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FFmpeg/libavcodec/dca.c
Michael Niedermayer e4de71677f Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-03 03:00:30 +01:00

1957 lines
71 KiB
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 "libavutil/audioconvert.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"
#include "fmtconvert.h"
#if ARCH_ARM
# include "arm/dca.h"
#endif
//#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
};
/* -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,
};
/* extensions that reside in core substream */
#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)
/* 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 -> AV_CH_LAYOUT_STEREO
*/
static const uint64_t dca_core_channel_layout[] = {
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
};
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
#define DCA_BUFFER_PADDING_SIZE 1024
/** 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;
AVFrame frame;
/* 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) */
DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
int hist_index[DCA_PRIM_CHANNELS_MAX];
DECLARE_ALIGNED(32, float, raXin)[32];
int output; ///< type of output
float scale_bias; ///< output scale
DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
DECLARE_ALIGNED(32, 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 + DCA_BUFFER_PADDING_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;
int core_ext_mask; ///< present extensions in the core substream
/* XCh extension information */
int xch_present; ///< XCh extension present and valid
int xch_base_channel; ///< index of first (only) channel containing XCH data
/* 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;
FmtConvertContext fmt_conv;
} 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 */
skip_bits_long(&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 AVERROR_INVALIDDATA;
s->amode = get_bits(&s->gb, 6);
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
return AVERROR_INVALIDDATA;
s->bit_rate_index = get_bits(&s->gb, 5);
s->bit_rate = dca_bit_rates[s->bit_rate_index];
if (!s->bit_rate)
return AVERROR_INVALIDDATA;
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 (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
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 AVERROR_INVALIDDATA;
} 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 AVERROR_INVALIDDATA;
}
}
}
/* 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]);
}
}
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
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);
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
/* 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 (!base_channel && 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)
{
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;
for (i = sb_act; i < 32; i++)
s->raXin[i] = 0.0;
/* 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++){
unsigned sign = (i - 1) & 2;
uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
AV_WN32A(&s->raXin[i], v);
}
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);
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)
{
/* 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);
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+c]; \
u = samples[i+l]; \
v = samples[i+r]; \
samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * 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],
const int8_t *channel_mapping)
{
int c,l,r,sl,sr,s;
int i;
float t, u, v;
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:
c = channel_mapping[0] * 256;
l = channel_mapping[1] * 256;
r = channel_mapping[2] * 256;
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
break;
case DCA_2F1R:
s = channel_mapping[2] * 256;
DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef),);
break;
case DCA_3F1R:
c = channel_mapping[0] * 256;
l = channel_mapping[1] * 256;
r = channel_mapping[2] * 256;
s = channel_mapping[3] * 256;
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR1(samples, i + s, 3, coef));
break;
case DCA_2F2R:
sl = channel_mapping[2] * 256;
sr = channel_mapping[3] * 256;
DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef),);
break;
case DCA_3F2R:
c = channel_mapping[0] * 256;
l = channel_mapping[1] * 256;
r = channel_mapping[2] * 256;
sl = channel_mapping[3] * 256;
sr = channel_mapping[4] * 256;
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR2(samples, i + sl, i + sr, 3, coef));
break;
}
}
#ifndef decode_blockcodes
/* 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;
}
return code;
}
static int decode_blockcodes(int code1, int code2, int levels, int *values)
{
return decode_blockcode(code1, levels, values) |
decode_blockcode(code2, levels, values + 4);
}
#endif
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 };
#ifndef int8x8_fmul_int32
static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
{
float fscale = scale / 16.0;
int i;
for (i = 0; i < 8; i++)
dst[i] = src[i] * fscale;
}
#endif
static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
{
int k, l;
int subsubframe = s->current_subsubframe;
const float *quant_step_table;
