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/*
* DCA compatible decoder
* Copyright ( C ) 2004 Gildas Bazin
* Copyright ( C ) 2004 Benjamin Zores
* Copyright ( C ) 2006 Benjamin Larsson
* Copyright ( C ) 2007 Konstantin Shishkov
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* Copyright ( C ) 2012 Paul B Mahol
* Copyright ( C ) 2014 Niels Möller
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*
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* This file is part of Libav .
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*
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* Libav is free software ; you can redistribute it and / or
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* 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 .
*
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* Libav is distributed in the hope that it will be useful ,
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* 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
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* License along with Libav ; if not , write to the Free Software
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* Foundation , Inc . , 51 Franklin Street , Fifth Floor , Boston , MA 02110 - 1301 USA
*/
# include <math.h>
# include <stddef.h>
# include <stdio.h>
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# include "libavutil/attributes.h"
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# include "libavutil/channel_layout.h"
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# include "libavutil/common.h"
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# include "libavutil/float_dsp.h"
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# include "libavutil/internal.h"
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# include "libavutil/intreadwrite.h"
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# include "libavutil/mathematics.h"
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# include "libavutil/opt.h"
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# include "libavutil/samplefmt.h"
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# include "avcodec.h"
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# include "dca.h"
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# include "dca_syncwords.h"
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# include "dcadata.h"
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# include "dcadsp.h"
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# include "dcahuff.h"
# include "fft.h"
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# include "fmtconvert.h"
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# include "get_bits.h"
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# include "internal.h"
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# include "mathops.h"
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# include "profiles.h"
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# include "put_bits.h"
# include "synth_filter.h"
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# if ARCH_ARM
# include "arm / dca.h"
# endif
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enum DCAMode {
DCA_MONO = 0 ,
DCA_CHANNEL ,
DCA_STEREO ,
DCA_STEREO_SUMDIFF ,
DCA_STEREO_TOTAL ,
DCA_3F ,
DCA_2F1R ,
DCA_3F1R ,
DCA_2F2R ,
DCA_3F2R ,
DCA_4F2R
} ;
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/* -1 are reserved or unknown */
static const int dca_ext_audio_descr_mask [ ] = {
DCA_EXT_XCH ,
- 1 ,
DCA_EXT_X96 ,
DCA_EXT_XCH | DCA_EXT_X96 ,
- 1 ,
- 1 ,
DCA_EXT_XXCH ,
- 1 ,
} ;
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/* Tables for mapping dts channel configurations to libavcodec multichannel api.
* Some compromises have been made for special configurations . Most configurations
* are never used so complete accuracy is not needed .
*
* L = left , R = right , C = center , S = surround , F = front , R = rear , T = total , OV = overhead .
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* S - > side , when both rear and back are configured move one of them to the side channel
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* OV - > center back
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* All 2 channel configurations - > AV_CH_LAYOUT_STEREO
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*/
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static const uint64_t dca_core_channel_layout [ ] = {
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AV_CH_FRONT_CENTER , ///< 1, A
AV_CH_LAYOUT_STEREO , ///< 2, A + B (dual mono)
AV_CH_LAYOUT_STEREO , ///< 2, L + R (stereo)
AV_CH_LAYOUT_STEREO , ///< 2, (L + R) + (L - R) (sum-difference)
AV_CH_LAYOUT_STEREO , ///< 2, LT + RT (left and right total)
AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER , ///< 3, C + L + R
AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER , ///< 3, L + R + S
AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER , ///< 4, C + L + R + S
AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT , ///< 4, L + R + SL + SR
AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
AV_CH_SIDE_RIGHT , ///< 5, C + L + R + SL + SR
AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER , ///< 6, CL + CR + L + R + SL + SR
AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER , ///< 6, C + L + R + LR + RR + OV
AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER |
AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT , ///< 6, CF + CR + LF + RF + LR + RR
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT , ///< 7, CL + C + CR + L + R + SL + SR
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT , ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT , ///< 8, CL + C + CR + L + R + SL + S + SR
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} ;
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# define DCA_DOLBY 101 /* FIXME */
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# define DCA_CHANNEL_BITS 6
# define DCA_CHANNEL_MASK 0x3F
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# define DCA_LFE 0x80
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# define HEADER_SIZE 14
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# define DCA_NSYNCAUX 0x9A1105A0
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/** Bit allocation */
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typedef struct BitAlloc {
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int offset ; ///< code values offset
int maxbits [ 8 ] ; ///< max bits in VLC
int wrap ; ///< wrap for get_vlc2()
VLC vlc [ 8 ] ; ///< actual codes
} BitAlloc ;
static BitAlloc dca_bitalloc_index ; ///< indexes for samples VLC select
static BitAlloc dca_tmode ; ///< transition mode VLCs
static BitAlloc dca_scalefactor ; ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc [ 11 ] ; ///< samples VLCs
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static av_always_inline int get_bitalloc ( GetBitContext * gb , BitAlloc * ba ,
int idx )
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{
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return get_vlc2 ( gb , ba - > vlc [ idx ] . table , ba - > vlc [ idx ] . bits , ba - > wrap ) +
ba - > offset ;
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}
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static av_cold void dca_init_vlcs ( void )
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{
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static int vlcs_initialized = 0 ;
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int i , j , c = 14 ;
static VLC_TYPE dca_table [ 23622 ] [ 2 ] ;
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if ( vlcs_initialized )
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return ;
dca_bitalloc_index . offset = 1 ;
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dca_bitalloc_index . wrap = 2 ;
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for ( i = 0 ; i < 5 ; i + + ) {
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dca_bitalloc_index . vlc [ i ] . table = & dca_table [ ff_dca_vlc_offs [ i ] ] ;
dca_bitalloc_index . vlc [ i ] . table_allocated = ff_dca_vlc_offs [ i + 1 ] - ff_dca_vlc_offs [ i ] ;
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init_vlc ( & dca_bitalloc_index . vlc [ i ] , bitalloc_12_vlc_bits [ i ] , 12 ,
bitalloc_12_bits [ i ] , 1 , 1 ,
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bitalloc_12_codes [ i ] , 2 , 2 , INIT_VLC_USE_NEW_STATIC ) ;
}
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dca_scalefactor . offset = - 64 ;
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dca_scalefactor . wrap = 2 ;
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for ( i = 0 ; i < 5 ; i + + ) {
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dca_scalefactor . vlc [ i ] . table = & dca_table [ ff_dca_vlc_offs [ i + 5 ] ] ;
dca_scalefactor . vlc [ i ] . table_allocated = ff_dca_vlc_offs [ i + 6 ] - ff_dca_vlc_offs [ i + 5 ] ;
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init_vlc ( & dca_scalefactor . vlc [ i ] , SCALES_VLC_BITS , 129 ,
scales_bits [ i ] , 1 , 1 ,
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scales_codes [ i ] , 2 , 2 , INIT_VLC_USE_NEW_STATIC ) ;
}
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dca_tmode . offset = 0 ;
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dca_tmode . wrap = 1 ;
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for ( i = 0 ; i < 4 ; i + + ) {
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dca_tmode . vlc [ i ] . table = & dca_table [ ff_dca_vlc_offs [ i + 10 ] ] ;
dca_tmode . vlc [ i ] . table_allocated = ff_dca_vlc_offs [ i + 11 ] - ff_dca_vlc_offs [ i + 10 ] ;
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init_vlc ( & dca_tmode . vlc [ i ] , tmode_vlc_bits [ i ] , 4 ,
tmode_bits [ i ] , 1 , 1 ,
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tmode_codes [ i ] , 2 , 2 , INIT_VLC_USE_NEW_STATIC ) ;
}
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for ( i = 0 ; i < 10 ; i + + )
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for ( j = 0 ; j < 7 ; j + + ) {
if ( ! bitalloc_codes [ i ] [ j ] )
break ;
dca_smpl_bitalloc [ i + 1 ] . offset = bitalloc_offsets [ i ] ;
dca_smpl_bitalloc [ i + 1 ] . wrap = 1 + ( j > 4 ) ;
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dca_smpl_bitalloc [ i + 1 ] . vlc [ j ] . table = & dca_table [ ff_dca_vlc_offs [ c ] ] ;
dca_smpl_bitalloc [ i + 1 ] . vlc [ j ] . table_allocated = ff_dca_vlc_offs [ c + 1 ] - ff_dca_vlc_offs [ c ] ;
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init_vlc ( & dca_smpl_bitalloc [ i + 1 ] . vlc [ j ] , bitalloc_maxbits [ i ] [ j ] ,
