
0x1D2717, 0x286F9A, 0x1B7C71, 0x261FD3, 0x187000, 0x21E552, 0x140904, 0x1BCA27,
0x1FEDDC, 0x1CBC47, 0x1A2D62, 0x188A62, 0x0E6022, 0x1588DA, 0x083540, 0x0B6284,
0x1448FE, 0x124192, 0x0B7D84, 0x0A574B, 0x2616FF, 0x2247AA, 0x23E98D, 0x2051FA,
0x198000, 0x140904, 0x15B46F, 0x110DAA, 0x0DCCEE, 0x07541E, 0x0BBF1F, 0x05FD04,
0x0F868B, 0x0C32C8, 0x08CB57, 0x0723D4, 0x08B6AD, 0x047130, 0x04EB08, 0x02AB42,
},
{
0x160000, 0x206C0C, 0x243C86, 0x3242E8, 0x1E8A79, 0x1D1A59, 0x31D646, 0x2CDA25,
0x18C000, 0x13722A, 0x2624C3, 0x1DF820, 0x0E2385, 0x073537, 0x14A4A7, 0x0B0CCC,
0x25BA1D, 0x1F0DAE, 0x252FE4, 0x1D37BB, 0x1563D6, 0x0AE78E, 0x142021, 0x0A4288,
0x25BA1D, 0x345430, 0x2391FB, 0x31565C, 0x1FA000, 0x2BDD7A, 0x19ED8D, 0x23F68C,
0x2951EF, 0x252FE4, 0x21E061, 0x1FC224, 0x129A87, 0x1BDE47, 0x0A9F44, 0x0EBBBA,
0x1A4058, 0x17A026, 0x0EDEAB, 0x0D61E9, 0x314AEF, 0x2C5CBE, 0x2E798A, 0x29D380,
0x210000, 0x19ED8D, 0x1C16AE, 0x1611AE, 0x11DC06, 0x097BEA, 0x0F3391, 0x07BFE7,
0x141787, 0x0FC93E, 0x0B617F, 0x093D6D, 0x0B46C1, 0x05BFA8, 0x065D55, 0x037437,
},
{
0x1C0000, 0x2943B2, 0x2E1E7C, 0x3FF810, 0x26DEC9, 0x250A43, 0x3F6DCE, 0x3915A3,
0x1F8000, 0x18BFD8, 0x308BE1, 0x262485, 0x11FEA9, 0x092C75, 0x1A45EB, 0x0E1049,
0x300425, 0x2785C6, 0x2F5439, 0x252FA8, 0x1B393F, 0x0DE0E4, 0x199D41, 0x0D0EDC,
0x300425, 0x4299B2, 0x2D456E, 0x3ECB00, 0x284000, 0x37D40F, 0x20FFCB, 0x2DC56D,
0x3496D3, 0x2F5439, 0x2B1D93, 0x286B74, 0x17AD66, 0x2377FE, 0x0D84E2, 0x12C062,
0x21692A, 0x1E11A5, 0x12ECDA, 0x110840, 0x3EBC76, 0x387608, 0x3B2652, 0x353BBA,
0x2A0000, 0x20FFCB, 0x23BFC6, 0x1C1681, 0x16BAF1, 0x0C1213, 0x1358E8, 0x09DCF8,
0x19924F, 0x141767, 0x0E7C16, 0x0BC28A, 0x0E5A0D, 0x075104, 0x0819B2, 0x04655D,
},
{
0x220000, 0x321B58, 0x380072, 0x4DAD38, 0x2F3318, 0x2CFA2D, 0x4D0556, 0x455122,
0x264000, 0x1E0D86, 0x3AF2FE, 0x2E50EB, 0x15D9CE, 0x0B23B2, 0x1FE730, 0x1113C7,
0x3A4E2D, 0x2FFDDF, 0x39788E, 0x2D2795, 0x210EA8, 0x10DA39, 0x1F1A61, 0x0FDB2F,
0x3A4E2D, 0x50DF33, 0x36F8E1, 0x4C3FA5, 0x30E000, 0x43CAA5, 0x281209, 0x37944D,
0x3FDBB7, 0x39788E, 0x345AC4, 0x3114C3, 0x1CC044, 0x2B11B4, 0x106A80, 0x16C509,
0x2891FC, 0x248324, 0x16FB08, 0x14AE97, 0x4C2DFD, 0x448F54, 0x47D31B, 0x40A3F5,
0x330000, 0x281209, 0x2B68DF, 0x221B53, 0x1B99DB, 0x0EA83B, 0x177E3E, 0x0BFA09,
0x1F0D17, 0x18658F, 0x1196AE, 0x0E47A8, 0x116D5A, 0x08E260, 0x09D60F, 0x055684,
},
{
0x2C0000, 0x40D818, 0x48790C, 0x6485D0, 0x3D14F2, 0x3A34B2, 0x63AC8D, 0x59B44A,
0x318000, 0x26E454, 0x4C4986, 0x3BF03F, 0x1C470A, 0x0E6A6E, 0x29494D, 0x161998,
0x4B743A, 0x3E1B5C, 0x4A5FC7, 0x3A6F75, 0x2AC7AC, 0x15CF1D, 0x284041, 0x148510,
0x4B743A, 0x68A861, 0x4723F6, 0x62ACB8, 0x3F4000, 0x57BAF3, 0x33DB1A, 0x47ED19,
0x52A3DE, 0x4A5FC7, 0x43C0C2, 0x3F8448, 0x25350D, 0x37BC8E, 0x153E87, 0x1D7775,
0x3480B0, 0x2F404C, 0x1DBD56, 0x1AC3D2, 0x6295DE, 0x58B97B, 0x5CF313, 0x53A701,
0x420000, 0x33DB1A, 0x382D5C, 0x2C235D, 0x23B80D, 0x12F7D4, 0x1E6723, 0x0F7FCF,
0x282F0E, 0x1F927D, 0x16C2FF, 0x127AD9, 0x168D83, 0x0B7F50, 0x0CBAAA, 0x06E86E,
},
};
#endif /* AVCODEC_BINKDATA_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/binkidct.c
-112
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@@ -1,112 +0,0 @@
/*
* Bink IDCT algorithm
* Copyright (c) 2009 Kostya Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Bink IDCT algorithm
*/
#include "dsputil.h"
#define A1 2896 /* (1/sqrt(2))<<12 */
#define A2 2217
#define A3 3784
#define A4 -5352
#define IDCT_TRANSFORM(dest,s0,s1,s2,s3,s4,s5,s6,s7,d0,d1,d2,d3,d4,d5,d6,d7,munge,src) {\
const int a0 = (src)[s0] + (src)[s4]; \
const int a1 = (src)[s0] - (src)[s4]; \
const int a2 = (src)[s2] + (src)[s6]; \
const int a3 = (A1*((src)[s2] - (src)[s6])) >> 11; \
const int a4 = (src)[s5] + (src)[s3]; \
const int a5 = (src)[s5] - (src)[s3]; \
const int a6 = (src)[s1] + (src)[s7]; \
const int a7 = (src)[s1] - (src)[s7]; \
const int b0 = a4 + a6; \
const int b1 = (A3*(a5 + a7)) >> 11; \
const int b2 = ((A4*a5) >> 11) - b0 + b1; \
const int b3 = (A1*(a6 - a4) >> 11) - b2; \
const int b4 = ((A2*a7) >> 11) + b3 - b1; \
(dest)[d0] = munge(a0+a2 +b0); \
(dest)[d1] = munge(a1+a3-a2+b2); \
(dest)[d2] = munge(a1-a3+a2+b3); \
(dest)[d3] = munge(a0-a2 -b4); \
(dest)[d4] = munge(a0-a2 +b4); \
(dest)[d5] = munge(a1-a3+a2-b3); \
(dest)[d6] = munge(a1+a3-a2-b2); \
(dest)[d7] = munge(a0+a2 -b0); \
}
/* end IDCT_TRANSFORM macro */
#define MUNGE_NONE(x) (x)
#define IDCT_COL(dest,src) IDCT_TRANSFORM(dest,0,8,16,24,32,40,48,56,0,8,16,24,32,40,48,56,MUNGE_NONE,src)
#define MUNGE_ROW(x) (((x) + 0x7F)>>8)
#define IDCT_ROW(dest,src) IDCT_TRANSFORM(dest,0,1,2,3,4,5,6,7,0,1,2,3,4,5,6,7,MUNGE_ROW,src)
static inline void bink_idct_col(DCTELEM *dest, const DCTELEM *src)
{
if ((src[8]|src[16]|src[24]|src[32]|src[40]|src[48]|src[56])==0) {
dest[0] =
dest[8] =
dest[16] =
dest[24] =
dest[32] =
dest[40] =
dest[48] =
dest[56] = src[0];
} else {
IDCT_COL(dest, src);
}
}
void ff_bink_idct_c(DCTELEM *block)
{
int i;
DCTELEM temp[64];
for (i = 0; i < 8; i++)
bink_idct_col(&temp[i], &block[i]);
for (i = 0; i < 8; i++) {
IDCT_ROW( (&block[8*i]), (&temp[8*i]) );
}
}
void ff_bink_idct_add_c(uint8_t *dest, int linesize, DCTELEM *block)
{
int i, j;
ff_bink_idct_c(block);
for (i = 0; i < 8; i++, dest += linesize, block += 8)
for (j = 0; j < 8; j++)
dest[j] += block[j];
}
void ff_bink_idct_put_c(uint8_t *dest, int linesize, DCTELEM *block)
{
int i;
DCTELEM temp[64];
for (i = 0; i < 8; i++)
bink_idct_col(&temp[i], &block[i]);
for (i = 0; i < 8; i++) {
IDCT_ROW( (&dest[i*linesize]), (&temp[8*i]) );