/* FIXME */
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
LOCAL_ALIGNED_16(int, block, [8]);
/*
* Audio data
*/
/* Select quantization step size table */
if (s->bit_rate_index == 0x1f)
quant_step_table = lossless_quant_d;
else
quant_step_table = lossy_quant_d;
for (k = base_channel; k < s->prim_channels; k++) {
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
/* Select the mid-tread linear quantizer */
int abits = s->bitalloc[k][l];
float quant_step_size = quant_step_table[abits];
/*
* Determine quantization index code book and its type
*/
/* Select quantization index code book */
int sel = s->quant_index_huffman[k][abits];
/*
* Extract bits from the bit stream
*/
if (!abits){
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
} else {
/* Deal with transients */
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
if (abits <= 7){
/* Block code */
int block_code1, block_code2, size, levels, err;
size = abits_sizes[abits-1];
levels = abits_levels[abits-1];
block_code1 = get_bits(&s->gb, size);
block_code2 = get_bits(&s->gb, size);
err = decode_blockcodes(block_code1, block_code2,
levels, block);
if (err) {
av_log(s->avctx, AV_LOG_ERROR,
"ERROR: block code look-up failed\n");
return AVERROR_INVALIDDATA;
}
}else{
/* no coding */
for (m = 0; m < 8; m++)
block[m] = get_sbits(&s->gb, abits - 3);
}
}else{
/* Huffman coded */
for (m = 0; m < 8; m++)
block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
}
s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
block, rscale, 8);
}
/*
* Inverse ADPCM if in prediction mode
*/
if (s->prediction_mode[k][l]) {
int n;
for (m = 0; m < 8; m++) {
for (n = 1; n <= 4; n++)
if (m >= n)
subband_samples[k][l][m] +=
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
subband_samples[k][l][m - n] / 8192);
else if (s->predictor_history)
subband_samples[k][l][m] +=
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
s->subband_samples_hist[k][l][m - n +
4] / 8192);
}
}
}
/*
* Decode VQ encoded high frequencies
*/
for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
/* 1 vector -> 32 samples but we only need the 8 samples
* for this subsubframe. */
int hfvq = s->high_freq_vq[k][l];
if (!s->debug_flag & 0x01) {
av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01;
}
int8x8_fmul_int32(subband_samples[k][l],
&high_freq_vq[hfvq][subsubframe * 8],
s->scale_factor[k][l][0]);
}
}
/* Check for DSYNC after subsubframe */
if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
#endif
} else {
av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
}
}
/* Backup predictor history for adpcm */
for (k = base_channel; k < s->prim_channels; k++)
for (l = 0; l < s->vq_start_subband[k]; l++)
memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
4 * sizeof(subband_samples[0][0][0]));
return 0;
}
static int dca_filter_channels(DCAContext * s, int block_index)
{
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
int k;
/* 32 subbands QMF */
for (k = 0; k < s->prim_channels; k++) {
/* static float pcm_to_double[8] =
{32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );
}
/* Down mixing */
if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->output & DCA_LFE) {
lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
s->lfe_data + 2 * s->lfe * (block_index + 4),
&s->samples[256 * dca_lfe_index[s->amode]],
(1.0/256.0)*s->scale_bias);
/* Outputs 20bits pcm samples */
}
return 0;
}
static int dca_subframe_footer(DCAContext * s, int base_channel)
{
int aux_data_count = 0, i;
/*
* Unpack optional information
*/
/* presumably optional information only appears in the core? */
if (!base_channel) {
if (s->timestamp)
skip_bits_long(&s->gb, 32);
if (s->aux_data)
aux_data_count = get_bits(&s->gb, 6);
for (i = 0; i < aux_data_count; i++)
get_bits(&s->gb, 8);
if (s->crc_present && (s->downmix || s->dynrange))
get_bits(&s->gb, 16);
}
return 0;
}
/**
* Decode a dca frame block
*
* @param s pointer to the DCAContext
*/
static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
{
int ret;
/* Sanity check */
if (s->current_subframe >= s->subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
s->current_subframe, s->subframes);
return AVERROR_INVALIDDATA;
}
if (!s->current_subsubframe) {
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
/* Read subframe header */
if ((ret = dca_subframe_header(s, base_channel, block_index)))
return ret;
}
/* Read subsubframe */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
if ((ret = dca_subsubframe(s, base_channel, block_index)))
return ret;
/* Update state */
s->current_subsubframe++;
if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
s->current_subsubframe = 0;
s->current_subframe++;
}
if (s->current_subframe >= s->subframes) {
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
/* Read subframe footer */
if ((ret = dca_subframe_footer(s, base_channel)))
return ret;
}
return 0;
}
/**
* Convert bitstream to one representation based on sync marker
*/
static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
int max_size)
{
uint32_t mrk;
int i, tmp;
const uint16_t *ssrc = (const uint16_t *) src;
uint16_t *sdst = (uint16_t *) dst;
PutBitContext pb;
if ((unsigned)src_size > (unsigned)max_size) {