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bitalloc_sizes [ i ] ,
bitalloc_bits [ i ] [ j ] , 1 , 1 ,
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bitalloc_codes [ i ] [ j ] , 2 , 2 , INIT_VLC_USE_NEW_STATIC ) ;
c + + ;
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}
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vlcs_initialized = 1 ;
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}
static inline void get_array ( GetBitContext * gb , int * dst , int len , int bits )
{
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while ( len - - )
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* dst + + = get_bits ( gb , bits ) ;
}
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static int dca_parse_audio_coding_header ( DCAContext * s , int base_channel )
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{
int i , j ;
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static const uint8_t adj_table [ 4 ] = { 16 , 18 , 20 , 23 } ;
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static const int bitlen [ 11 ] = { 0 , 1 , 2 , 2 , 2 , 2 , 3 , 3 , 3 , 3 , 3 } ;
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static const int thr [ 11 ] = { 0 , 1 , 3 , 3 , 3 , 3 , 7 , 7 , 7 , 7 , 7 } ;
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s - > audio_header . total_channels = get_bits ( & s - > gb , 3 ) + 1 + base_channel ;
s - > audio_header . prim_channels = s - > audio_header . total_channels ;
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if ( s - > audio_header . prim_channels > DCA_PRIM_CHANNELS_MAX )
s - > audio_header . prim_channels = DCA_PRIM_CHANNELS_MAX ;
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for ( i = base_channel ; i < s - > audio_header . prim_channels ; i + + ) {
s - > audio_header . subband_activity [ i ] = get_bits ( & s - > gb , 5 ) + 2 ;
if ( s - > audio_header . subband_activity [ i ] > DCA_SUBBANDS )
s - > audio_header . subband_activity [ i ] = DCA_SUBBANDS ;
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}
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for ( i = base_channel ; i < s - > audio_header . prim_channels ; i + + ) {
s - > audio_header . vq_start_subband [ i ] = get_bits ( & s - > gb , 5 ) + 1 ;
if ( s - > audio_header . vq_start_subband [ i ] > DCA_SUBBANDS )
s - > audio_header . vq_start_subband [ i ] = DCA_SUBBANDS ;
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}
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get_array ( & s - > gb , s - > audio_header . joint_intensity + base_channel ,
s - > audio_header . prim_channels - base_channel , 3 ) ;
get_array ( & s - > gb , s - > audio_header . transient_huffman + base_channel ,
s - > audio_header . prim_channels - base_channel , 2 ) ;
get_array ( & s - > gb , s - > audio_header . scalefactor_huffman + base_channel ,
s - > audio_header . prim_channels - base_channel , 3 ) ;
get_array ( & s - > gb , s - > audio_header . bitalloc_huffman + base_channel ,
s - > audio_header . prim_channels - base_channel , 3 ) ;
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/* Get codebooks quantization indexes */
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if ( ! base_channel )
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memset ( s - > audio_header . quant_index_huffman , 0 , sizeof ( s - > audio_header . quant_index_huffman ) ) ;
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for ( j = 1 ; j < 11 ; j + + )
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for ( i = base_channel ; i < s - > audio_header . prim_channels ; i + + )
s - > audio_header . quant_index_huffman [ i ] [ j ] = get_bits ( & s - > gb , bitlen [ j ] ) ;
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/* Get scale factor adjustment */
for ( j = 0 ; j < 11 ; j + + )
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for ( i = base_channel ; i < s - > audio_header . prim_channels ; i + + )
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s - > audio_header . scalefactor_adj [ i ] [ j ] = 16 ;
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for ( j = 1 ; j < 11 ; j + + )
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for ( i = base_channel ; i < s - > audio_header . prim_channels ; i + + )
if ( s - > audio_header . quant_index_huffman [ i ] [ j ] < thr [ j ] )
s - > audio_header . scalefactor_adj [ i ] [ j ] = adj_table [ get_bits ( & s - > gb , 2 ) ] ;
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if ( s - > crc_present ) {
/* Audio header CRC check */
get_bits ( & s - > gb , 16 ) ;
}
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s - > current_subframe = 0 ;
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s - > current_subsubframe = 0 ;
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return 0 ;
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}
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static int dca_parse_frame_header ( DCAContext * s )
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{
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init_get_bits ( & s - > gb , s - > dca_buffer , s - > dca_buffer_size * 8 ) ;
/* Sync code */
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skip_bits_long ( & s - > gb , 32 ) ;
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/* Frame header */
s - > frame_type = get_bits ( & s - > gb , 1 ) ;
s - > samples_deficit = get_bits ( & s - > gb , 5 ) + 1 ;
s - > crc_present = get_bits ( & s - > gb , 1 ) ;
s - > sample_blocks = get_bits ( & s - > gb , 7 ) + 1 ;
s - > frame_size = get_bits ( & s - > gb , 14 ) + 1 ;
if ( s - > frame_size < 95 )
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return AVERROR_INVALIDDATA ;
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s - > amode = get_bits ( & s - > gb , 6 ) ;
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s - > sample_rate = avpriv_dca_sample_rates [ get_bits ( & s - > gb , 4 ) ] ;
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if ( ! s - > sample_rate )
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return AVERROR_INVALIDDATA ;
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s - > bit_rate_index = get_bits ( & s - > gb , 5 ) ;
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s - > bit_rate = ff_dca_bit_rates [ s - > bit_rate_index ] ;
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if ( ! s - > bit_rate )
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return AVERROR_INVALIDDATA ;
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skip_bits1 ( & s - > gb ) ; // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
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s - > dynrange = get_bits ( & s - > gb , 1 ) ;
s - > timestamp = get_bits ( & s - > gb , 1 ) ;
s - > aux_data = get_bits ( & s - > gb , 1 ) ;
s - > hdcd = get_bits ( & s - > gb , 1 ) ;
s - > ext_descr = get_bits ( & s - > gb , 3 ) ;
s - > ext_coding = get_bits ( & s - > gb , 1 ) ;
s - > aspf = get_bits ( & s - > gb , 1 ) ;
s - > lfe = get_bits ( & s - > gb , 2 ) ;
s - > predictor_history = get_bits ( & s - > gb , 1 ) ;
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if ( s - > lfe > 2 ) {
av_log ( s - > avctx , AV_LOG_ERROR , " Invalid LFE value: %d \n " , s - > lfe ) ;
return AVERROR_INVALIDDATA ;
}
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/* TODO: check CRC */
if ( s - > crc_present )
s - > header_crc = get_bits ( & s - > gb , 16 ) ;
s - > multirate_inter = get_bits ( & s - > gb , 1 ) ;
s - > version = get_bits ( & s - > gb , 4 ) ;
s - > copy_history = get_bits ( & s - > gb , 2 ) ;
s - > source_pcm_res = get_bits ( & s - > gb , 3 ) ;
s - > front_sum = get_bits ( & s - > gb , 1 ) ;
s - > surround_sum = get_bits ( & s - > gb , 1 ) ;
s - > dialog_norm = get_bits ( & s - > gb , 4 ) ;
/* FIXME: channels mixing levels */
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s - > output = s - > amode ;
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if ( s - > lfe )
s - > output | = DCA_LFE ;
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/* Primary audio coding header */
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s - > audio_header . subframes = get_bits ( & s - > gb , 4 ) + 1 ;
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return dca_parse_audio_coding_header ( s , 0 ) ;
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}
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static inline int get_scale ( GetBitContext * gb , int level , int value , int log2range )
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{
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if ( level < 5 ) {
/* huffman encoded */
value + = get_bitalloc ( gb , & dca_scalefactor , level ) ;
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value = av_clip ( value , 0 , ( 1 < < log2range ) - 1 ) ;
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} else if ( level < 8 ) {
if ( level + 1 > log2range ) {
skip_bits ( gb , level + 1 - log2range ) ;
value = get_bits ( gb , log2range ) ;
} else {
value = get_bits ( gb , level + 1 ) ;
}
}
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return value ;
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}
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static int dca_subframe_header ( DCAContext * s , int base_channel , int block_index )
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{
/* Primary audio coding side information */
int j , k ;
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if ( get_bits_left ( & s - > gb ) < 0 )
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return AVERROR_INVALIDDATA ;
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2010-06-22 10:34:51 +02:00
if ( ! base_channel ) {
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s - > subsubframes [ s - > current_subframe ] = get_bits ( & s - > gb , 2 ) + 1 ;
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s - > partial_samples [ s - > current_subframe ] = get_bits ( & s - > gb , 3 ) ;
}
2015-10-02 17:53:26 +02:00
for ( j = base_channel ; j < s - > audio_header . prim_channels ; j + + ) {
for ( k = 0 ; k < s - > audio_header . subband_activity [ j ] ; k + + )
s - > dca_chan [ j ] . prediction_mode [ k ] = get_bits ( & s - > gb , 1 ) ;
2007-02-27 07:30:40 +01:00
}
/* Get prediction codebook */
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for ( j = base_channel ; j < s - > audio_header . prim_channels ; j + + ) {
for ( k = 0 ; k < s - > audio_header . subband_activity [ j ] ; k + + ) {
if ( s - > dca_chan [ j ] . prediction_mode [ k ] > 0 ) {
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/* (Prediction coefficient VQ address) */
2015-10-02 17:53:26 +02:00