}
}
src/add-ons/media/plugins/ffmpeg/libavcodec/bitstream.c
-482
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@@ -1,482 +0,0 @@
/*
* Common bit i/o utils
* Copyright (c) 2000, 2001 Fabrice Bellard
* Copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2010 Loren Merritt
*
* alternative bitstream reader & writer by Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* bitstream api.
*/
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
const uint8_t ff_log2_run[32]={
0, 0, 0, 0, 1, 1, 1, 1,
2, 2, 2, 2, 3, 3, 3, 3,
4, 4, 5, 5, 6, 6, 7, 7,
8, 9,10,11,12,13,14,15
};
void align_put_bits(PutBitContext *s)
{
#ifdef ALT_BITSTREAM_WRITER
put_bits(s,( - s->index) & 7,0);
#else
put_bits(s,s->bit_left & 7,0);
#endif
}
void ff_put_string(PutBitContext *pb, const char *string, int terminate_string)
{
while(*string){
put_bits(pb, 8, *string);
string++;
}
if(terminate_string)
put_bits(pb, 8, 0);
}
void ff_copy_bits(PutBitContext *pb, const uint8_t *src, int length)
{
int words= length>>4;
int bits= length&15;
int i;
if(length==0) return;
if(CONFIG_SMALL || words < 16 || put_bits_count(pb)&7){
for(i=0; i<words; i++) put_bits(pb, 16, AV_RB16(src + 2*i));
}else{
for(i=0; put_bits_count(pb)&31; i++)
put_bits(pb, 8, src[i]);
flush_put_bits(pb);
memcpy(put_bits_ptr(pb), src+i, 2*words-i);
skip_put_bytes(pb, 2*words-i);
}
put_bits(pb, bits, AV_RB16(src + 2*words)>>(16-bits));
}
/* VLC decoding */
//#define DEBUG_VLC
#define GET_DATA(v, table, i, wrap, size) \
{\
const uint8_t *ptr = (const uint8_t *)table + i * wrap;\
switch(size) {\
case 1:\
v = *(const uint8_t *)ptr;\
break;\
case 2:\
v = *(const uint16_t *)ptr;\
break;\
default:\
v = *(const uint32_t *)ptr;\
break;\
}\
}
static int alloc_table(VLC *vlc, int size, int use_static)
{
int index;
index = vlc->table_size;
vlc->table_size += size;
if (vlc->table_size > vlc->table_allocated) {
if(use_static)
abort(); //cant do anything, init_vlc() is used with too little memory
vlc->table_allocated += (1 << vlc->bits);
vlc->table = av_realloc(vlc->table,
sizeof(VLC_TYPE) * 2 * vlc->table_allocated);
if (!vlc->table)
return -1;
}
return index;
}
static av_always_inline uint32_t bitswap_32(uint32_t x) {
return av_reverse[x&0xFF]<<24
| av_reverse[(x>>8)&0xFF]<<16
| av_reverse[(x>>16)&0xFF]<<8
| av_reverse[x>>24];
}
typedef struct {
uint8_t bits;
uint16_t symbol;
/** codeword, with the first bit-to-be-read in the msb
* (even if intended for a little-endian bitstream reader) */
uint32_t code;
} VLCcode;
static int compare_vlcspec(const void *a, const void *b)
{
const VLCcode *sa=a, *sb=b;
return (sa->code >> 1) - (sb->code >> 1);
}
#if __GNUC__ >= 3
/**
* Build VLC decoding tables suitable for use with get_vlc().
*
* @param vlc the context to be initted
*
* @param table_nb_bits max length of vlc codes to store directly in this table
* (Longer codes are delegated to subtables.)
*
* @param nb_codes number of elements in codes[]
*
* @param codes descriptions of the vlc codes
* These must be ordered such that codes going into the same subtable are contiguous.
* Sorting by VLCcode.code is sufficient, though not necessary.
*/
static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
VLCcode *codes, int flags)
{
int table_size, table_index, index, code_prefix, symbol, subtable_bits;
int i, j, k, n, nb, inc;
uint32_t code;
VLC_TYPE (*table)[2];
table_size = 1 << table_nb_bits;
table_index = alloc_table(vlc, table_size, flags & INIT_VLC_USE_NEW_STATIC);
#ifdef DEBUG_VLC
av_log(NULL,AV_LOG_DEBUG,"new table index=%d size=%d\n",
table_index, table_size);
#endif
if (table_index < 0)
return -1;
table = &vlc->table[table_index];
for (i = 0; i < table_size; i++) {
table[i][1] = 0; //bits
table[i][0] = -1; //codes
}
/* first pass: map codes and compute auxillary table sizes */
for (i = 0; i < nb_codes; i++) {
n = codes[i].bits;
code = codes[i].code;
symbol = codes[i].symbol;
#if defined(DEBUG_VLC) && 0
av_log(NULL,AV_LOG_DEBUG,"i=%d n=%d code=0x%x\n", i, n, code);
#endif
if (n <= table_nb_bits) {
/* no need to add another table */
j = code >> (32 - table_nb_bits);
nb = 1 << (table_nb_bits - n);
inc = 1;
if (flags & INIT_VLC_LE) {
j = bitswap_32(code);
inc = 1 << n;
}
for (k = 0; k < nb; k++) {
#ifdef DEBUG_VLC
av_log(NULL, AV_LOG_DEBUG, "%4x: code=%d n=%d\n",
j, i, n);
#endif
if (table[j][1] /*bits*/ != 0) {
av_log(NULL, AV_LOG_ERROR, "incorrect codes\n");
return -1;
}
table[j][1] = n; //bits
table[j][0] = symbol;
j += inc;
}
} else {
/* fill auxiliary table recursively */
n -= table_nb_bits;
code_prefix = code >> (32 - table_nb_bits);
subtable_bits = n;
codes[i].bits = n;
codes[i].code = code << table_nb_bits;
for (k = i+1; k < nb_codes; k++) {
n = codes[k].bits - table_nb_bits;
if (n <= 0)
break;
code = codes[k].code;
if (code >> (32 - table_nb_bits) != code_prefix)
break;
codes[k].bits = n;
codes[k].code = code << table_nb_bits;
subtable_bits = FFMAX(subtable_bits, n);
}
subtable_bits = FFMIN(subtable_bits, table_nb_bits);
j = (flags & INIT_VLC_LE) ? bitswap_32(code_prefix) >> (32 - table_nb_bits) : code_prefix;
table[j][1] = -subtable_bits;
#ifdef DEBUG_VLC
av_log(NULL,AV_LOG_DEBUG,"%4x: n=%d (subtable)\n",
j, codes[i].bits + table_nb_bits);
#endif
index = build_table(vlc, subtable_bits, k-i, codes+i, flags);
if (index < 0)
return -1;
/* note: realloc has been done, so reload tables */
table = &vlc->table[table_index];
table[j][0] = index; //code
i = k-1;
}
}
return table_index;
}
/* Build VLC decoding tables suitable for use with get_vlc().
