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dca.c
1839 lines (1571 loc) · 67.1 KB
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dca.c
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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
*
* 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 <math.h>
#include <stddef.h>
#include <stdio.h>
#include "libavutil/common.h"
#include "libavutil/intmath.h"
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "get_bits.h"
#include "put_bits.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dca.h"
#include "synth_filter.h"
#include "dcadsp.h"
//#define TRACE
#define DCA_PRIM_CHANNELS_MAX (7)
#define DCA_SUBBANDS (32)
#define DCA_ABITS_MAX (32) /* Should be 28 */
#define DCA_SUBSUBFRAMES_MAX (4)
#define DCA_SUBFRAMES_MAX (16)
#define DCA_BLOCKS_MAX (16)
#define DCA_LFE_MAX (3)
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
};
/* these are unconfirmed but should be mostly correct */
enum DCAExSSSpeakerMask {
DCA_EXSS_FRONT_CENTER = 0x0001,
DCA_EXSS_FRONT_LEFT_RIGHT = 0x0002,
DCA_EXSS_SIDE_REAR_LEFT_RIGHT = 0x0004,
DCA_EXSS_LFE = 0x0008,
DCA_EXSS_REAR_CENTER = 0x0010,
DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,
DCA_EXSS_REAR_LEFT_RIGHT = 0x0040,
DCA_EXSS_FRONT_HIGH_CENTER = 0x0080,
DCA_EXSS_OVERHEAD = 0x0100,
DCA_EXSS_CENTER_LEFT_RIGHT = 0x0200,
DCA_EXSS_WIDE_LEFT_RIGHT = 0x0400,
DCA_EXSS_SIDE_LEFT_RIGHT = 0x0800,
DCA_EXSS_LFE2 = 0x1000,
DCA_EXSS_SIDE_HIGH_LEFT_RIGHT = 0x2000,
DCA_EXSS_REAR_HIGH_CENTER = 0x4000,
DCA_EXSS_REAR_HIGH_LEFT_RIGHT = 0x8000,
};
enum DCAExtensionMask {
DCA_EXT_CORE = 0x001, ///< core in core substream
DCA_EXT_XXCH = 0x002, ///< XXCh channels extension in core substream
DCA_EXT_X96 = 0x004, ///< 96/24 extension in core substream
DCA_EXT_XCH = 0x008, ///< XCh channel extension in core substream
DCA_EXT_EXSS_CORE = 0x010, ///< core in ExSS (extension substream)
DCA_EXT_EXSS_XBR = 0x020, ///< extended bitrate extension in ExSS
DCA_EXT_EXSS_XXCH = 0x040, ///< XXCh channels extension in ExSS
DCA_EXT_EXSS_X96 = 0x080, ///< 96/24 extension in ExSS
DCA_EXT_EXSS_LBR = 0x100, ///< low bitrate component in ExSS
DCA_EXT_EXSS_XLL = 0x200, ///< lossless extension in ExSS
};
/* 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.
* S -> side, when both rear and back are configured move one of them to the side channel
* OV -> center back
* All 2 channel configurations -> CH_LAYOUT_STEREO
*/
static const int64_t dca_core_channel_layout[] = {
CH_FRONT_CENTER, ///< 1, A
CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
CH_LAYOUT_STEREO, ///< 2, (L+R) + (L-R) (sum-difference)
CH_LAYOUT_STEREO, ///< 2, LT +RT (left and right total)
CH_LAYOUT_STEREO|CH_FRONT_CENTER, ///< 3, C+L+R
CH_LAYOUT_STEREO|CH_BACK_CENTER, ///< 3, L+R+S
CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 4, C + L + R+ S
CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 4, L + R +SL+ SR
CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 5, C + L + R+ SL+SR
CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 6, C + L + R+ LR + RR + OV
CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_FRONT_LEFT_OF_CENTER|CH_BACK_CENTER|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 6, CF+ CR+LF+ RF+LR + RR
CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2+ SR1 + SR2
CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_BACK_CENTER|CH_SIDE_RIGHT, ///< 8, CL + C+ CR + L + R + SL + S+ SR
};
static const int8_t dca_lfe_index[] = {
1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
};
static const int8_t dca_channel_reorder_lfe[][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 2, 0, 1, -1, -1, -1, -1, -1, -1},
{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
{ 2, 0, 1, 4, -1, -1, -1, -1, -1},
{ 0, 1, 3, 4, -1, -1, -1, -1, -1},
{ 2, 0, 1, 4, 5, -1, -1, -1, -1},
{ 3, 4, 0, 1, 5, 6, -1, -1, -1},
{ 2, 0, 1, 4, 5, 6, -1, -1, -1},
{ 0, 6, 4, 5, 2, 3, -1, -1, -1},
{ 4, 2, 5, 0, 1, 6, 7, -1, -1},
{ 5, 6, 0, 1, 7, 3, 8, 4, -1},
{ 4, 2, 5, 0, 1, 6, 8, 7, -1},
};
static const int8_t dca_channel_reorder_lfe_xch[][9] = {
{ 0, 2, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
{ 2, 0, 1, 4, -1, -1, -1, -1, -1},
{ 0, 1, 3, 4, -1, -1, -1, -1, -1},
{ 2, 0, 1, 4, 5, -1, -1, -1, -1},
{ 0, 1, 4, 5, 3, -1, -1, -1, -1},
{ 2, 0, 1, 5, 6, 4, -1, -1, -1},
{ 3, 4, 0, 1, 6, 7, 5, -1, -1},
{ 2, 0, 1, 4, 5, 6, 7, -1, -1},
{ 0, 6, 4, 5, 2, 3, 7, -1, -1},
{ 4, 2, 5, 0, 1, 7, 8, 6, -1},
{ 5, 6, 0, 1, 8, 3, 9, 4, 7},
{ 4, 2, 5, 0, 1, 6, 9, 8, 7},
};