s - > dca_chan [ j ] . prediction_vq [ k ] = get_bits ( & s - > gb , 12 ) ;
2007-02-27 07:30:40 +01:00
}
}
}
/* Bit allocation index */
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for ( j = base_channel ; j < s - > audio_header . prim_channels ; j + + ) {
for ( k = 0 ; k < s - > audio_header . vq_start_subband [ j ] ; k + + ) {
if ( s - > audio_header . bitalloc_huffman [ j ] = = 6 )
s - > dca_chan [ j ] . bitalloc [ k ] = get_bits ( & s - > gb , 5 ) ;
else if ( s - > audio_header . bitalloc_huffman [ j ] = = 5 )
s - > dca_chan [ j ] . bitalloc [ k ] = get_bits ( & s - > gb , 4 ) ;
else if ( s - > audio_header . bitalloc_huffman [ j ] = = 7 ) {
2008-03-06 15:22:02 +01:00
av_log ( s - > avctx , AV_LOG_ERROR ,
" Invalid bit allocation index \n " ) ;
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return AVERROR_INVALIDDATA ;
2008-03-06 15:22:02 +01:00
} else {
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s - > dca_chan [ j ] . bitalloc [ k ] =
get_bitalloc ( & s - > gb , & dca_bitalloc_index , s - > audio_header . bitalloc_huffman [ j ] ) ;
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}
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if ( s - > dca_chan [ j ] . bitalloc [ k ] > 26 ) {
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ff_dlog ( s - > avctx , " bitalloc index [%i][%i] too big (%i) \n " ,
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j , k , s - > dca_chan [ j ] . bitalloc [ k ] ) ;
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return AVERROR_INVALIDDATA ;
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}
}
}
/* Transition mode */
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for ( j = base_channel ; j < s - > audio_header . prim_channels ; j + + ) {
for ( k = 0 ; k < s - > audio_header . subband_activity [ j ] ; k + + ) {
s - > dca_chan [ j ] . transition_mode [ k ] = 0 ;
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if ( s - > subsubframes [ s - > current_subframe ] > 1 & &
2015-10-02 17:53:26 +02:00
k < s - > audio_header . vq_start_subband [ j ] & & s - > dca_chan [ j ] . bitalloc [ k ] > 0 ) {
s - > dca_chan [ j ] . transition_mode [ k ] =
get_bitalloc ( & s - > gb , & dca_tmode , s - > audio_header . transient_huffman [ j ] ) ;
2007-02-27 07:30:40 +01:00
}
}
}
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if ( get_bits_left ( & s - > gb ) < 0 )
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return AVERROR_INVALIDDATA ;
2011-01-10 00:01:01 +01:00
2015-10-02 17:53:26 +02:00
for ( j = base_channel ; j < s - > audio_header . prim_channels ; j + + ) {
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const uint32_t * scale_table ;
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int scale_sum , log_size ;
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2015-10-02 17:53:26 +02:00
memset ( s - > dca_chan [ j ] . scale_factor , 0 ,
s - > audio_header . subband_activity [ j ] * sizeof ( s - > dca_chan [ j ] . scale_factor [ 0 ] [ 0 ] ) * 2 ) ;
2007-02-27 07:30:40 +01:00
2015-10-02 17:53:26 +02:00
if ( s - > audio_header . scalefactor_huffman [ j ] = = 6 ) {
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scale_table = ff_dca_scale_factor_quant7 ;
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log_size = 7 ;
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} else {
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scale_table = ff_dca_scale_factor_quant6 ;
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log_size = 6 ;
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}
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/* When huffman coded, only the difference is encoded */
scale_sum = 0 ;
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for ( k = 0 ; k < s - > audio_header . subband_activity [ j ] ; k + + ) {
if ( k > = s - > audio_header . vq_start_subband [ j ] | | s - > dca_chan [ j ] . bitalloc [ k ] > 0 ) {
scale_sum = get_scale ( & s - > gb , s - > audio_header . scalefactor_huffman [ j ] , scale_sum , log_size ) ;
s - > dca_chan [ j ] . scale_factor [ k ] [ 0 ] = scale_table [ scale_sum ] ;
2007-02-27 07:30:40 +01:00
}
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if ( k < s - > audio_header . vq_start_subband [ j ] & & s - > dca_chan [ j ] . transition_mode [ k ] ) {
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/* Get second scale factor */
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scale_sum = get_scale ( & s - > gb , s - > audio_header . scalefactor_huffman [ j ] , scale_sum , log_size ) ;
s - > dca_chan [ j ] . scale_factor [ k ] [ 1 ] = scale_table [ scale_sum ] ;
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}
}
}
/* Joint subband scale factor codebook select */
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for ( j = base_channel ; j < s - > audio_header . prim_channels ; j + + ) {
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/* Transmitted only if joint subband coding enabled */
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if ( s - > audio_header . joint_intensity [ j ] > 0 )
s - > dca_chan [ j ] . joint_huff = get_bits ( & s - > gb , 3 ) ;
2007-02-27 07:30:40 +01:00
}
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if ( get_bits_left ( & s - > gb ) < 0 )
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return AVERROR_INVALIDDATA ;
2011-01-10 00:01:01 +01:00
2007-02-27 07:30:40 +01:00
/* Scale factors for joint subband coding */
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for ( j = base_channel ; j < s - > audio_header . prim_channels ; j + + ) {
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int source_channel ;
/* Transmitted only if joint subband coding enabled */
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if ( s - > audio_header . joint_intensity [ j ] > 0 ) {
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int scale = 0 ;
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source_channel = s - > audio_header . joint_intensity [ j ] - 1 ;
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/* When huffman coded, only the difference is encoded
* ( is this valid as well for joint scales ? ? ? ) */
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for ( k = s - > audio_header . subband_activity [ j ] ;
k < s - > audio_header . subband_activity [ source_channel ] ; k + + ) {
scale = get_scale ( & s - > gb , s - > dca_chan [ j ] . joint_huff , 64 /* bias */ , 7 ) ;
s - > dca_chan [ j ] . joint_scale_factor [ k ] = scale ; /*joint_scale_table[scale]; */
2007-02-27 07:30:40 +01:00
}
2009-10-10 10:49:57 +02:00
if ( ! ( s - > debug_flag & 0x02 ) ) {
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av_log ( s - > avctx , AV_LOG_DEBUG ,
" Joint stereo coding not supported \n " ) ;
s - > debug_flag | = 0x02 ;
}
}
}
/* Dynamic range coefficient */
2010-08-01 00:15:59 +02:00
if ( ! base_channel & & s - > dynrange )
2007-02-27 07:30:40 +01:00
s - > dynrange_coef = get_bits ( & s - > gb , 8 ) ;
/* Side information CRC check word */
if ( s - > crc_present ) {
get_bits ( & s - > gb , 16 ) ;
}
/*
* Primary audio data arrays
*/
/* VQ encoded high frequency subbands */
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for ( j = base_channel ; j < s - > audio_header . prim_channels ; j + + )
for ( k = s - > audio_header . vq_start_subband [ j ] ; k < s - > audio_header . subband_activity [ j ] ; k + + )
2007-02-27 07:30:40 +01:00
/* 1 vector -> 32 samples */
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s - > dca_chan [ j ] . high_freq_vq [ k ] = get_bits ( & s - > gb , 10 ) ;
2007-02-27 07:30:40 +01:00
/* Low frequency effect data */
2010-06-22 10:34:51 +02:00
if ( ! base_channel & & s - > lfe ) {
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/* LFE samples */
2014-09-08 19:40:26 +02:00
int lfe_samples = 2 * s - > lfe * ( 4 + block_index ) ;
2010-06-22 10:34:51 +02:00
int lfe_end_sample = 2 * s - > lfe * ( 4 + block_index + s - > subsubframes [ s - > current_subframe ] ) ;
2007-02-27 07:30:40 +01:00
float lfe_scale ;
2010-06-22 10:33:00 +02:00
for ( j = lfe_samples ; j < lfe_end_sample ; j + + ) {
2007-02-27 07:30:40 +01:00
/* Signed 8 bits int */
s - > lfe_data [ j ] = get_sbits ( & s - > gb , 8 ) ;
}
/* Scale factor index */
2012-02-29 03:11:59 +01:00
skip_bits ( & s - > gb , 1 ) ;
2015-02-15 20:03:03 +01:00
s - > lfe_scale_factor = ff_dca_scale_factor_quant7 [ get_bits ( & s - > gb , 7 ) ] ;
2007-02-27 07:30:40 +01:00
/* Quantization step size * scale factor */
lfe_scale = 0.035 * s - > lfe_scale_factor ;
2010-06-22 10:33:00 +02:00
for ( j = lfe_samples ; j < lfe_end_sample ; j + + )
2007-02-27 07:30:40 +01:00
s - > lfe_data [ j ] * = lfe_scale ;
}
return 0 ;
}
2012-01-03 11:49:28 +01:00
static void qmf_32_subbands ( DCAContext * s , int chans ,
2015-12-30 15:29:03 +01:00
float samples_in [ DCA_SUBBANDS ] [ SAMPLES_PER_SUBBAND ] , float * samples_out ,
2011-01-27 16:20:43 +01:00
float scale )
2007-02-27 07:30:40 +01:00
{
2007-12-02 22:12:43 +01:00
const float * prCoeff ;
2007-02-27 07:30:40 +01:00
2015-10-02 17:53:26 +02:00
int sb_act = s - > audio_header . subband_activity [ chans ] ;
2007-02-27 07:30:40 +01:00
2012-01-03 11:49:28 +01:00
scale * = sqrt ( 1 / 8.0 ) ;
2007-02-27 07:30:40 +01:00
/* Select filter */
if ( ! s - > multirate_inter ) /* Non-perfect reconstruction */
2015-02-15 20:03:03 +01:00
prCoeff = ff_dca_fir_32bands_nonperfect ;
2007-02-27 07:30:40 +01:00
else /* Perfect reconstruction */
2015-02-15 20:03:03 +01:00
prCoeff = ff_dca_fir_32bands_perfect ;
2007-02-27 07:30:40 +01:00
2013-07-15 19:28:16 +02:00
s - > dcadsp . qmf_32_subbands ( samples_in , sb_act , & s - > synth , & s - > imdct ,
2015-10-02 17:53:26 +02:00
s - > dca_chan [ chans ] . subband_fir_hist ,
& s - > dca_chan [ chans ] . hist_index ,
s - > dca_chan [ chans ] . subband_fir_noidea , prCoeff ,
2013-07-15 19:28:16 +02:00
samples_out , s - > raXin , scale ) ;
2007-02-27 07:30:40 +01:00
}
2014-02-11 11:45:27 +01:00
static QMF64_table * qmf64_precompute ( void )
{
unsigned i , j ;
QMF64_table * table = av_malloc ( sizeof ( * table ) ) ;
if ( ! table )
return NULL ;
for ( i = 0 ; i < 32 ; i + + )
for ( j = 0 ; j < 32 ; j + + )
table - > dct4_coeff [ i ] [ j ] = cos ( ( 2 * i + 1 ) * ( 2 * j + 1 ) * M_PI / 128 ) ;
for ( i = 0 ; i < 32 ; i + + )
for ( j = 0 ; j < 32 ; j + + )
table - > dct2_coeff [ i ] [ j ] = cos ( ( 2 * i + 1 ) * j * M_PI / 64 ) ;
/* FIXME: Is the factor 0.125 = 1/8 right? */
for ( i = 0 ; i < 32 ; i + + )
table - > rcos [ i ] = 0.125 / cos ( ( 2 * i + 1 ) * M_PI / 256 ) ;
for ( i = 0 ; i < 32 ; i + + )
table - > rsin [ i ] = - 0.125 / sin ( ( 2 * i + 1 ) * M_PI / 256 ) ;
return table ;
}
/* FIXME: Totally unoptimized. Based on the reference code and