'nb_bits' set thee decoding table size (2^nb_bits) entries. The
bigger it is, the faster is the decoding. But it should not be too
big to save memory and L1 cache. '9' is a good compromise.
'nb_codes' : number of vlcs codes
'bits' : table which gives the size (in bits) of each vlc code.
'codes' : table which gives the bit pattern of of each vlc code.
'symbols' : table which gives the values to be returned from get_vlc().
'xxx_wrap' : give the number of bytes between each entry of the
'bits' or 'codes' tables.
'xxx_size' : gives the number of bytes of each entry of the 'bits'
or 'codes' tables.
'wrap' and 'size' allows to use any memory configuration and types
(byte/word/long) to store the 'bits', 'codes', and 'symbols' tables.
'use_static' should be set to 1 for tables, which should be freed
with av_free_static(), 0 if free_vlc() will be used.
*/
int init_vlc_sparse(VLC *vlc, int nb_bits, int nb_codes,
const void *bits, int bits_wrap, int bits_size,
const void *codes, int codes_wrap, int codes_size,
const void *symbols, int symbols_wrap, int symbols_size,
int flags)
{
VLCcode buf[nb_codes];
int i, j;
vlc->bits = nb_bits;
if(flags & INIT_VLC_USE_NEW_STATIC){
if(vlc->table_size && vlc->table_size == vlc->table_allocated){
return 0;
}else if(vlc->table_size){
abort(); // fatal error, we are called on a partially initialized table
}
}else {
vlc->table = NULL;
vlc->table_allocated = 0;
vlc->table_size = 0;
}
#ifdef DEBUG_VLC
av_log(NULL,AV_LOG_DEBUG,"build table nb_codes=%d\n", nb_codes);
#endif
assert(symbols_size <= 2 || !symbols);
j = 0;
#define COPY(condition)\
for (i = 0; i < nb_codes; i++) {\
GET_DATA(buf[j].bits, bits, i, bits_wrap, bits_size);\
if (!(condition))\
continue;\
GET_DATA(buf[j].code, codes, i, codes_wrap, codes_size);\
if (flags & INIT_VLC_LE)\
buf[j].code = bitswap_32(buf[j].code);\
else\
buf[j].code <<= 32 - buf[j].bits;\
if (symbols)\
GET_DATA(buf[j].symbol, symbols, i, symbols_wrap, symbols_size)\
else\
buf[j].symbol = i;\
j++;\
}
COPY(buf[j].bits > nb_bits);
// qsort is the slowest part of init_vlc, and could probably be improved or avoided
qsort(buf, j, sizeof(VLCcode), compare_vlcspec);
COPY(buf[j].bits && buf[j].bits <= nb_bits);
nb_codes = j;
if (build_table(vlc, nb_bits, nb_codes, buf, flags) < 0) {
av_freep(&vlc->table);
return -1;
}
if((flags & INIT_VLC_USE_NEW_STATIC) && vlc->table_size != vlc->table_allocated)
av_log(NULL, AV_LOG_ERROR, "needed %d had %d\n", vlc->table_size, vlc->table_allocated);
return 0;
}
#else
static int build_table(VLC *vlc, int table_nb_bits,
int nb_codes,
const void *bits, int bits_wrap, int bits_size,
const void *codes, int codes_wrap, int codes_size,
const void *symbols, int symbols_wrap, int symbols_size,
uint32_t code_prefix, int n_prefix, int flags)
{
int i, j, k, n, table_size, table_index, nb, n1, index, code_prefix2, symbol;
uint32_t code;
VLC_TYPE (*table)[2];
table_size = 1 << table_nb_bits;
table_index = alloc_table(vlc, table_size, flags & INIT_VLC_USE_NEW_STATIC);
#ifdef DEBUG_VLC
av_log(NULL,AV_LOG_DEBUG,"new table index=%d size=%d code_prefix=%x n=%d\n",
table_index, table_size, code_prefix, n_prefix);
#endif
if (table_index < 0)
return -1;
table = &vlc->table[table_index];
for(i=0;i<table_size;i++) {
table[i][1] = 0; //bits
table[i][0] = -1; //codes
}
/* first pass: map codes and compute auxillary table sizes */
for(i=0;i<nb_codes;i++) {
GET_DATA(n, bits, i, bits_wrap, bits_size);
GET_DATA(code, codes, i, codes_wrap, codes_size);
/* we accept tables with holes */
if (n <= 0)
continue;
if (!symbols)
symbol = i;
else
GET_DATA(symbol, symbols, i, symbols_wrap, symbols_size);
#if defined(DEBUG_VLC) && 0
av_log(NULL,AV_LOG_DEBUG,"i=%d n=%d code=0x%x\n", i, n, code);
#endif
/* if code matches the prefix, it is in the table */
n -= n_prefix;
if(flags & INIT_VLC_LE)
code_prefix2= code & (n_prefix>=32 ? 0xffffffff : (1 << n_prefix)-1);
else
code_prefix2= code >> n;
if (n > 0 && code_prefix2 == code_prefix) {
if (n <= table_nb_bits) {
/* no need to add another table */
j = (code << (table_nb_bits - n)) & (table_size - 1);
nb = 1 << (table_nb_bits - n);
for(k=0;k<nb;k++) {
if(flags & INIT_VLC_LE)
j = (code >> n_prefix) + (k<<n);
#ifdef DEBUG_VLC
av_log(NULL, AV_LOG_DEBUG, "%4x: code=%d n=%d\n",
j, i, n);
#endif
if (table[j][1] /*bits*/ != 0) {
av_log(NULL, AV_LOG_ERROR, "incorrect codes\n");
return -1;
}
table[j][1] = n; //bits
table[j][0] = symbol;
j++;
}
} else {
n -= table_nb_bits;
j = (code >> ((flags & INIT_VLC_LE) ? n_prefix : n)) & ((1 << table_nb_bits) - 1);
#ifdef DEBUG_VLC
av_log(NULL,AV_LOG_DEBUG,"%4x: n=%d (subtable)\n",
j, n);
#endif
/* compute table size */
n1 = -table[j][1]; //bits
if (n > n1)
n1 = n;
table[j][1] = -n1; //bits
}
}
}
/* second pass : fill auxillary tables recursively */
for(i=0;i<table_size;i++) {
n = table[i][1]; //bits
if (n < 0) {
n = -n;
if (n > table_nb_bits) {
n = table_nb_bits;
table[i][1] = -n; //bits
}
index = build_table(vlc, n, nb_codes,
bits, bits_wrap, bits_size,
codes, codes_wrap, codes_size,
symbols, symbols_wrap, symbols_size,
(flags & INIT_VLC_LE) ? (code_prefix | (i << n_prefix)) : ((code_prefix << table_nb_bits) | i),
n_prefix + table_nb_bits, flags);
if (index < 0)
return -1;
/* note: realloc has been done, so reload tables */
table = &vlc->table[table_index];
table[i][0] = index; //code
}
}
return table_index;
}
int init_vlc_sparse(VLC *vlc, int nb_bits, int nb_codes,
const void *bits, int bits_wrap, int bits_size,
const void *codes, int codes_wrap, int codes_size,
const void *symbols, int symbols_wrap, int symbols_size,
int flags)
{
vlc->bits = nb_bits;
if(flags & INIT_VLC_USE_NEW_STATIC){
if(vlc->table_size && vlc->table_size == vlc->table_allocated){
return 0;
}else if(vlc->table_size){
abort(); // fatal error, we are called on a partially initialized table