static const int8_t dca_channel_reorder_nolfe[][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 2, 0, 1, -1, -1, -1, -1, -1, -1},
{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
{ 2, 0, 1, 3, -1, -1, -1, -1, -1},
{ 0, 1, 2, 3, -1, -1, -1, -1, -1},
{ 2, 0, 1, 3, 4, -1, -1, -1, -1},
{ 2, 3, 0, 1, 4, 5, -1, -1, -1},
{ 2, 0, 1, 3, 4, 5, -1, -1, -1},
{ 0, 5, 3, 4, 1, 2, -1, -1, -1},
{ 3, 2, 4, 0, 1, 5, 6, -1, -1},
{ 4, 5, 0, 1, 6, 2, 7, 3, -1},
{ 3, 2, 4, 0, 1, 5, 7, 6, -1},
};
static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
{ 2, 0, 1, 3, -1, -1, -1, -1, -1},
{ 0, 1, 2, 3, -1, -1, -1, -1, -1},
{ 2, 0, 1, 3, 4, -1, -1, -1, -1},
{ 0, 1, 3, 4, 2, -1, -1, -1, -1},
{ 2, 0, 1, 4, 5, 3, -1, -1, -1},
{ 2, 3, 0, 1, 5, 6, 4, -1, -1},
{ 2, 0, 1, 3, 4, 5, 6, -1, -1},
{ 0, 5, 3, 4, 1, 2, 6, -1, -1},
{ 3, 2, 4, 0, 1, 6, 7, 5, -1},
{ 4, 5, 0, 1, 7, 2, 8, 3, 6},
{ 3, 2, 4, 0, 1, 5, 8, 7, 6},
};
#define DCA_DOLBY 101 /* FIXME */
#define DCA_CHANNEL_BITS 6
#define DCA_CHANNEL_MASK 0x3F
#define DCA_LFE 0x80
#define HEADER_SIZE 14
#define DCA_MAX_FRAME_SIZE 16384
#define DCA_MAX_EXSS_HEADER_SIZE 4096
/** Bit allocation */
typedef struct {
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
static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
{
return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
}
typedef struct {
AVCodecContext *avctx;
/* Frame header */
int frame_type; ///< type of the current frame
int samples_deficit; ///< deficit sample count
int crc_present; ///< crc is present in the bitstream
int sample_blocks; ///< number of PCM sample blocks
int frame_size; ///< primary frame byte size
int amode; ///< audio channels arrangement
int sample_rate; ///< audio sampling rate
int bit_rate; ///< transmission bit rate
int bit_rate_index; ///< transmission bit rate index
int downmix; ///< embedded downmix enabled
int dynrange; ///< embedded dynamic range flag
int timestamp; ///< embedded time stamp flag
int aux_data; ///< auxiliary data flag
int hdcd; ///< source material is mastered in HDCD
int ext_descr; ///< extension audio descriptor flag
int ext_coding; ///< extended coding flag
int aspf; ///< audio sync word insertion flag
int lfe; ///< low frequency effects flag
int predictor_history; ///< predictor history flag
int header_crc; ///< header crc check bytes
int multirate_inter; ///< multirate interpolator switch
int version; ///< encoder software revision
int copy_history; ///< copy history
int source_pcm_res; ///< source pcm resolution
int front_sum; ///< front sum/difference flag
int surround_sum; ///< surround sum/difference flag
int dialog_norm; ///< dialog normalisation parameter
/* Primary audio coding header */
int subframes; ///< number of subframes
int total_channels; ///< number of channels including extensions
int prim_channels; ///< number of primary audio channels
int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
/* Primary audio coding side information */
int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
int dynrange_coef; ///< dynamic range coefficient
int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
int hist_index[DCA_PRIM_CHANNELS_MAX];
DECLARE_ALIGNED(16, float, raXin)[32];
int output; ///< type of output
float add_bias; ///< output bias
float scale_bias; ///< output scale
DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
DECLARE_ALIGNED(16, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];
const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX+1];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
const int8_t* channel_order_tab; ///< channel reordering table, lfe and non lfe
GetBitContext gb;
/* Current position in DCA frame */
int current_subframe;
int current_subsubframe;
/* XCh extension information */
int xch_present;
int xch_base_channel; ///< index of first (only) channel containing XCH data
/* Other detected extensions in the core substream */
int xxch_present;
int x96_present;
/* ExSS header parser */
int static_fields; ///< static fields present
int mix_metadata; ///< mixing metadata present
int num_mix_configs; ///< number of mix out configurations
int mix_config_num_ch[4]; ///< number of channels in each mix out configuration
int profile;
int debug_flag; ///< used for suppressing repeated error messages output
DSPContext dsp;
FFTContext imdct;
SynthFilterContext synth;
DCADSPContext dcadsp;
} DCAContext;
static const uint16_t dca_vlc_offs[] = {
0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364,
5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508,
5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564,