* http : //multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
* for doubling the size . */
2015-12-30 15:29:03 +01:00
static void qmf_64_subbands ( DCAContext * s , int chans ,
float samples_in [ DCA_SUBBANDS_X96K ] [ SAMPLES_PER_SUBBAND ] ,
2014-02-11 11:45:27 +01:00
float * samples_out , float scale )
{
float raXin [ 64 ] ;
float A [ 32 ] , B [ 32 ] ;
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float * raX = s - > dca_chan [ chans ] . subband_fir_hist ;
float * raZ = s - > dca_chan [ chans ] . subband_fir_noidea ;
2014-02-11 11:45:27 +01:00
unsigned i , j , k , subindex ;
2015-12-30 15:29:03 +01:00
for ( i = s - > audio_header . subband_activity [ chans ] ; i < DCA_SUBBANDS_X96K ; i + + )
2014-02-11 11:45:27 +01:00
raXin [ i ] = 0.0 ;
2015-09-18 13:34:20 +02:00
for ( subindex = 0 ; subindex < SAMPLES_PER_SUBBAND ; subindex + + ) {
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for ( i = 0 ; i < s - > audio_header . subband_activity [ chans ] ; i + + )
2014-02-11 11:45:27 +01:00
raXin [ i ] = samples_in [ i ] [ subindex ] ;
for ( k = 0 ; k < 32 ; k + + ) {
A [ k ] = 0.0 ;
for ( i = 0 ; i < 32 ; i + + )
A [ k ] + = ( raXin [ 2 * i ] + raXin [ 2 * i + 1 ] ) * s - > qmf64_table - > dct4_coeff [ k ] [ i ] ;
}
for ( k = 0 ; k < 32 ; k + + ) {
B [ k ] = raXin [ 0 ] * s - > qmf64_table - > dct2_coeff [ k ] [ 0 ] ;
for ( i = 1 ; i < 32 ; i + + )
B [ k ] + = ( raXin [ 2 * i ] + raXin [ 2 * i - 1 ] ) * s - > qmf64_table - > dct2_coeff [ k ] [ i ] ;
}
for ( k = 0 ; k < 32 ; k + + ) {
raX [ k ] = s - > qmf64_table - > rcos [ k ] * ( A [ k ] + B [ k ] ) ;
raX [ 63 - k ] = s - > qmf64_table - > rsin [ k ] * ( A [ k ] - B [ k ] ) ;
}
2015-12-30 15:29:03 +01:00
for ( i = 0 ; i < DCA_SUBBANDS_X96K ; i + + ) {
2014-02-11 11:45:27 +01:00
float out = raZ [ i ] ;
for ( j = 0 ; j < 1024 ; j + = 128 )
out + = ff_dca_fir_64bands [ j + i ] * ( raX [ j + i ] - raX [ j + 63 - i ] ) ;
* samples_out + + = out * scale ;
}
2015-12-30 15:29:03 +01:00
for ( i = 0 ; i < DCA_SUBBANDS_X96K ; i + + ) {
2014-02-11 11:45:27 +01:00
float hist = 0.0 ;
for ( j = 0 ; j < 1024 ; j + = 128 )
hist + = ff_dca_fir_64bands [ 64 + j + i ] * ( - raX [ i + j ] - raX [ j + 63 - i ] ) ;
raZ [ i ] = hist ;
}
/* FIXME: Make buffer circular, to avoid this move. */
memmove ( raX + 64 , raX , ( 1024 - 64 ) * sizeof ( * raX ) ) ;
}
}
static void lfe_interpolation_fir ( DCAContext * s , const float * samples_in ,
2014-02-14 16:03:06 +01:00
float * samples_out )
2007-02-27 07:30:40 +01:00
{
/* samples_in: An array holding decimated samples.
* Samples in current subframe starts from samples_in [ 0 ] ,
* while samples_in [ - 1 ] , samples_in [ - 2 ] , . . . , stores samples
* from last subframe as history .
*
* samples_out : An array holding interpolated samples
*/
2014-02-06 00:40:52 +01:00
int idx ;
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const float * prCoeff ;
int deciindex ;
/* Select decimation filter */
2014-02-11 11:45:27 +01:00
if ( s - > lfe = = 1 ) {
2014-09-08 19:40:26 +02:00
idx = 1 ;
2015-02-15 20:03:03 +01:00
prCoeff = ff_dca_lfe_fir_128 ;
2007-02-27 07:30:40 +01:00
} else {
2014-02-11 11:45:27 +01:00
idx = 0 ;
if ( s - > exss_ext_mask & DCA_EXT_EXSS_XLL )
prCoeff = ff_dca_lfe_xll_fir_64 ;
else
prCoeff = ff_dca_lfe_fir_64 ;
2007-02-27 07:30:40 +01:00
}
/* Interpolation */
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for ( deciindex = 0 ; deciindex < 2 * s - > lfe ; deciindex + + ) {
2014-02-14 16:03:06 +01:00
s - > dcadsp . lfe_fir [ idx ] ( samples_out , samples_in , prCoeff ) ;
2010-04-12 13:14:55 +02:00
samples_in + + ;
2014-02-06 00:40:52 +01:00
samples_out + = 2 * 32 * ( 1 + idx ) ;
2007-02-27 07:30:40 +01:00
}
}
/* downmixing routines */
2012-08-27 17:43:34 +02:00
# define MIX_REAR1(samples, s1, rs, coef) \
samples [ 0 ] [ i ] + = samples [ s1 ] [ i ] * coef [ rs ] [ 0 ] ; \
samples [ 1 ] [ i ] + = samples [ s1 ] [ i ] * coef [ rs ] [ 1 ] ;
2012-01-03 11:49:28 +01:00
2012-08-27 17:43:34 +02:00
# define MIX_REAR2(samples, s1, s2, rs, coef) \
samples [ 0 ] [ i ] + = samples [ s1 ] [ i ] * coef [ rs ] [ 0 ] + samples [ s2 ] [ i ] * coef [ rs + 1 ] [ 0 ] ; \
samples [ 1 ] [ i ] + = samples [ s1 ] [ i ] * coef [ rs ] [ 1 ] + samples [ s2 ] [ i ] * coef [ rs + 1 ] [ 1 ] ;
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# define MIX_FRONT3(samples, coef) \
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t = samples [ c ] [ i ] ; \
u = samples [ l ] [ i ] ; \
v = samples [ r ] [ i ] ; \
samples [ 0 ] [ i ] = t * coef [ 0 ] [ 0 ] + u * coef [ 1 ] [ 0 ] + v * coef [ 2 ] [ 0 ] ; \
samples [ 1 ] [ i ] = t * coef [ 0 ] [ 1 ] + u * coef [ 1 ] [ 1 ] + v * coef [ 2 ] [ 1 ] ;
2007-02-27 07:30:40 +01:00
2012-01-03 11:49:28 +01:00
# define DOWNMIX_TO_STEREO(op1, op2) \
for ( i = 0 ; i < 256 ; i + + ) { \
op1 \
op2 \
2007-02-27 07:30:40 +01:00
}
2013-11-22 23:47:03 +01:00
static void dca_downmix ( float * * samples , int srcfmt , int lfe_present ,
float coef [ DCA_PRIM_CHANNELS_MAX + 1 ] [ 2 ] ,
2011-01-27 16:20:43 +01:00
const int8_t * channel_mapping )
2007-02-27 07:30:40 +01:00
{
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int c , l , r , sl , sr , s ;
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int i ;
2010-07-27 23:45:43 +02:00
float t , u , v ;
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switch ( srcfmt ) {
case DCA_MONO :
case DCA_4F2R :
av_log ( NULL , 0 , " Not implemented! \n " ) ;
break ;
2013-11-22 22:59:54 +01:00
case DCA_CHANNEL :
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case DCA_STEREO :
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case DCA_STEREO_TOTAL :
case DCA_STEREO_SUMDIFF :
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break ;
case DCA_3F :
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c = channel_mapping [ 0 ] ;
l = channel_mapping [ 1 ] ;
r = channel_mapping [ 2 ] ;
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DOWNMIX_TO_STEREO ( MIX_FRONT3 ( samples , coef ) , ) ;
2007-02-27 07:30:40 +01:00
break ;
case DCA_2F1R :
2012-08-27 17:43:34 +02:00
s = channel_mapping [ 2 ] ;
DOWNMIX_TO_STEREO ( MIX_REAR1 ( samples , s , 2 , coef ) , ) ;
2007-02-27 07:30:40 +01:00
break ;
case DCA_3F1R :
2012-08-27 17:43:34 +02:00
c = channel_mapping [ 0 ] ;
l = channel_mapping [ 1 ] ;
r = channel_mapping [ 2 ] ;
s = channel_mapping [ 3 ] ;
2007-05-02 22:33:32 +02:00
DOWNMIX_TO_STEREO ( MIX_FRONT3 ( samples , coef ) ,
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MIX_REAR1 ( samples , s , 3 , coef ) ) ;
2007-02-27 07:30:40 +01:00
break ;
case DCA_2F2R :
2012-08-27 17:43:34 +02:00
sl = channel_mapping [ 2 ] ;
sr = channel_mapping [ 3 ] ;
DOWNMIX_TO_STEREO ( MIX_REAR2 ( samples , sl , sr , 2 , coef ) , ) ;
2007-02-27 07:30:40 +01:00
break ;
case DCA_3F2R :
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c = channel_mapping [ 0 ] ;
l = channel_mapping [ 1 ] ;
r = channel_mapping [ 2 ] ;
sl = channel_mapping [ 3 ] ;
sr = channel_mapping [ 4 ] ;
2007-05-02 22:33:32 +02:00
DOWNMIX_TO_STEREO ( MIX_FRONT3 ( samples , coef ) ,
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MIX_REAR2 ( samples , sl , sr , 3 , coef ) ) ;
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break ;
}
2013-11-22 23:47:03 +01:00
if ( lfe_present ) {
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int lf_buf = ff_dca_lfe_index [ srcfmt ] ;
int lf_idx = ff_dca_channels [ srcfmt ] ;
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for ( i = 0 ; i < 256 ; i + + ) {
samples [ 0 ] [ i ] + = samples [ lf_buf ] [ i ] * coef [ lf_idx ] [ 0 ] ;
samples [ 1 ] [ i ] + = samples [ lf_buf ] [ i ] * coef [ lf_idx ] [ 1 ] ;
}
}
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}
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# ifndef decode_blockcodes
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/* Very compact version of the block code decoder that does not use table
* look - up but is slightly slower */
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static int decode_blockcode ( int code , int levels , int32_t * values )
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{
int i ;
int offset = ( levels - 1 ) > > 1 ;
for ( i = 0 ; i < 4 ; i + + ) {
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int div = FASTDIV ( code , levels ) ;
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values [ i ] = code - offset - div * levels ;
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code = div ;
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}
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return code ;
}
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static int decode_blockcodes ( int code1 , int code2 , int levels , int32_t * values )
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{
return decode_blockcode ( code1 , levels , values ) |
decode_blockcode ( code2 , levels , values + 4 ) ;
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}
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# endif
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static const uint8_t abits_sizes [ 7 ] = { 7 , 10 , 12 , 13 , 15 , 17 , 19 } ;
static const uint8_t abits_levels [ 7 ] = { 3 , 5 , 7 , 9 , 13 , 17 , 25 } ;
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static int dca_subsubframe ( DCAContext * s , int base_channel , int block_index )
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{
int k , l ;
int subsubframe = s - > current_subsubframe ;
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const uint32_t * quant_step_table ;
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/*
* Audio data
*/
/* Select quantization step size table */
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if ( s - > bit_rate_index = = 0x1f )
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quant_step_table = ff_dca_lossless_quant ;
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else
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quant_step_table = ff_dca_lossy_quant ;
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for ( k = base_channel ; k < s - > audio_header . prim_channels ; k + + ) {
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int32_t ( * subband_samples ) [ 8 ] = s - > dca_chan [ k ] . subband_samples [ block_index ] ;
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if ( get_bits_left ( & s - > gb ) < 0 )
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return AVERROR_INVALIDDATA ;
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for ( l = 0 ; l < s - > audio_header . vq_start_subband [ k ] ; l + + ) {
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int m ;
/* Select the mid-tread linear quantizer */
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int abits = s - > dca_chan [ k ] . bitalloc [ l ] ;