}
}else {
vlc->table = NULL;
vlc->table_allocated = 0;
vlc->table_size = 0;
}
#ifdef DEBUG_VLC
av_log(NULL,AV_LOG_DEBUG,"build table nb_codes=%d\n", nb_codes);
#endif
if (build_table(vlc, nb_bits, nb_codes,
bits, bits_wrap, bits_size,
codes, codes_wrap, codes_size,
symbols, symbols_wrap, symbols_size,
0, 0, flags) < 0) {
av_freep(&vlc->table);
return -1;
}
if((flags & INIT_VLC_USE_NEW_STATIC) && vlc->table_size != vlc->table_allocated)
av_log(NULL, AV_LOG_ERROR, "needed %d had %d\n", vlc->table_size, vlc->table_allocated);
return 0;
}
#endif // GCC2 version
void free_vlc(VLC *vlc)
{
av_freep(&vlc->table);
}
src/add-ons/media/plugins/ffmpeg/libavcodec/bitstream_filter.c
-65
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@@ -1,65 +0,0 @@
/*
* copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
static AVBitStreamFilter *first_bitstream_filter= NULL;
AVBitStreamFilter *av_bitstream_filter_next(AVBitStreamFilter *f){
if(f) return f->next;
else return first_bitstream_filter;
}
void av_register_bitstream_filter(AVBitStreamFilter *bsf){
bsf->next = first_bitstream_filter;
first_bitstream_filter= bsf;
}
AVBitStreamFilterContext *av_bitstream_filter_init(const char *name){
AVBitStreamFilter *bsf= first_bitstream_filter;
while(bsf){
if(!strcmp(name, bsf->name)){
AVBitStreamFilterContext *bsfc= av_mallocz(sizeof(AVBitStreamFilterContext));
bsfc->filter= bsf;
bsfc->priv_data= av_mallocz(bsf->priv_data_size);
return bsfc;
}
bsf= bsf->next;
}
return NULL;
}
void av_bitstream_filter_close(AVBitStreamFilterContext *bsfc){
if(bsfc->filter->close)
bsfc->filter->close(bsfc);
av_freep(&bsfc->priv_data);
av_parser_close(bsfc->parser);
av_free(bsfc);
}
int av_bitstream_filter_filter(AVBitStreamFilterContext *bsfc,
AVCodecContext *avctx, const char *args,
uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size, int keyframe){
*poutbuf= (uint8_t *) buf;
*poutbuf_size= buf_size;
return bsfc->filter->filter(bsfc, avctx, args, poutbuf, poutbuf_size, buf, buf_size, keyframe);
}
src/add-ons/media/plugins/ffmpeg/libavcodec/bmp.c
-349
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@@ -1,349 +0,0 @@
/*
* BMP image format decoder
* Copyright (c) 2005 Mans Rullgard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "bytestream.h"
#include "bmp.h"
#include "msrledec.h"
static av_cold int bmp_decode_init(AVCodecContext *avctx){
BMPContext *s = avctx->priv_data;
avcodec_get_frame_defaults((AVFrame*)&s->picture);
avctx->coded_frame = (AVFrame*)&s->picture;
return 0;
}
static int bmp_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
BMPContext *s = avctx->priv_data;
AVFrame *picture = data;
AVFrame *p = &s->picture;
unsigned int fsize, hsize;
int width, height;
unsigned int depth;
BiCompression comp;
unsigned int ihsize;
int i, j, n, linesize;
uint32_t rgb[3];
uint8_t *ptr;
int dsize;
const uint8_t *buf0 = buf;
if(buf_size < 14){
av_log(avctx, AV_LOG_ERROR, "buf size too small (%d)\n", buf_size);
return -1;
}
if(bytestream_get_byte(&buf) != 'B' ||
bytestream_get_byte(&buf) != 'M') {
av_log(avctx, AV_LOG_ERROR, "bad magic number\n");
return -1;
}
fsize = bytestream_get_le32(&buf);
if(buf_size < fsize){
av_log(avctx, AV_LOG_ERROR, "not enough data (%d < %d), trying to decode anyway\n",
buf_size, fsize);
fsize = buf_size;
}
buf += 2; /* reserved1 */
buf += 2; /* reserved2 */
hsize = bytestream_get_le32(&buf); /* header size */
ihsize = bytestream_get_le32(&buf); /* more header size */
if(ihsize + 14 > hsize){
av_log(avctx, AV_LOG_ERROR, "invalid header size %d\n", hsize);
return -1;
}
/* sometimes file size is set to some headers size, set a real size in that case */
if(fsize == 14 || fsize == ihsize + 14)
fsize = buf_size - 2;
if(fsize <= hsize){
av_log(avctx, AV_LOG_ERROR, "declared file size is less than header size (%d < %d)\n",
fsize, hsize);
return -1;
}
switch(ihsize){
case 40: // windib v3
case 64: // OS/2 v2
case 108: // windib v4
case 124: // windib v5
width = bytestream_get_le32(&buf);
height = bytestream_get_le32(&buf);
break;
case 12: // OS/2 v1
width = bytestream_get_le16(&buf);
height = bytestream_get_le16(&buf);
break;
default:
av_log(avctx, AV_LOG_ERROR, "unsupported BMP file, patch welcome\n");
return -1;
}
if(bytestream_get_le16(&buf) != 1){ /* planes */
av_log(avctx, AV_LOG_ERROR, "invalid BMP header\n");
return -1;
}
depth = bytestream_get_le16(&buf);
if(ihsize == 40)
comp = bytestream_get_le32(&buf);
else
comp = BMP_RGB;
if(comp != BMP_RGB && comp != BMP_BITFIELDS && comp != BMP_RLE4 && comp != BMP_RLE8){
av_log(avctx, AV_LOG_ERROR, "BMP coding %d not supported\n", comp);
return -1;
}
if(comp == BMP_BITFIELDS){
buf += 20;
rgb[0] = bytestream_get_le32(&buf);
rgb[1] = bytestream_get_le32(&buf);
rgb[2] = bytestream_get_le32(&buf);
}
avctx->width = width;
avctx->height = height > 0? height: -height;
avctx->pix_fmt = PIX_FMT_NONE;
switch(depth){
case 32:
if(comp == BMP_BITFIELDS){
rgb[0] = (rgb[0] >> 15) & 3;
rgb[1] = (rgb[1] >> 15) & 3;
rgb[2] = (rgb[2] >> 15) & 3;
if(rgb[0] + rgb[1] + rgb[2] != 3 ||
rgb[0] == rgb[1] || rgb[0] == rgb[2] || rgb[1] == rgb[2]){
break;
}
} else {
rgb[0] = 2;
rgb[1] = 1;
rgb[2] = 0;
}
avctx->pix_fmt = PIX_FMT_BGR24;
break;
case 24:
avctx->pix_fmt = PIX_FMT_BGR24;
break;
case 16:
if(comp == BMP_RGB)
avctx->pix_fmt = PIX_FMT_RGB555;
if(comp == BMP_BITFIELDS)
avctx->pix_fmt = rgb[1] == 0x07E0 ? PIX_FMT_RGB565 : PIX_FMT_RGB555;
break;
case 8:
if(hsize - ihsize - 14 > 0)
avctx->pix_fmt = PIX_FMT_PAL8;
else
avctx->pix_fmt = PIX_FMT_GRAY8;
break;
case 4:
if(hsize - ihsize - 14 > 0){
avctx->pix_fmt = PIX_FMT_PAL8;
}else{
av_log(avctx, AV_LOG_ERROR, "Unknown palette for 16-colour BMP\n");
return -1;
}
break;
case 1:
avctx->pix_fmt = PIX_FMT_MONOBLACK;
break;
default:
av_log(avctx, AV_LOG_ERROR, "depth %d not supported\n", depth);