7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240,
12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
};
static av_cold void dca_init_vlcs(void)
{
static int vlcs_initialized = 0;
int i, j, c = 14;
static VLC_TYPE dca_table[23622][2];
if (vlcs_initialized)
return;
dca_bitalloc_index.offset = 1;
dca_bitalloc_index.wrap = 2;
for (i = 0; i < 5; i++) {
dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
bitalloc_12_bits[i], 1, 1,
bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
}
dca_scalefactor.offset = -64;
dca_scalefactor.wrap = 2;
for (i = 0; i < 5; i++) {
dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
scales_bits[i], 1, 1,
scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
}
dca_tmode.offset = 0;
dca_tmode.wrap = 1;
for (i = 0; i < 4; i++) {
dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
tmode_bits[i], 1, 1,
tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
}
for (i = 0; i < 10; i++)
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);
dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
bitalloc_sizes[i],
bitalloc_bits[i][j], 1, 1,
bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
c++;
}
vlcs_initialized = 1;
}
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
{
while(len--)
*dst++ = get_bits(gb, bits);
}
static int dca_parse_audio_coding_header(DCAContext * s, int base_channel)
{
int i, j;
static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
s->prim_channels = s->total_channels;
if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
s->prim_channels = DCA_PRIM_CHANNELS_MAX;
for (i = base_channel; i < s->prim_channels; i++) {
s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
if (s->subband_activity[i] > DCA_SUBBANDS)
s->subband_activity[i] = DCA_SUBBANDS;
}
for (i = base_channel; i < s->prim_channels; i++) {
s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
if (s->vq_start_subband[i] > DCA_SUBBANDS)
s->vq_start_subband[i] = DCA_SUBBANDS;
}
get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3);
get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2);
get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3);
/* Get codebooks quantization indexes */
if (!base_channel)
memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
for (j = 1; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++)
s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
/* Get scale factor adjustment */
for (j = 0; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++)
s->scalefactor_adj[i][j] = 1;
for (j = 1; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++)
if (s->quant_index_huffman[i][j] < thr[j])
s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
if (s->crc_present) {
/* Audio header CRC check */
get_bits(&s->gb, 16);
}
s->current_subframe = 0;
s->current_subsubframe = 0;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
for (i = base_channel; i < s->prim_channels; i++){
av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
for (j = 0; j < 11; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %i",
s->quant_index_huffman[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
for (j = 0; j < 11; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
#endif
return 0;
}
static int dca_parse_frame_header(DCAContext * s)
{
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
get_bits(&s->gb, 32);
/* 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)
return -1;
s->amode = get_bits(&s->gb, 6);
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
return -1;
s->bit_rate_index = get_bits(&s->gb, 5);
s->bit_rate = dca_bit_rates[s->bit_rate_index];
if (!s->bit_rate)
return -1;
s->downmix = get_bits(&s->gb, 1);
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);
/* 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 */
s->output = s->amode;
if (s->lfe) s->output |= DCA_LFE;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
s->sample_blocks, s->sample_blocks * 32);
av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
s->amode, dca_channels[s->amode]);
av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n",
s->sample_rate);
av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",
s->bit_rate);
av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
s->predictor_history);
av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
s->multirate_inter);
av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
av_log(s->avctx, AV_LOG_DEBUG,
"source pcm resolution: %i (%i bits/sample)\n",
s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
#endif