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uint32_t quant_step_size = quant_step_table [ abits ] ;
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/*
* Extract bits from the bit stream
*/
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if ( ! abits )
memset ( subband_samples [ l ] , 0 , SAMPLES_PER_SUBBAND *
sizeof ( subband_samples [ l ] [ 0 ] ) ) ;
else {
uint32_t rscale ;
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/* Deal with transients */
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int sfi = s - > dca_chan [ k ] . transition_mode [ l ] & &
subsubframe > = s - > dca_chan [ k ] . transition_mode [ l ] ;
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/* Determine quantization index code book and its type.
Select quantization index code book */
int sel = s - > audio_header . quant_index_huffman [ k ] [ abits ] ;
rscale = ( s - > dca_chan [ k ] . scale_factor [ l ] [ sfi ] *
s - > audio_header . scalefactor_adj [ k ] [ sel ] + 8 ) > > 4 ;
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if ( abits > = 11 | | ! dca_smpl_bitalloc [ abits ] . vlc [ sel ] . table ) {
if ( abits < = 7 ) {
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/* Block code */
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int block_code1 , block_code2 , size , levels , err ;
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size = abits_sizes [ abits - 1 ] ;
levels = abits_levels [ abits - 1 ] ;
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block_code1 = get_bits ( & s - > gb , size ) ;
block_code2 = get_bits ( & s - > gb , size ) ;
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err = decode_blockcodes ( block_code1 , block_code2 ,
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levels , subband_samples [ l ] ) ;
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if ( err ) {
av_log ( s - > avctx , AV_LOG_ERROR ,
" ERROR: block code look-up failed \n " ) ;
return AVERROR_INVALIDDATA ;
}
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} else {
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/* no coding */
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for ( m = 0 ; m < SAMPLES_PER_SUBBAND ; m + + )
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subband_samples [ l ] [ m ] = get_sbits ( & s - > gb , abits - 3 ) ;
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}
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} else {
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/* Huffman coded */
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for ( m = 0 ; m < SAMPLES_PER_SUBBAND ; m + + )
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subband_samples [ l ] [ m ] = get_bitalloc ( & s - > gb ,
& dca_smpl_bitalloc [ abits ] , sel ) ;
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}
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s - > dcadsp . dequantize ( subband_samples [ l ] , quant_step_size , rscale ) ;
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}
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}
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for ( l = 0 ; l < s - > audio_header . vq_start_subband [ k ] ; l + + ) {
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int m ;
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/*
* Inverse ADPCM if in prediction mode
*/
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if ( s - > dca_chan [ k ] . prediction_mode [ l ] ) {
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int n ;
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if ( s - > predictor_history )
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subband_samples [ l ] [ 0 ] + = ( ff_dca_adpcm_vb [ s - > dca_chan [ k ] . prediction_vq [ l ] ] [ 0 ] *
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( int64_t ) s - > dca_chan [ k ] . subband_samples_hist [ l ] [ 3 ] +
ff_dca_adpcm_vb [ s - > dca_chan [ k ] . prediction_vq [ l ] ] [ 1 ] *
( int64_t ) s - > dca_chan [ k ] . subband_samples_hist [ l ] [ 2 ] +
ff_dca_adpcm_vb [ s - > dca_chan [ k ] . prediction_vq [ l ] ] [ 2 ] *
( int64_t ) s - > dca_chan [ k ] . subband_samples_hist [ l ] [ 1 ] +
ff_dca_adpcm_vb [ s - > dca_chan [ k ] . prediction_vq [ l ] ] [ 3 ] *
( int64_t ) s - > dca_chan [ k ] . subband_samples_hist [ l ] [ 0 ] ) +
( 1 < < 12 ) > > 13 ;
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for ( m = 1 ; m < SAMPLES_PER_SUBBAND ; m + + ) {
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int64_t sum = ff_dca_adpcm_vb [ s - > dca_chan [ k ] . prediction_vq [ l ] ] [ 0 ] *
( int64_t ) subband_samples [ l ] [ m - 1 ] ;
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for ( n = 2 ; n < = 4 ; n + + )
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if ( m > = n )
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sum + = ff_dca_adpcm_vb [ s - > dca_chan [ k ] . prediction_vq [ l ] ] [ n - 1 ] *
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( int64_t ) subband_samples [ l ] [ m - n ] ;
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else if ( s - > predictor_history )
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sum + = ff_dca_adpcm_vb [ s - > dca_chan [ k ] . prediction_vq [ l ] ] [ n - 1 ] *
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( int64_t ) s - > dca_chan [ k ] . subband_samples_hist [ l ] [ m - n + 4 ] ;
subband_samples [ l ] [ m ] + = ( int32_t ) ( sum + ( 1 < < 12 ) > > 13 ) ;
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}
}
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}
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/* Backup predictor history for adpcm */
for ( l = 0 ; l < DCA_SUBBANDS ; l + + )
AV_COPY128 ( s - > dca_chan [ k ] . subband_samples_hist [ l ] , & subband_samples [ l ] [ 4 ] ) ;
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/*
* Decode VQ encoded high frequencies
*/
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if ( s - > audio_header . subband_activity [ k ] > s - > audio_header . vq_start_subband [ k ] ) {
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if ( ! s - > debug_flag & 0x01 ) {
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av_log ( s - > avctx , AV_LOG_DEBUG ,
" Stream with high frequencies VQ coding \n " ) ;
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s - > debug_flag | = 0x01 ;
}
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s - > dcadsp . decode_hf ( subband_samples , s - > dca_chan [ k ] . high_freq_vq ,
ff_dca_high_freq_vq ,
subsubframe * SAMPLES_PER_SUBBAND ,
s - > dca_chan [ k ] . scale_factor ,
s - > audio_header . vq_start_subband [ k ] ,
s - > audio_header . subband_activity [ k ] ) ;
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}
}
/* Check for DSYNC after subsubframe */
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if ( s - > aspf | | subsubframe = = s - > subsubframes [ s - > current_subframe ] - 1 ) {
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if ( get_bits ( & s - > gb , 16 ) ! = 0xFFFF ) {
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av_log ( s - > avctx , AV_LOG_ERROR , " Didn't get subframe DSYNC \n " ) ;
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return AVERROR_INVALIDDATA ;
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}
}
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return 0 ;
}
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static int dca_filter_channels ( DCAContext * s , int block_index , int upsample )
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{
int k ;
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if ( upsample ) {
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LOCAL_ALIGNED ( 32 , float , samples , [ DCA_SUBBANDS_X96K ] , [ SAMPLES_PER_SUBBAND ] ) ;
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if ( ! s - > qmf64_table ) {
s - > qmf64_table = qmf64_precompute ( ) ;
if ( ! s - > qmf64_table )
return AVERROR ( ENOMEM ) ;
}
/* 64 subbands QMF */
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for ( k = 0 ; k < s - > audio_header . prim_channels ; k + + ) {
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int32_t ( * subband_samples ) [ SAMPLES_PER_SUBBAND ] =
s - > dca_chan [ k ] . subband_samples [ block_index ] ;
s - > fmt_conv . int32_to_float ( samples [ 0 ] , subband_samples [ 0 ] ,
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DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND ) ;
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if ( s - > channel_order_tab [ k ] > = 0 )
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qmf_64_subbands ( s , k , samples ,
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s - > samples_chanptr [ s - > channel_order_tab [ k ] ] ,
/* Upsampling needs a factor 2 here. */
M_SQRT2 / 32768.0 ) ;
}
} else {
/* 32 subbands QMF */
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LOCAL_ALIGNED ( 32 , float , samples , [ DCA_SUBBANDS ] , [ SAMPLES_PER_SUBBAND ] ) ;
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for ( k = 0 ; k < s - > audio_header . prim_channels ; k + + ) {
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int32_t ( * subband_samples ) [ SAMPLES_PER_SUBBAND ] =
s - > dca_chan [ k ] . subband_samples [ block_index ] ;
s - > fmt_conv . int32_to_float ( samples [ 0 ] , subband_samples [ 0 ] ,
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DCA_SUBBANDS * SAMPLES_PER_SUBBAND ) ;
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if ( s - > channel_order_tab [ k ] > = 0 )
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qmf_32_subbands ( s , k , samples ,
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s - > samples_chanptr [ s - > channel_order_tab [ k ] ] ,
M_SQRT1_2 / 32768.0 ) ;
}
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}
/* Generate LFE samples for this subsubframe FIXME!!! */
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if ( s - > lfe ) {
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float * samples = s - > samples_chanptr [ ff_dca_lfe_index [ s - > amode ] ] ;
lfe_interpolation_fir ( s ,
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s - > lfe_data + 2 * s - > lfe * ( block_index + 4 ) ,
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samples ) ;
if ( upsample ) {
unsigned i ;
/* Should apply the filter in Table 6-11 when upsampling. For
* now , just duplicate . */
for ( i = 511 ; i > 0 ; i - - ) {
samples [ 2 * i ] =
samples [ 2 * i + 1 ] = samples [ i ] ;
}
samples [ 1 ] = samples [ 0 ] ;
}
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}
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/* FIXME: This downmixing is probably broken with upsample.