return -1;
}
if(avctx->pix_fmt == PIX_FMT_NONE){
av_log(avctx, AV_LOG_ERROR, "unsupported pixel format\n");
return -1;
}
if(p->data[0])
avctx->release_buffer(avctx, p);
p->reference = 0;
if(avctx->get_buffer(avctx, p) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
p->pict_type = FF_I_TYPE;
p->key_frame = 1;
buf = buf0 + hsize;
dsize = buf_size - hsize;
/* Line size in file multiple of 4 */
n = ((avctx->width * depth) / 8 + 3) & ~3;
if(n * avctx->height > dsize && comp != BMP_RLE4 && comp != BMP_RLE8){
av_log(avctx, AV_LOG_ERROR, "not enough data (%d < %d)\n",
dsize, n * avctx->height);
return -1;
}
// RLE may skip decoding some picture areas, so blank picture before decoding
if(comp == BMP_RLE4 || comp == BMP_RLE8)
memset(p->data[0], 0, avctx->height * p->linesize[0]);
if(depth == 4 || depth == 8)
memset(p->data[1], 0, 1024);
if(height > 0){
ptr = p->data[0] + (avctx->height - 1) * p->linesize[0];
linesize = -p->linesize[0];
} else {
ptr = p->data[0];
linesize = p->linesize[0];
}
if(avctx->pix_fmt == PIX_FMT_PAL8){
int colors = 1 << depth;
if(ihsize >= 36){
int t;
buf = buf0 + 46;
t = bytestream_get_le32(&buf);
if(t < 0 || t > (1 << depth)){
av_log(avctx, AV_LOG_ERROR, "Incorrect number of colors - %X for bitdepth %d\n", t, depth);
}else if(t){
colors = t;
}
}
buf = buf0 + 14 + ihsize; //palette location
if((hsize-ihsize-14) < (colors << 2)){ // OS/2 bitmap, 3 bytes per palette entry
for(i = 0; i < colors; i++)
((uint32_t*)p->data[1])[i] = bytestream_get_le24(&buf);
}else{
for(i = 0; i < colors; i++)
((uint32_t*)p->data[1])[i] = bytestream_get_le32(&buf);
}
buf = buf0 + hsize;
}
if(comp == BMP_RLE4 || comp == BMP_RLE8){
if(height < 0){
p->data[0] += p->linesize[0] * (avctx->height - 1);
p->linesize[0] = -p->linesize[0];
}
ff_msrle_decode(avctx, (AVPicture*)p, depth, buf, dsize);
if(height < 0){
p->data[0] += p->linesize[0] * (avctx->height - 1);
p->linesize[0] = -p->linesize[0];
}
}else{
switch(depth){
case 1:
case 8:
case 24:
for(i = 0; i < avctx->height; i++){
memcpy(ptr, buf, n);
buf += n;
ptr += linesize;
}
break;
case 4:
for(i = 0; i < avctx->height; i++){
int j;
for(j = 0; j < n; j++){
ptr[j*2+0] = (buf[j] >> 4) & 0xF;
ptr[j*2+1] = buf[j] & 0xF;
}
buf += n;
ptr += linesize;
}
break;
case 16:
for(i = 0; i < avctx->height; i++){
const uint16_t *src = (const uint16_t *) buf;
uint16_t *dst = (uint16_t *) ptr;
for(j = 0; j < avctx->width; j++)
*dst++ = le2me_16(*src++);
buf += n;
ptr += linesize;
}
break;
case 32:
for(i = 0; i < avctx->height; i++){
const uint8_t *src = buf;
uint8_t *dst = ptr;
for(j = 0; j < avctx->width; j++){
dst[0] = src[rgb[2]];
dst[1] = src[rgb[1]];
dst[2] = src[rgb[0]];
dst += 3;
src += 4;
}
buf += n;
ptr += linesize;
}
break;
default:
av_log(avctx, AV_LOG_ERROR, "BMP decoder is broken\n");
return -1;
}
}
*picture = s->picture;
*data_size = sizeof(AVPicture);
return buf_size;
}
static av_cold int bmp_decode_end(AVCodecContext *avctx)
{
BMPContext* c = avctx->priv_data;
if (c->picture.data[0])
avctx->release_buffer(avctx, &c->picture);
return 0;
}
AVCodec bmp_decoder = {
"bmp",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_BMP,
sizeof(BMPContext),
bmp_decode_init,
NULL,
bmp_decode_end,
bmp_decode_frame,
CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("BMP image"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/bmp.h
-38
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@@ -1,38 +0,0 @@
/*
* internals for BMP codecs
* Copyright (c) 2005 Mans Rullgard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_BMP_H
#define AVCODEC_BMP_H
#include "avcodec.h"
typedef struct BMPContext {
AVFrame picture;
} BMPContext;
typedef enum {
BMP_RGB =0,
BMP_RLE8 =1,
BMP_RLE4 =2,
BMP_BITFIELDS =3,
} BiCompression;
#endif /* AVCODEC_BMP_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/bmpenc.c
-149
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@@ -1,149 +0,0 @@
/*
* BMP image format encoder
* Copyright (c) 2006, 2007 Michel Bardiaux
* Copyright (c) 2009 Daniel Verkamp <daniel at drv.nu>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "bytestream.h"
#include "bmp.h"
static const uint32_t monoblack_pal[] = { 0x000000, 0xFFFFFF };
static const uint32_t rgb565_masks[] = { 0xF800, 0x07E0, 0x001F };
static av_cold int bmp_encode_init(AVCodecContext *avctx){
BMPContext *s = avctx->priv_data;
avcodec_get_frame_defaults((AVFrame*)&s->picture);
avctx->coded_frame = (AVFrame*)&s->picture;
return 0;
}
static int bmp_encode_frame(AVCodecContext *avctx, unsigned char *buf, int buf_size, void *data){
BMPContext *s = avctx->priv_data;
AVFrame *pict = data;
AVFrame * const p= (AVFrame*)&s->picture;
int n_bytes_image, n_bytes_per_row, n_bytes, i, n, hsize;
const uint32_t *pal = NULL;
int pad_bytes_per_row, bit_count, pal_entries = 0, compression = BMP_RGB;
uint8_t *ptr;
unsigned char* buf0 = buf;
*p = *pict;
p->pict_type= FF_I_TYPE;
p->key_frame= 1;
switch (avctx->pix_fmt) {
case PIX_FMT_BGR24:
bit_count = 24;
break;
case PIX_FMT_RGB555:
bit_count = 16;
break;
case PIX_FMT_RGB565:
bit_count = 16;
compression = BMP_BITFIELDS;
pal = rgb565_masks; // abuse pal to hold color masks
pal_entries = 3;
break;
case PIX_FMT_RGB8:
case PIX_FMT_BGR8:
case PIX_FMT_RGB4_BYTE:
case PIX_FMT_BGR4_BYTE:
case PIX_FMT_GRAY8:
case PIX_FMT_PAL8:
bit_count = 8;
pal = (uint32_t *)p->data[1];
break;
case PIX_FMT_MONOBLACK:
bit_count = 1;
pal = monoblack_pal;
break;
default:
return -1;
}
if (pal && !pal_entries) pal_entries = 1 << bit_count;
n_bytes_per_row = ((int64_t)avctx->width * (int64_t)bit_count + 7LL) >> 3LL;
pad_bytes_per_row = (4 - n_bytes_per_row) & 3;
n_bytes_image = avctx->height * (n_bytes_per_row + pad_bytes_per_row);
// STRUCTURE.field refer to the MSVC documentation for BITMAPFILEHEADER
// and related pages.