/* Primary audio coding header */
s->subframes = get_bits(&s->gb, 4) + 1;
return dca_parse_audio_coding_header(s, 0);
}
static inline int get_scale(GetBitContext *gb, int level, int value)
{
if (level < 5) {
/* huffman encoded */
value += get_bitalloc(gb, &dca_scalefactor, level);
} else if (level < 8)
value = get_bits(gb, level + 1);
return value;
}
static int dca_subframe_header(DCAContext * s, int base_channel, int block_index)
{
/* Primary audio coding side information */
int j, k;
if (!base_channel) {
s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
}
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
s->prediction_mode[j][k] = get_bits(&s->gb, 1);
}
/* Get prediction codebook */
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) {
if (s->prediction_mode[j][k] > 0) {
/* (Prediction coefficient VQ address) */
s->prediction_vq[j][k] = get_bits(&s->gb, 12);
}
}
}
/* Bit allocation index */
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->vq_start_subband[j]; k++) {
if (s->bitalloc_huffman[j] == 6)
s->bitalloc[j][k] = get_bits(&s->gb, 5);
else if (s->bitalloc_huffman[j] == 5)
s->bitalloc[j][k] = get_bits(&s->gb, 4);
else if (s->bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
return -1;
} else {
s->bitalloc[j][k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
}
if (s->bitalloc[j][k] > 26) {
// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
// j, k, s->bitalloc[j][k]);
return -1;
}
}
}
/* Transition mode */
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) {
s->transition_mode[j][k] = 0;
if (s->subsubframes[s->current_subframe] > 1 &&
k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
s->transition_mode[j][k] =
get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
}
}
}
for (j = base_channel; j < s->prim_channels; j++) {
const uint32_t *scale_table;
int scale_sum;
memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
if (s->scalefactor_huffman[j] == 6)
scale_table = scale_factor_quant7;
else
scale_table = scale_factor_quant6;
/* When huffman coded, only the difference is encoded */
scale_sum = 0;
for (k = 0; k < s->subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
s->scale_factor[j][k][0] = scale_table[scale_sum];
}
if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
/* Get second scale factor */
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
s->scale_factor[j][k][1] = scale_table[scale_sum];
}
}
}
/* Joint subband scale factor codebook select */
for (j = base_channel; j < s->prim_channels; j++) {
/* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0)
s->joint_huff[j] = get_bits(&s->gb, 3);
}
/* Scale factors for joint subband coding */
for (j = base_channel; j < s->prim_channels; j++) {
int source_channel;
/* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0) {
int scale = 0;
source_channel = s->joint_intensity[j] - 1;
/* When huffman coded, only the difference is encoded
* (is this valid as well for joint scales ???) */
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
scale = get_scale(&s->gb, s->joint_huff[j], 0);
scale += 64; /* bias */
s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
}
if (!(s->debug_flag & 0x02)) {
av_log(s->avctx, AV_LOG_DEBUG,
"Joint stereo coding not supported\n");
s->debug_flag |= 0x02;
}
}
}
/* Stereo downmix coefficients */
if (!base_channel && s->prim_channels > 2) {
if (s->downmix) {
for (j = base_channel; j < s->prim_channels; j++) {
s->downmix_coef[j][0] = get_bits(&s->gb, 7);
s->downmix_coef[j][1] = get_bits(&s->gb, 7);
}
} else {
int am = s->amode & DCA_CHANNEL_MASK;
for (j = base_channel; j < s->prim_channels; j++) {
s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
}
}
}
/* Dynamic range coefficient */
if (s->dynrange)
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 */
for (j = base_channel; j < s->prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
/* 1 vector -> 32 samples */
s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
/* Low frequency effect data */
if (!base_channel && s->lfe) {
/* LFE samples */
int lfe_samples = 2 * s->lfe * (4 + block_index);
int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
float lfe_scale;
for (j = lfe_samples; j < lfe_end_sample; j++) {
/* Signed 8 bits int */
s->lfe_data[j] = get_sbits(&s->gb, 8);
}
/* Scale factor index */
s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];
/* Quantization step size * scale factor */
lfe_scale = 0.035 * s->lfe_scale_factor;
for (j = lfe_samples; j < lfe_end_sample; j++)
s->lfe_data[j] *= lfe_scale;
}
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes[s->current_subframe]);