* Probably totally broken also with XLL in general . */
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/* Downmixing to Stereo */
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if ( s - > audio_header . prim_channels + ! ! s - > lfe > 2 & &
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s - > avctx - > request_channel_layout = = AV_CH_LAYOUT_STEREO ) {
dca_downmix ( s - > samples_chanptr , s - > amode , ! ! s - > lfe , s - > downmix_coef ,
s - > channel_order_tab ) ;
}
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return 0 ;
}
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static int dca_subframe_footer ( DCAContext * s , int base_channel )
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{
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int in , out , aux_data_count , aux_data_end , reserved ;
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uint32_t nsyncaux ;
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/*
* Unpack optional information
*/
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/* presumably optional information only appears in the core? */
if ( ! base_channel ) {
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if ( s - > timestamp )
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skip_bits_long ( & s - > gb , 32 ) ;
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if ( s - > aux_data ) {
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aux_data_count = get_bits ( & s - > gb , 6 ) ;
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// align (32-bit)
skip_bits_long ( & s - > gb , ( - get_bits_count ( & s - > gb ) ) & 31 ) ;
aux_data_end = 8 * aux_data_count + get_bits_count ( & s - > gb ) ;
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if ( ( nsyncaux = get_bits_long ( & s - > gb , 32 ) ) ! = DCA_NSYNCAUX ) {
av_log ( s - > avctx , AV_LOG_ERROR , " nSYNCAUX mismatch %# " PRIx32 " \n " ,
nsyncaux ) ;
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return AVERROR_INVALIDDATA ;
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}
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if ( get_bits1 ( & s - > gb ) ) { // bAUXTimeStampFlag
avpriv_request_sample ( s - > avctx ,
" Auxiliary Decode Time Stamp Flag " ) ;
// align (4-bit)
skip_bits ( & s - > gb , ( - get_bits_count ( & s - > gb ) ) & 4 ) ;
// 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
skip_bits_long ( & s - > gb , 44 ) ;
}
if ( ( s - > core_downmix = get_bits1 ( & s - > gb ) ) ) {
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int am = get_bits ( & s - > gb , 3 ) ;
switch ( am ) {
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case 0 :
s - > core_downmix_amode = DCA_MONO ;
break ;
case 1 :
s - > core_downmix_amode = DCA_STEREO ;
break ;
case 2 :
s - > core_downmix_amode = DCA_STEREO_TOTAL ;
break ;
case 3 :
s - > core_downmix_amode = DCA_3F ;
break ;
case 4 :
s - > core_downmix_amode = DCA_2F1R ;
break ;
case 5 :
s - > core_downmix_amode = DCA_2F2R ;
break ;
case 6 :
s - > core_downmix_amode = DCA_3F1R ;
break ;
default :
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av_log ( s - > avctx , AV_LOG_ERROR ,
" Invalid mode %d for embedded downmix coefficients \n " ,
am ) ;
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return AVERROR_INVALIDDATA ;
}
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for ( out = 0 ; out < ff_dca_channels [ s - > core_downmix_amode ] ; out + + ) {
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for ( in = 0 ; in < s - > audio_header . prim_channels + ! ! s - > lfe ; in + + ) {
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uint16_t tmp = get_bits ( & s - > gb , 9 ) ;
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if ( ( tmp & 0xFF ) > 241 ) {
av_log ( s - > avctx , AV_LOG_ERROR ,
" Invalid downmix coefficient code % " PRIu16 " \n " ,
tmp ) ;
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return AVERROR_INVALIDDATA ;
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}
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s - > core_downmix_codes [ in ] [ out ] = tmp ;
}
}
}
align_get_bits ( & s - > gb ) ; // byte align
skip_bits ( & s - > gb , 16 ) ; // nAUXCRC16
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/*
* additional data ( reserved , cf . ETSI TS 102 114 V1 .4 .1 )
*
* Note : don ' t check for overreads , aux_data_count can ' t be trusted .
*/
if ( ( reserved = ( aux_data_end - get_bits_count ( & s - > gb ) ) ) > 0 ) {
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avpriv_request_sample ( s - > avctx ,
" Core auxiliary data reserved content " ) ;
skip_bits_long ( & s - > gb , reserved ) ;
}
}
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if ( s - > crc_present & & s - > dynrange )
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get_bits ( & s - > gb , 16 ) ;
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}
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return 0 ;
}
/**
* Decode a dca frame block
*
* @ param s pointer to the DCAContext
*/
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static int dca_decode_block ( DCAContext * s , int base_channel , int block_index )
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{
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int ret ;
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/* Sanity check */
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if ( s - > current_subframe > = s - > audio_header . subframes ) {
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av_log ( s - > avctx , AV_LOG_DEBUG , " check failed: %i>%i " ,
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s - > current_subframe , s - > audio_header . subframes ) ;
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return AVERROR_INVALIDDATA ;
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}
if ( ! s - > current_subsubframe ) {
/* Read subframe header */
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if ( ( ret = dca_subframe_header ( s , base_channel , block_index ) ) )
return ret ;
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}
/* Read subsubframe */
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if ( ( ret = dca_subsubframe ( s , base_channel , block_index ) ) )
return ret ;
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/* Update state */
s - > current_subsubframe + + ;
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if ( s - > current_subsubframe > = s - > subsubframes [ s - > current_subframe ] ) {
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s - > current_subsubframe = 0 ;
s - > current_subframe + + ;
}
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if ( s - > current_subframe > = s - > audio_header . subframes ) {
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/* Read subframe footer */
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if ( ( ret = dca_subframe_footer ( s , base_channel ) ) )
return ret ;
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}
return 0 ;
}
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static float dca_dmix_code ( unsigned code )
{
int sign = ( code > > 8 ) - 1 ;
code & = 0xff ;
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return ( ( ff_dca_dmixtable [ code ] ^ sign ) - sign ) * ( 1.0 / ( 1U < < 15 ) ) ;
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}
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static int scan_for_extensions ( AVCodecContext * avctx )
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{
DCAContext * s = avctx - > priv_data ;
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int core_ss_end , ret = 0 ;
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2011-01-06 23:34:12 +01:00
core_ss_end = FFMIN ( s - > frame_size , s - > dca_buffer_size ) * 8 ;
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/* only scan for extensions if ext_descr was unknown or indicated a
* supported XCh extension */
if ( s - > core_ext_mask < 0 | | s - > core_ext_mask & DCA_EXT_XCH ) {
/* if ext_descr was unknown, clear s->core_ext_mask so that the
* extensions scan can fill it up */
s - > core_ext_mask = FFMAX ( s - > core_ext_mask , 0 ) ;
/* extensions start at 32-bit boundaries into bitstream */
skip_bits_long ( & s - > gb , ( - get_bits_count ( & s - > gb ) ) & 31 ) ;
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while ( core_ss_end - get_bits_count ( & s - > gb ) > = 32 ) {
uint32_t bits = get_bits_long ( & s - > gb , 32 ) ;
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int i ;
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switch ( bits ) {
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case DCA_SYNCWORD_XCH : {
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int ext_amode , xch_fsize ;
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s - > xch_base_channel = s - > audio_header . prim_channels ;
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/* validate sync word using XCHFSIZE field */
xch_fsize = show_bits ( & s - > gb , 10 ) ;
if ( ( s - > frame_size ! = ( get_bits_count ( & s - > gb ) > > 3 ) - 4 + xch_fsize ) & &
( s - > frame_size ! = ( get_bits_count ( & s - > gb ) > > 3 ) - 4 + xch_fsize + 1 ) )
continue ;
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/* skip length-to-end-of-frame field for the moment */
skip_bits ( & s - > gb , 10 ) ;
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s - > core_ext_mask | = DCA_EXT_XCH ;
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/* extension amode(number of channels in extension) should be 1 */
/* AFAIK XCh is not used for more channels */
if ( ( ext_amode = get_bits ( & s - > gb , 4 ) ) ! = 1 ) {
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av_log ( avctx , AV_LOG_ERROR ,
" XCh extension amode %d not supported! \n " ,
ext_amode ) ;
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continue ;
}
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/* much like core primary audio coding header */
dca_parse_audio_coding_header ( s , s - > xch_base_channel ) ;
for ( i = 0 ; i < ( s - > sample_blocks / 8 ) ; i + + )
if ( ( ret = dca_decode_block ( s , s - > xch_base_channel , i ) ) ) {
av_log ( avctx , AV_LOG_ERROR , " error decoding XCh extension \n " ) ;
continue ;
}
s - > xch_present = 1 ;
break ;
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}
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case DCA_SYNCWORD_XXCH :
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/* XXCh: extended channels */
/* usually found either in core or HD part in DTS-HD HRA streams,
* but not in DTS - ES which contains XCh extensions instead */
s - > core_ext_mask | = DCA_EXT_XXCH ;
break ;
case 0x1d95f262 : {
int fsize96 = show_bits ( & s - > gb , 12 ) + 1 ;
if ( s - > frame_size ! = ( get_bits_count ( & s - > gb ) > > 3 ) - 4 + fsize96 )
continue ;
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av_log ( avctx , AV_LOG_DEBUG , " X96 extension found at %d bits \n " ,
get_bits_count ( & s - > gb ) ) ;
skip_bits ( & s - > gb , 12 ) ;
av_log ( avctx , AV_LOG_DEBUG , " FSIZE96 = %d bytes \n " , fsize96 ) ;