#define SIZE_BITMAPFILEHEADER 14
#define SIZE_BITMAPINFOHEADER 40
hsize = SIZE_BITMAPFILEHEADER + SIZE_BITMAPINFOHEADER + (pal_entries << 2);
n_bytes = n_bytes_image + hsize;
if(n_bytes>buf_size) {
av_log(avctx, AV_LOG_ERROR, "buf size too small (need %d, got %d)\n", n_bytes, buf_size);
return -1;
}
bytestream_put_byte(&buf, 'B'); // BITMAPFILEHEADER.bfType
bytestream_put_byte(&buf, 'M'); // do.
bytestream_put_le32(&buf, n_bytes); // BITMAPFILEHEADER.bfSize
bytestream_put_le16(&buf, 0); // BITMAPFILEHEADER.bfReserved1
bytestream_put_le16(&buf, 0); // BITMAPFILEHEADER.bfReserved2
bytestream_put_le32(&buf, hsize); // BITMAPFILEHEADER.bfOffBits
bytestream_put_le32(&buf, SIZE_BITMAPINFOHEADER); // BITMAPINFOHEADER.biSize
bytestream_put_le32(&buf, avctx->width); // BITMAPINFOHEADER.biWidth
bytestream_put_le32(&buf, avctx->height); // BITMAPINFOHEADER.biHeight
bytestream_put_le16(&buf, 1); // BITMAPINFOHEADER.biPlanes
bytestream_put_le16(&buf, bit_count); // BITMAPINFOHEADER.biBitCount
bytestream_put_le32(&buf, compression); // BITMAPINFOHEADER.biCompression
bytestream_put_le32(&buf, n_bytes_image); // BITMAPINFOHEADER.biSizeImage
bytestream_put_le32(&buf, 0); // BITMAPINFOHEADER.biXPelsPerMeter
bytestream_put_le32(&buf, 0); // BITMAPINFOHEADER.biYPelsPerMeter
bytestream_put_le32(&buf, 0); // BITMAPINFOHEADER.biClrUsed
bytestream_put_le32(&buf, 0); // BITMAPINFOHEADER.biClrImportant
for (i = 0; i < pal_entries; i++)
bytestream_put_le32(&buf, pal[i] & 0xFFFFFF);
// BMP files are bottom-to-top so we start from the end...
ptr = p->data[0] + (avctx->height - 1) * p->linesize[0];
buf = buf0 + hsize;
for(i = 0; i < avctx->height; i++) {
if (bit_count == 16) {
const uint16_t *src = (const uint16_t *) ptr;
uint16_t *dst = (uint16_t *) buf;
for(n = 0; n < avctx->width; n++)
AV_WL16(dst + n, src[n]);
} else {
memcpy(buf, ptr, n_bytes_per_row);
}
buf += n_bytes_per_row;
memset(buf, 0, pad_bytes_per_row);
buf += pad_bytes_per_row;
ptr -= p->linesize[0]; // ... and go back
}
return n_bytes;
}
AVCodec bmp_encoder = {
"bmp",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_BMP,
sizeof(BMPContext),
bmp_encode_init,
bmp_encode_frame,
NULL, //encode_end,
.pix_fmts = (const enum PixelFormat[]){
PIX_FMT_BGR24,
PIX_FMT_RGB555, PIX_FMT_RGB565,
PIX_FMT_RGB8, PIX_FMT_BGR8, PIX_FMT_RGB4_BYTE, PIX_FMT_BGR4_BYTE, PIX_FMT_GRAY8, PIX_FMT_PAL8,
PIX_FMT_MONOBLACK,
PIX_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("BMP image"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/bytestream.h
-71
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@@ -1,71 +0,0 @@
/*
* Bytestream functions
* copyright (c) 2006 Baptiste Coudurier <baptiste.coudurier@free.fr>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_BYTESTREAM_H
#define AVCODEC_BYTESTREAM_H
#include <string.h>
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"
#define DEF_T(type, name, bytes, read, write) \
static av_always_inline type bytestream_get_ ## name(const uint8_t **b){\
(*b) += bytes;\
return read(*b - bytes);\
}\
static av_always_inline void bytestream_put_ ##name(uint8_t **b, const type value){\
write(*b, value);\
(*b) += bytes;\
}
#define DEF(name, bytes, read, write) \
DEF_T(unsigned int, name, bytes, read, write)
#define DEF64(name, bytes, read, write) \
DEF_T(uint64_t, name, bytes, read, write)
DEF64(le64, 8, AV_RL64, AV_WL64)
DEF (le32, 4, AV_RL32, AV_WL32)
DEF (le24, 3, AV_RL24, AV_WL24)
DEF (le16, 2, AV_RL16, AV_WL16)
DEF64(be64, 8, AV_RB64, AV_WB64)
DEF (be32, 4, AV_RB32, AV_WB32)
DEF (be24, 3, AV_RB24, AV_WB24)
DEF (be16, 2, AV_RB16, AV_WB16)
DEF (byte, 1, AV_RB8 , AV_WB8 )
#undef DEF
#undef DEF64
#undef DEF_T
static av_always_inline unsigned int bytestream_get_buffer(const uint8_t **b, uint8_t *dst, unsigned int size)
{
memcpy(dst, *b, size);
(*b) += size;
return size;
}
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
{
memcpy(*b, src, size);
(*b) += size;
}
#endif /* AVCODEC_BYTESTREAM_H */
src/add-ons/media/plugins/ffmpeg/libavcodec/c93.c
-256
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@@ -1,256 +0,0 @@
/*
* Interplay C93 video decoder
* Copyright (c) 2007 Anssi Hannula <anssi.hannula@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "bytestream.h"
typedef struct {
AVFrame pictures[2];
int currentpic;
} C93DecoderContext;
typedef enum {
C93_8X8_FROM_PREV = 0x02,
C93_4X4_FROM_PREV = 0x06,
C93_4X4_FROM_CURR = 0x07,
C93_8X8_2COLOR = 0x08,
C93_4X4_2COLOR = 0x0A,
C93_4X4_4COLOR_GRP = 0x0B,
C93_4X4_4COLOR = 0x0D,
C93_NOOP = 0x0E,
C93_8X8_INTRA = 0x0F,
} C93BlockType;
#define WIDTH 320
#define HEIGHT 192
#define C93_HAS_PALETTE 0x01
#define C93_FIRST_FRAME 0x02
static av_cold int decode_init(AVCodecContext *avctx)