av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
s->partial_samples[s->current_subframe]);
for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG,
"prediction coefs: %f, %f, %f, %f\n",
(float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
(float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
(float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
(float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
}
for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
for (k = 0; k < s->vq_start_subband[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
for (k = 0; k < s->subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
}
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = base_channel; j < s->prim_channels; j++) {
if (s->joint_intensity[j] > 0) {
int source_channel = s->joint_intensity[j] - 1;
av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
}
if (!base_channel && s->prim_channels > 2 && s->downmix) {
av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
for (j = 0; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
}
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = base_channel; j < s->prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
if (!base_channel && s->lfe) {
int lfe_samples = 2 * s->lfe * (4 + block_index);
int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
for (j = lfe_samples; j < lfe_end_sample; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
#endif
return 0;
}
static void qmf_32_subbands(DCAContext * s, int chans,
float samples_in[32][8], float *samples_out,
float scale, float bias)
{
const float *prCoeff;
int i;
int sb_act = s->subband_activity[chans];
int subindex;
scale *= sqrt(1/8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
prCoeff = fir_32bands_nonperfect;
else /* Perfect reconstruction */
prCoeff = fir_32bands_perfect;
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
/* Load in one sample from each subband and clear inactive subbands */
for (i = 0; i < sb_act; i++){
uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ ((i-1)&2)<<30;
AV_WN32A(&s->raXin[i], v);
}
for (; i < 32; i++)
s->raXin[i] = 0.0;
s->synth.synth_filter_float(&s->imdct,
s->subband_fir_hist[chans], &s->hist_index[chans],
s->subband_fir_noidea[chans], prCoeff,
samples_out, s->raXin, scale, bias);
samples_out+= 32;
}
}
static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
int num_deci_sample, float *samples_in,
float *samples_out, float scale,
float bias)
{
/* 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
*/
int decifactor;
const float *prCoeff;
int deciindex;
/* Select decimation filter */
if (decimation_select == 1) {
decifactor = 64;
prCoeff = lfe_fir_128;
} else {
decifactor = 32;
prCoeff = lfe_fir_64;
}
/* Interpolation */
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,
scale, bias);
samples_in++;
samples_out += 2 * decifactor;
}
}
/* downmixing routines */
#define MIX_REAR1(samples, si1, rs, coef) \
samples[i] += samples[si1] * coef[rs][0]; \
samples[i+256] += samples[si1] * coef[rs][1];
#define MIX_REAR2(samples, si1, si2, rs, coef) \
samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
#define MIX_FRONT3(samples, coef) \
t = samples[i]; \
samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1];
#define DOWNMIX_TO_STEREO(op1, op2) \
for (i = 0; i < 256; i++){ \
op1 \
op2 \
}
static void dca_downmix(float *samples, int srcfmt,
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2])
{
int i;
float t;
float coef[DCA_PRIM_CHANNELS_MAX][2];
for (i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
}
switch (srcfmt) {
case DCA_MONO:
case DCA_CHANNEL:
case DCA_STEREO_TOTAL:
case DCA_STEREO_SUMDIFF:
case DCA_4F2R:
av_log(NULL, 0, "Not implemented!\n");
break;
case DCA_STEREO:
break;
case DCA_3F:
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
break;
case DCA_2F1R:
DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
break;
case DCA_3F1R:
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR1(samples, i + 768, 3, coef));
break;
case DCA_2F2R:
DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
break;
case DCA_3F2R:
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR2(samples, i + 768, i + 1024, 3, coef));
break;
}
}
/* Very compact version of the block code decoder that does not use table
* look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int *values)
{
int i;
int offset = (levels - 1) >> 1;
for (i = 0; i < 4; i++) {
int div = FASTDIV(code, levels);
values[i] = code - offset - div*levels;
code = div;