av_log ( avctx , AV_LOG_DEBUG , " REVNO = %d \n " , get_bits ( & s - > gb , 4 ) ) ;
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s - > core_ext_mask | = DCA_EXT_X96 ;
break ;
}
}
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skip_bits_long ( & s - > gb , ( - get_bits_count ( & s - > gb ) ) & 31 ) ;
}
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} else {
/* no supported extensions, skip the rest of the core substream */
skip_bits_long ( & s - > gb , core_ss_end - get_bits_count ( & s - > gb ) ) ;
}
if ( s - > core_ext_mask & DCA_EXT_X96 )
s - > profile = FF_PROFILE_DTS_96_24 ;
else if ( s - > core_ext_mask & ( DCA_EXT_XCH | DCA_EXT_XXCH ) )
s - > profile = FF_PROFILE_DTS_ES ;
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/* check for ExSS (HD part) */
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if ( s - > dca_buffer_size - s - > frame_size > 32 & &
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get_bits_long ( & s - > gb , 32 ) = = DCA_SYNCWORD_SUBSTREAM )
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ff_dca_exss_parse_header ( s ) ;
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return ret ;
}
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static int set_channel_layout ( AVCodecContext * avctx , int channels , int num_core_channels )
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{
DCAContext * s = avctx - > priv_data ;
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int i ;
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if ( s - > amode < 16 ) {
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avctx - > channel_layout = dca_core_channel_layout [ s - > amode ] ;
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if ( s - > audio_header . prim_channels + ! ! s - > lfe > 2 & &
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avctx - > request_channel_layout = = AV_CH_LAYOUT_STEREO ) {
/*
* Neither the core ' s auxiliary data nor our default tables contain
* downmix coefficients for the additional channel coded in the XCh
* extension , so when we ' re doing a Stereo downmix , don ' t decode it .
*/
s - > xch_disable = 1 ;
}
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if ( s - > xch_present & & ! s - > xch_disable ) {
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avctx - > channel_layout | = AV_CH_BACK_CENTER ;
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if ( s - > lfe ) {
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avctx - > channel_layout | = AV_CH_LOW_FREQUENCY ;
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s - > channel_order_tab = ff_dca_channel_reorder_lfe_xch [ s - > amode ] ;
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} else {
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s - > channel_order_tab = ff_dca_channel_reorder_nolfe_xch [ s - > amode ] ;
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}
} else {
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channels = num_core_channels + ! ! s - > lfe ;
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s - > xch_present = 0 ; /* disable further xch processing */
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if ( s - > lfe ) {
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avctx - > channel_layout | = AV_CH_LOW_FREQUENCY ;
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s - > channel_order_tab = ff_dca_channel_reorder_lfe [ s - > amode ] ;
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} else
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s - > channel_order_tab = ff_dca_channel_reorder_nolfe [ s - > amode ] ;
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}
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if ( channels > ! ! s - > lfe & &
s - > channel_order_tab [ channels - 1 - ! ! s - > lfe ] < 0 )
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return AVERROR_INVALIDDATA ;
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if ( num_core_channels + ! ! s - > lfe > 2 & &
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avctx - > request_channel_layout = = AV_CH_LAYOUT_STEREO ) {
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channels = 2 ;
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s - > output = s - > audio_header . prim_channels = = 2 ? s - > amode : DCA_STEREO ;
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avctx - > channel_layout = AV_CH_LAYOUT_STEREO ;
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/* Stereo downmix coefficients
*
* The decoder can only downmix to 2 - channel , so we need to ensure
* embedded downmix coefficients are actually targeting 2 - channel .
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*/
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if ( s - > core_downmix & & ( s - > core_downmix_amode = = DCA_STEREO | |
s - > core_downmix_amode = = DCA_STEREO_TOTAL ) ) {
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for ( i = 0 ; i < num_core_channels + ! ! s - > lfe ; i + + ) {
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/* Range checked earlier */
s - > downmix_coef [ i ] [ 0 ] = dca_dmix_code ( s - > core_downmix_codes [ i ] [ 0 ] ) ;
s - > downmix_coef [ i ] [ 1 ] = dca_dmix_code ( s - > core_downmix_codes [ i ] [ 1 ] ) ;
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}
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s - > output = s - > core_downmix_amode ;
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} else {
int am = s - > amode & DCA_CHANNEL_MASK ;
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if ( am > = FF_ARRAY_ELEMS ( ff_dca_default_coeffs ) ) {
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av_log ( s - > avctx , AV_LOG_ERROR ,
" Invalid channel mode %d \n " , am ) ;
return AVERROR_INVALIDDATA ;
}
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if ( num_core_channels + ! ! s - > lfe >
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FF_ARRAY_ELEMS ( ff_dca_default_coeffs [ 0 ] ) ) {
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avpriv_request_sample ( s - > avctx , " Downmixing %d channels " ,
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s - > audio_header . prim_channels + ! ! s - > lfe ) ;
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return AVERROR_PATCHWELCOME ;
}
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for ( i = 0 ; i < num_core_channels + ! ! s - > lfe ; i + + ) {
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s - > downmix_coef [ i ] [ 0 ] = ff_dca_default_coeffs [ am ] [ i ] [ 0 ] ;
s - > downmix_coef [ i ] [ 1 ] = ff_dca_default_coeffs [ am ] [ i ] [ 1 ] ;
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}
}
2015-03-16 09:57:36 +01:00
ff_dlog ( s - > avctx , " Stereo downmix coeffs: \n " ) ;
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for ( i = 0 ; i < num_core_channels + ! ! s - > lfe ; i + + ) {
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ff_dlog ( s - > avctx , " L, input channel %d = %f \n " , i ,
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s - > downmix_coef [ i ] [ 0 ] ) ;
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ff_dlog ( s - > avctx , " R, input channel %d = %f \n " , i ,
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s - > downmix_coef [ i ] [ 1 ] ) ;
}
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ff_dlog ( s - > avctx , " \n " ) ;
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}
} else {
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av_log ( avctx , AV_LOG_ERROR , " Nonstandard configuration %d ! \n " , s - > amode ) ;
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return AVERROR_INVALIDDATA ;
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}
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return 0 ;
}
/**
* Main frame decoding function
* FIXME add arguments
*/
static int dca_decode_frame ( AVCodecContext * avctx , void * data ,
int * got_frame_ptr , AVPacket * avpkt )
{
AVFrame * frame = data ;
const uint8_t * buf = avpkt - > data ;
int buf_size = avpkt - > size ;
int lfe_samples ;
int num_core_channels = 0 ;
int i , ret ;
float * * samples_flt ;
DCAContext * s = avctx - > priv_data ;
int channels , full_channels ;
int upsample = 0 ;
s - > exss_ext_mask = 0 ;
s - > xch_present = 0 ;
s - > dca_buffer_size = ff_dca_convert_bitstream ( buf , buf_size , s - > dca_buffer ,
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE ) ;
if ( s - > dca_buffer_size = = AVERROR_INVALIDDATA ) {
av_log ( avctx , AV_LOG_ERROR , " Not a valid DCA frame \n " ) ;
return AVERROR_INVALIDDATA ;
}
if ( ( ret = dca_parse_frame_header ( s ) ) < 0 ) {
// seems like the frame is corrupt, try with the next one
return ret ;
}
// set AVCodec values with parsed data
avctx - > sample_rate = s - > sample_rate ;
avctx - > bit_rate = s - > bit_rate ;
s - > profile = FF_PROFILE_DTS ;
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for ( i = 0 ; i < ( s - > sample_blocks / SAMPLES_PER_SUBBAND ) ; i + + ) {
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if ( ( ret = dca_decode_block ( s , 0 , i ) ) ) {
av_log ( avctx , AV_LOG_ERROR , " error decoding block \n " ) ;
return ret ;
}
}
/* record number of core channels incase less than max channels are requested */
2015-10-02 17:53:26 +02:00
num_core_channels = s - > audio_header . prim_channels ;
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if ( s - > ext_coding )
s - > core_ext_mask = dca_ext_audio_descr_mask [ s - > ext_descr ] ;
else
s - > core_ext_mask = 0 ;
ret = scan_for_extensions ( avctx ) ;
avctx - > profile = s - > profile ;
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full_channels = channels = s - > audio_header . prim_channels + ! ! s - > lfe ;
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ret = set_channel_layout ( avctx , channels , num_core_channels ) ;
if ( ret < 0 )
return ret ;
2012-10-21 21:12:36 +02:00
avctx - > channels = channels ;
2008-04-10 08:15:17 +02:00
2011-09-06 18:17:45 +02:00
/* get output buffer */
2015-09-18 13:34:20 +02:00
frame - > nb_samples = 256 * ( s - > sample_blocks / SAMPLES_PER_SUBBAND ) ;
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if ( s - > exss_ext_mask & DCA_EXT_EXSS_XLL ) {
int xll_nb_samples = s - > xll_segments * s - > xll_smpl_in_seg ;
/* Check for invalid/unsupported conditions first */
if ( s - > xll_residual_channels > channels ) {
av_log ( s - > avctx , AV_LOG_WARNING ,
" DCA: too many residual channels (%d, core channels %d). Disabling XLL \n " ,
s - > xll_residual_channels , channels ) ;
s - > exss_ext_mask & = ~ DCA_EXT_EXSS_XLL ;
} else if ( xll_nb_samples ! = frame - > nb_samples & &
2 * frame - > nb_samples ! = xll_nb_samples ) {
av_log ( s - > avctx , AV_LOG_WARNING ,
" DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL \n " ,
xll_nb_samples , frame - > nb_samples ) ;
s - > exss_ext_mask & = ~ DCA_EXT_EXSS_XLL ;
} else {
if ( 2 * frame - > nb_samples = = xll_nb_samples ) {
av_log ( s - > avctx , AV_LOG_INFO ,
" XLL: upsampling core channels by a factor of 2 \n " ) ;
upsample = 1 ;
frame - > nb_samples = xll_nb_samples ;
// FIXME: Is it good enough to copy from the first channel set?