{
avctx->pix_fmt = PIX_FMT_PAL8;
return 0;
}
static av_cold int decode_end(AVCodecContext *avctx)
{
C93DecoderContext * const c93 = avctx->priv_data;
if (c93->pictures[0].data[0])
avctx->release_buffer(avctx, &c93->pictures[0]);
if (c93->pictures[1].data[0])
avctx->release_buffer(avctx, &c93->pictures[1]);
return 0;
}
static inline int copy_block(AVCodecContext *avctx, uint8_t *to,
uint8_t *from, int offset, int height, int stride)
{
int i;
int width = height;
int from_x = offset % WIDTH;
int from_y = offset / WIDTH;
int overflow = from_x + width - WIDTH;
if (!from) {
/* silently ignoring predictive blocks in first frame */
return 0;
}
if (from_y + height > HEIGHT) {
av_log(avctx, AV_LOG_ERROR, "invalid offset %d during C93 decoding\n",
offset);
return -1;
}
if (overflow > 0) {
width -= overflow;
for (i = 0; i < height; i++) {
memcpy(&to[i*stride+width], &from[(from_y+i)*stride], overflow);
}
}
for (i = 0; i < height; i++) {
memcpy(&to[i*stride], &from[(from_y+i)*stride+from_x], width);
}
return 0;
}
static inline void draw_n_color(uint8_t *out, int stride, int width,
int height, int bpp, uint8_t cols[4], uint8_t grps[4], uint32_t col)
{
int x, y;
for (y = 0; y < height; y++) {
if (grps)
cols[0] = grps[3 * (y >> 1)];
for (x = 0; x < width; x++) {
if (grps)
cols[1]= grps[(x >> 1) + 1];
out[x + y*stride] = cols[col & ((1 << bpp) - 1)];
col >>= bpp;
}
}
}
static int decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
C93DecoderContext * const c93 = avctx->priv_data;
AVFrame * const newpic = &c93->pictures[c93->currentpic];
AVFrame * const oldpic = &c93->pictures[c93->currentpic^1];
AVFrame *picture = data;
uint8_t *out;
int stride, i, x, y, bt = 0;
c93->currentpic ^= 1;
newpic->reference = 1;
newpic->buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE |
FF_BUFFER_HINTS_REUSABLE | FF_BUFFER_HINTS_READABLE;
if (avctx->reget_buffer(avctx, newpic)) {
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
return -1;
}
stride = newpic->linesize[0];
if (buf[0] & C93_FIRST_FRAME) {
newpic->pict_type = FF_I_TYPE;
newpic->key_frame = 1;
} else {
newpic->pict_type = FF_P_TYPE;
newpic->key_frame = 0;
}
if (*buf++ & C93_HAS_PALETTE) {
uint32_t *palette = (uint32_t *) newpic->data[1];
const uint8_t *palbuf = buf + buf_size - 768 - 1;
for (i = 0; i < 256; i++) {
palette[i] = bytestream_get_be24(&palbuf);
}
} else {
if (oldpic->data[1])
memcpy(newpic->data[1], oldpic->data[1], 256 * 4);
}
for (y = 0; y < HEIGHT; y += 8) {
out = newpic->data[0] + y * stride;
for (x = 0; x < WIDTH; x += 8) {
uint8_t *copy_from = oldpic->data[0];
unsigned int offset, j;
uint8_t cols[4], grps[4];
C93BlockType block_type;
if (!bt)
bt = *buf++;
block_type= bt & 0x0F;
switch (block_type) {
case C93_8X8_FROM_PREV:
offset = bytestream_get_le16(&buf);
if (copy_block(avctx, out, copy_from, offset, 8, stride))
return -1;
break;
case C93_4X4_FROM_CURR:
copy_from = newpic->data[0];
case C93_4X4_FROM_PREV:
for (j = 0; j < 8; j += 4) {
for (i = 0; i < 8; i += 4) {
offset = bytestream_get_le16(&buf);
if (copy_block(avctx, &out[j*stride+i],
copy_from, offset, 4, stride))
return -1;
}
}
break;
case C93_8X8_2COLOR:
bytestream_get_buffer(&buf, cols, 2);
for (i = 0; i < 8; i++) {
draw_n_color(out + i*stride, stride, 8, 1, 1, cols,
NULL, *buf++);
}
break;
case C93_4X4_2COLOR:
case C93_4X4_4COLOR:
case C93_4X4_4COLOR_GRP:
for (j = 0; j < 8; j += 4) {
for (i = 0; i < 8; i += 4) {
if (block_type == C93_4X4_2COLOR) {
bytestream_get_buffer(&buf, cols, 2);
draw_n_color(out + i + j*stride, stride, 4, 4,
1, cols, NULL, bytestream_get_le16(&buf));
} else if (block_type == C93_4X4_4COLOR) {
bytestream_get_buffer(&buf, cols, 4);
draw_n_color(out + i + j*stride, stride, 4, 4,
2, cols, NULL, bytestream_get_le32(&buf));
} else {
bytestream_get_buffer(&buf, grps, 4);
draw_n_color(out + i + j*stride, stride, 4, 4,
1, cols, grps, bytestream_get_le16(&buf));
}
}
}
break;
case C93_NOOP:
break;
case C93_8X8_INTRA:
for (j = 0; j < 8; j++)
bytestream_get_buffer(&buf, out + j*stride, 8);
break;
default:
av_log(avctx, AV_LOG_ERROR, "unexpected type %x at %dx%d\n",
block_type, x, y);
return -1;
}
bt >>= 4;
out += 8;
}
}
*picture = *newpic;
*data_size = sizeof(AVFrame);
return buf_size;
}
AVCodec c93_decoder = {
"c93",
AVMEDIA_TYPE_VIDEO,
CODEC_ID_C93,
sizeof(C93DecoderContext),
decode_init,
NULL,
decode_end,
decode_frame,
CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Interplay C93"),
};
src/add-ons/media/plugins/ffmpeg/libavcodec/cabac.c
-268
View File
@@ -1,268 +0,0 @@
/*
* H.26L/H.264/AVC/JVT/14496-10/... encoder/decoder
* Copyright (c) 2003 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Context Adaptive Binary Arithmetic Coder.