avctx - > sample_rate = s - > xll_chsets [ 0 ] . sampling_frequency ;
}
/* If downmixing to stereo, don't decode additional channels.
* FIXME : Using the xch_disable flag for this doesn ' t seem right . */
if ( ! s - > xch_disable )
avctx - > channels + = s - > xll_channels - s - > xll_residual_channels ;
}
}
/* FIXME: This is an ugly hack, to just revert to the default
* layout if we have additional channels . Need to convert the XLL
* channel masks to libav channel_layout mask . */
if ( av_get_channel_layout_nb_channels ( avctx - > channel_layout ) ! = avctx - > channels )
avctx - > channel_layout = 0 ;
2012-11-21 21:34:46 +01:00
if ( ( ret = ff_get_buffer ( avctx , frame , 0 ) ) < 0 ) {
2011-09-06 18:17:45 +02:00
av_log ( avctx , AV_LOG_ERROR , " get_buffer() failed \n " ) ;
return ret ;
}
2014-09-08 19:40:26 +02:00
samples_flt = ( float * * ) frame - > extended_data ;
2010-06-22 10:33:00 +02:00
2012-10-11 20:21:56 +02:00
/* allocate buffer for extra channels if downmixing */
if ( avctx - > channels < full_channels ) {
ret = av_samples_get_buffer_size ( NULL , full_channels - channels ,
2012-12-23 23:43:47 +01:00
frame - > nb_samples ,
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avctx - > sample_fmt , 0 ) ;
if ( ret < 0 )
return ret ;
av_fast_malloc ( & s - > extra_channels_buffer ,
& s - > extra_channels_buffer_size , ret ) ;
if ( ! s - > extra_channels_buffer )
return AVERROR ( ENOMEM ) ;
2014-09-08 19:40:26 +02:00
ret = av_samples_fill_arrays ( ( uint8_t * * ) s - > extra_channels , NULL ,
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s - > extra_channels_buffer ,
full_channels - channels ,
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frame - > nb_samples , avctx - > sample_fmt , 0 ) ;
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if ( ret < 0 )
return ret ;
}
2010-06-22 10:33:00 +02:00
/* filter to get final output */
2015-09-18 13:34:20 +02:00
for ( i = 0 ; i < ( s - > sample_blocks / SAMPLES_PER_SUBBAND ) ; i + + ) {
2012-08-27 17:43:34 +02:00
int ch ;
2014-02-11 11:45:27 +01:00
unsigned block = upsample ? 512 : 256 ;
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for ( ch = 0 ; ch < channels ; ch + + )
2014-02-11 11:45:27 +01:00
s - > samples_chanptr [ ch ] = samples_flt [ ch ] + i * block ;
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for ( ; ch < full_channels ; ch + + )
2014-02-11 11:45:27 +01:00
s - > samples_chanptr [ ch ] = s - > extra_channels [ ch - channels ] + i * block ;
2012-08-27 17:43:34 +02:00
2014-02-11 11:45:27 +01:00
dca_filter_channels ( s , i , upsample ) ;
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/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
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if ( ( s - > source_pcm_res & 1 ) & & s - > xch_present ) {
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float * back_chan = s - > samples_chanptr [ s - > channel_order_tab [ s - > xch_base_channel ] ] ;
float * lt_chan = s - > samples_chanptr [ s - > channel_order_tab [ s - > xch_base_channel - 2 ] ] ;
float * rt_chan = s - > samples_chanptr [ s - > channel_order_tab [ s - > xch_base_channel - 1 ] ] ;
2012-06-08 19:49:56 +02:00
s - > fdsp . vector_fmac_scalar ( lt_chan , back_chan , - M_SQRT1_2 , 256 ) ;
s - > fdsp . vector_fmac_scalar ( rt_chan , back_chan , - M_SQRT1_2 , 256 ) ;
2010-07-12 11:10:53 +02:00
}
2007-02-27 07:30:40 +01:00
}
2010-06-22 10:33:00 +02:00
/* update lfe history */
2015-09-18 13:34:20 +02:00
lfe_samples = 2 * s - > lfe * ( s - > sample_blocks / SAMPLES_PER_SUBBAND ) ;
2012-01-03 11:49:28 +01:00
for ( i = 0 ; i < 2 * s - > lfe * 4 ; i + + )
2010-06-22 10:33:00 +02:00
s - > lfe_data [ i ] = s - > lfe_data [ i + lfe_samples ] ;
2014-02-11 11:45:27 +01:00
if ( s - > exss_ext_mask & DCA_EXT_EXSS_XLL ) {
ret = ff_dca_xll_decode_audio ( s , frame ) ;
if ( ret < 0 )
return ret ;
}
2013-12-11 03:03:31 +01:00
/* AVMatrixEncoding
*
* DCA_STEREO_TOTAL ( Lt / Rt ) is equivalent to Dolby Surround */
ret = ff_side_data_update_matrix_encoding ( frame ,
( s - > output & ~ DCA_LFE ) = = DCA_STEREO_TOTAL ?
AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE ) ;
if ( ret < 0 )
return ret ;
2012-12-23 23:43:47 +01:00
* got_frame_ptr = 1 ;
2011-09-06 18:17:45 +02:00
2007-02-27 07:30:40 +01:00
return buf_size ;
}
/**
* DCA initialization
*
* @ param avctx pointer to the AVCodecContext
*/
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static av_cold int dca_decode_init ( AVCodecContext * avctx )
2007-02-27 07:30:40 +01:00
{
DCAContext * s = avctx - > priv_data ;
s - > avctx = avctx ;
dca_init_vlcs ( ) ;
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avpriv_float_dsp_init ( & s - > fdsp , avctx - > flags & AV_CODEC_FLAG_BITEXACT ) ;
2009-05-16 16:17:08 +02:00
ff_mdct_init ( & s - > imdct , 6 , 1 , 1.0 ) ;
2010-04-10 18:27:53 +02:00
ff_synth_filter_init ( & s - > synth ) ;
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ff_dcadsp_init ( & s - > dcadsp ) ;
2011-01-30 16:06:46 +01:00
ff_fmt_convert_init ( & s - > fmt_conv , avctx ) ;
2008-01-05 22:23:57 +01:00
2012-08-27 17:43:34 +02:00
avctx - > sample_fmt = AV_SAMPLE_FMT_FLTP ;
2008-10-26 10:54:53 +01:00
2011-01-27 15:21:26 +01:00
/* allow downmixing to stereo */
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if ( avctx - > channels > 2 & &
avctx - > request_channel_layout = = AV_CH_LAYOUT_STEREO )
avctx - > channels = 2 ;
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return 0 ;
}
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static av_cold int dca_decode_end ( AVCodecContext * avctx )
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{
DCAContext * s = avctx - > priv_data ;
ff_mdct_end ( & s - > imdct ) ;
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av_freep ( & s - > extra_channels_buffer ) ;
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av_freep ( & s - > xll_sample_buf ) ;
av_freep ( & s - > qmf64_table ) ;
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return 0 ;
}
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static const AVOption options [ ] = {
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{ " disable_xch " , " disable decoding of the XCh extension " , offsetof ( DCAContext , xch_disable ) , AV_OPT_TYPE_INT , { . i64 = 0 } , 0 , 1 , AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM } ,
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{ " disable_xll " , " disable decoding of the XLL extension " , offsetof ( DCAContext , xll_disable ) , AV_OPT_TYPE_INT , { . i64 = 1 } , 0 , 1 , AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM } ,
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{ NULL } ,
} ;
static const AVClass dca_decoder_class = {
. class_name = " DCA decoder " ,
. item_name = av_default_item_name ,
. option = options ,
. version = LIBAVUTIL_VERSION_INT ,
} ;
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AVCodec ff_dca_decoder = {
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. name = " dca " ,
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. long_name = NULL_IF_CONFIG_SMALL ( " DCA (DTS Coherent Acoustics) " ) ,
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. type = AVMEDIA_TYPE_AUDIO ,
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. id = AV_CODEC_ID_DTS ,
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. priv_data_size = sizeof ( DCAContext ) ,
. init = dca_decode_init ,
. decode = dca_decode_frame ,
. close = dca_decode_end ,
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. capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1 ,
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. sample_fmts = ( const enum AVSampleFormat [ ] ) { AV_SAMPLE_FMT_FLTP ,
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AV_SAMPLE_FMT_NONE } ,
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. profiles = NULL_IF_CONFIG_SMALL ( ff_dca_profiles ) ,
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. priv_class = & dca_decoder_class ,
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} ;