*/
#include <string.h>
#include "libavutil/common.h"
#include "get_bits.h"
#include "cabac.h"
static const uint8_t lps_range[64][4]= {
{128,176,208,240}, {128,167,197,227}, {128,158,187,216}, {123,150,178,205},
{116,142,169,195}, {111,135,160,185}, {105,128,152,175}, {100,122,144,166},
{ 95,116,137,158}, { 90,110,130,150}, { 85,104,123,142}, { 81, 99,117,135},
{ 77, 94,111,128}, { 73, 89,105,122}, { 69, 85,100,116}, { 66, 80, 95,110},
{ 62, 76, 90,104}, { 59, 72, 86, 99}, { 56, 69, 81, 94}, { 53, 65, 77, 89},
{ 51, 62, 73, 85}, { 48, 59, 69, 80}, { 46, 56, 66, 76}, { 43, 53, 63, 72},
{ 41, 50, 59, 69}, { 39, 48, 56, 65}, { 37, 45, 54, 62}, { 35, 43, 51, 59},
{ 33, 41, 48, 56}, { 32, 39, 46, 53}, { 30, 37, 43, 50}, { 29, 35, 41, 48},
{ 27, 33, 39, 45}, { 26, 31, 37, 43}, { 24, 30, 35, 41}, { 23, 28, 33, 39},
{ 22, 27, 32, 37}, { 21, 26, 30, 35}, { 20, 24, 29, 33}, { 19, 23, 27, 31},
{ 18, 22, 26, 30}, { 17, 21, 25, 28}, { 16, 20, 23, 27}, { 15, 19, 22, 25},
{ 14, 18, 21, 24}, { 14, 17, 20, 23}, { 13, 16, 19, 22}, { 12, 15, 18, 21},
{ 12, 14, 17, 20}, { 11, 14, 16, 19}, { 11, 13, 15, 18}, { 10, 12, 15, 17},
{ 10, 12, 14, 16}, { 9, 11, 13, 15}, { 9, 11, 12, 14}, { 8, 10, 12, 14},
{ 8, 9, 11, 13}, { 7, 9, 11, 12}, { 7, 9, 10, 12}, { 7, 8, 10, 11},
{ 6, 8, 9, 11}, { 6, 7, 9, 10}, { 6, 7, 8, 9}, { 2, 2, 2, 2},
};
uint8_t ff_h264_mlps_state[4*64];
uint8_t ff_h264_lps_range[4*2*64];
uint8_t ff_h264_lps_state[2*64];
uint8_t ff_h264_mps_state[2*64];
static const uint8_t mps_state[64]= {
1, 2, 3, 4, 5, 6, 7, 8,
9,10,11,12,13,14,15,16,
17,18,19,20,21,22,23,24,
25,26,27,28,29,30,31,32,
33,34,35,36,37,38,39,40,
41,42,43,44,45,46,47,48,
49,50,51,52,53,54,55,56,
57,58,59,60,61,62,62,63,
};
static const uint8_t lps_state[64]= {
0, 0, 1, 2, 2, 4, 4, 5,
6, 7, 8, 9, 9,11,11,12,
13,13,15,15,16,16,18,18,
19,19,21,21,22,22,23,24,
24,25,26,26,27,27,28,29,
29,30,30,30,31,32,32,33,
33,33,34,34,35,35,35,36,
36,36,37,37,37,38,38,63,
};
#if 0
const uint8_t ff_h264_norm_shift_old[128]= {
7,6,5,5,4,4,4,4,3,3,3,3,3,3,3,3,
2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,
1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,
1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
};
#endif
const uint8_t ff_h264_norm_shift[512]= {
9,8,7,7,6,6,6,6,5,5,5,5,5,5,5,5,
4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,
3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,
2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,
2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,
2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,
2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,
1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,
1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,
1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,
1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
};
/**
*
* @param buf_size size of buf in bits
*/
void ff_init_cabac_encoder(CABACContext *c, uint8_t *buf, int buf_size){
init_put_bits(&c->pb, buf, buf_size);
c->low= 0;
c->range= 0x1FE;
c->outstanding_count= 0;
#ifdef STRICT_LIMITS
c->sym_count =0;
#endif
c->pb.bit_left++; //avoids firstBitFlag
}
/**
*
* @param buf_size size of buf in bits
*/
void ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size){
c->bytestream_start=
c->bytestream= buf;
c->bytestream_end= buf + buf_size;
#if CABAC_BITS == 16
c->low = (*c->bytestream++)<<18;
c->low+= (*c->bytestream++)<<10;
#else
c->low = (*c->bytestream++)<<10;
#endif
c->low+= ((*c->bytestream++)<<2) + 2;
c->range= 0x1FE;
}
void ff_init_cabac_states(CABACContext *c){
int i, j;
for(i=0; i<64; i++){
for(j=0; j<4; j++){ //FIXME check if this is worth the 1 shift we save
ff_h264_lps_range[j*2*64+2*i+0]=
ff_h264_lps_range[j*2*64+2*i+1]= lps_range[i][j];
}
ff_h264_mlps_state[128+2*i+0]=
ff_h264_mps_state[2*i+0]= 2*mps_state[i]+0;
ff_h264_mlps_state[128+2*i+1]=
ff_h264_mps_state[2*i+1]= 2*mps_state[i]+1;
if( i ){
#ifdef BRANCHLESS_CABAC_DECODER
ff_h264_mlps_state[128-2*i-1]= 2*lps_state[i]+0;
ff_h264_mlps_state[128-2*i-2]= 2*lps_state[i]+1;
}else{
ff_h264_mlps_state[128-2*i-1]= 1;
ff_h264_mlps_state[128-2*i-2]= 0;
#else
ff_h264_lps_state[2*i+0]= 2*lps_state[i]+0;
ff_h264_lps_state[2*i+1]= 2*lps_state[i]+1;
}else{
ff_h264_lps_state[2*i+0]= 1;
ff_h264_lps_state[2*i+1]= 0;
#endif
}
}
}
#ifdef TEST
#define SIZE 10240
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "cabac.h"
int main(void){
CABACContext c;
uint8_t b[9*SIZE];
uint8_t r[9*SIZE];
int i;
uint8_t state[10]= {0};
AVLFG prng;
av_lfg_init(&prng, 1);
ff_init_cabac_encoder(&c, b, SIZE);
ff_init_cabac_states(&c);
for(i=0; i<SIZE; i++){
r[i] = av_lfg_get(&prng) % 7;
}
for(i=0; i<SIZE; i++){
START_TIMER
put_cabac_bypass(&c, r[i]&1);
STOP_TIMER("put_cabac_bypass")
}
for(i=0; i<SIZE; i++){
START_TIMER
put_cabac(&c, state, r[i]&1);
STOP_TIMER("put_cabac")
}
for(i=0; i<SIZE; i++){
START_TIMER
put_cabac_u(&c, state, r[i], 6, 3, i&1);
STOP_TIMER("put_cabac_u")
}
for(i=0; i<SIZE; i++){
START_TIMER
put_cabac_ueg(&c, state, r[i], 3, 0, 1, 2);
STOP_TIMER("put_cabac_ueg")
}
put_cabac_terminate(&c, 1);
ff_init_cabac_decoder(&c, b, SIZE);
memset(state, 0, sizeof(state));
for(i=0; i<SIZE; i++){
START_TIMER
if( (r[i]&1) != get_cabac_bypass(&c) )
av_log(NULL, AV_LOG_ERROR, "CABAC bypass failure at %d\n", i);
STOP_TIMER("get_cabac_bypass")
}
for(i=0; i<SIZE; i++){
START_TIMER
if( (r[i]&1) != get_cabac(&c, state) )
av_log(NULL, AV_LOG_ERROR, "CABAC failure at %d\n", i);
STOP_TIMER("get_cabac")
}
#if 0
for(i=0; i<SIZE; i++){
START_TIMER
if( r[i] != get_cabac_u(&c, state, (i&1) ? 6 : 7, 3, i&1) )
av_log(NULL, AV_LOG_ERROR, "CABAC unary (truncated) binarization failure at %d\n", i);
STOP_TIMER("get_cabac_u")
}
for(i=0; i<SIZE; i++){
START_TIMER
if( r[i] != get_cabac_ueg(&c, state, 3, 0, 1, 2))
av_log(NULL, AV_LOG_ERROR, "CABAC unary (truncated) binarization failure at %d\n", i);
STOP_TIMER("get_cabac_ueg")
}
#endif
if(!get_cabac_terminate(&c))
av_log(NULL, AV_LOG_ERROR, "where's the Terminator?\n");
return 0;
}
#endif /* TEST */

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