// av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");
// return -1;
src_size = max_size;
}
mrk = AV_RB32(src);
switch (mrk) {
case DCA_MARKER_RAW_BE:
memcpy(dst, src, src_size);
return src_size;
case DCA_MARKER_RAW_LE:
for (i = 0; i < (src_size + 1) >> 1; i++)
*sdst++ = av_bswap16(*ssrc++);
return src_size;
case DCA_MARKER_14B_BE:
case DCA_MARKER_14B_LE:
init_put_bits(&pb, dst, max_size);
for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
put_bits(&pb, 14, tmp);
}
flush_put_bits(&pb);
return (put_bits_count(&pb) + 7) >> 3;
default:
return AVERROR_INVALIDDATA;
}
}
/**
* Return the number of channels in an ExSS speaker mask (HD)
*/
static int dca_exss_mask2count(int mask)
{
/* count bits that mean speaker pairs twice */
return av_popcount(mask)
+ av_popcount(mask & (
DCA_EXSS_CENTER_LEFT_RIGHT
| DCA_EXSS_FRONT_LEFT_RIGHT
| DCA_EXSS_FRONT_HIGH_LEFT_RIGHT
| DCA_EXSS_WIDE_LEFT_RIGHT
| DCA_EXSS_SIDE_LEFT_RIGHT
| DCA_EXSS_SIDE_HIGH_LEFT_RIGHT
| DCA_EXSS_SIDE_REAR_LEFT_RIGHT
| DCA_EXSS_REAR_LEFT_RIGHT
| DCA_EXSS_REAR_HIGH_LEFT_RIGHT
));
}
/**
* Skip mixing coefficients of a single mix out configuration (HD)
*/
static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
{
int i;
for (i = 0; i < channels; i++) {
int mix_map_mask = get_bits(gb, out_ch);
int num_coeffs = av_popcount(mix_map_mask);
skip_bits_long(gb, num_coeffs * 6);
}
}
/**
* Parse extension substream asset header (HD)
*/
static int dca_exss_parse_asset_header(DCAContext *s)
{
int header_pos = get_bits_count(&s->gb);
int header_size;
int channels;
int embedded_stereo = 0;
int embedded_6ch = 0;
int drc_code_present;
int extensions_mask;
int i, j;
if (get_bits_left(&s->gb) < 16)
return -1;
/* We will parse just enough to get to the extensions bitmask with which
* we can set the profile value. */
header_size = get_bits(&s->gb, 9) + 1;
skip_bits(&s->gb, 3); // asset index
if (s->static_fields) {
if (get_bits1(&s->gb))
skip_bits(&s->gb, 4); // asset type descriptor
if (get_bits1(&s->gb))
skip_bits_long(&s->gb, 24); // language descriptor
if (get_bits1(&s->gb)) {
/* How can one fit 1024 bytes of text here if the maximum value
* for the asset header size field above was 512 bytes? */
int text_length = get_bits(&s->gb, 10) + 1;
if (get_bits_left(&s->gb) < text_length * 8)
return -1;
skip_bits_long(&s->gb, text_length * 8); // info text
}
skip_bits(&s->gb, 5); // bit resolution - 1
skip_bits(&s->gb, 4); // max sample rate code
channels = get_bits(&s->gb, 8) + 1;
if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers
int spkr_remap_sets;
int spkr_mask_size = 16;
int num_spkrs[7];
if (channels > 2)
embedded_stereo = get_bits1(&s->gb);
if (channels > 6)
embedded_6ch = get_bits1(&s->gb);
if (get_bits1(&s->gb)) {
spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
skip_bits(&s->gb, spkr_mask_size); // spkr activity mask
}
spkr_remap_sets = get_bits(&s->gb, 3);
for (i = 0; i < spkr_remap_sets; i++) {
/* std layout mask for each remap set */
num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));
}
for (i = 0; i < spkr_remap_sets; i++) {
int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;
if (get_bits_left(&s->gb) < 0)
return -1;
for (j = 0; j < num_spkrs[i]; j++) {
int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);
int num_dec_ch = av_popcount(remap_dec_ch_mask);
skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes
}
}
} else {
skip_bits(&s->gb, 3); // representation type
}
}
drc_code_present = get_bits1(&s->gb);
if (drc_code_present)
get_bits(&s->gb, 8); // drc code
if (get_bits1(&s->gb))
skip_bits(&s->gb, 5); // dialog normalization code
if (drc_code_present && embedded_stereo)
get_bits(&s->gb, 8); // drc stereo code
if (s->mix_metadata && get_bits1(&s->gb)) {
skip_bits(&s->gb, 1); // external mix
skip_bits(&s->gb, 6); // post mix gain code
if (get_bits(&s->gb, 2) != 3) // mixer drc code
skip_bits(&s->gb, 3); // drc limit
else
skip_bits(&s->gb, 8); // custom drc code
if (get_bits1(&s->gb)) // channel specific scaling
for (i = 0; i < s->num_mix_configs; i++)
skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes
else
skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes
for (i = 0; i < s->num_mix_configs; i++) {
if (get_bits_left(&s->gb) < 0)
return -1;
dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);
if (embedded_6ch)
dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);
if (embedded_stereo)
dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);
}
}
switch (get_bits(&s->gb, 2)) {
case 0: extensions_mask = get_bits(&s->gb, 12); break;
case 1: extensions_mask = DCA_EXT_EXSS_XLL; break;
case 2: extensions_mask = DCA_EXT_EXSS_LBR; break;
case 3: extensions_mask = 0; /* aux coding */ break;
}
/* not parsed further, we were only interested in the extensions mask */
if (get_bits_left(&s->gb) < 0)
return -1;
if (get_bits_count(&s->gb) - header_pos > header_size * 8) {
av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");
return -1;
}
skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));
if (extensions_mask & DCA_EXT_EXSS_XLL)
s->profile = FF_PROFILE_DTS_HD_MA;
else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |
DCA_EXT_EXSS_XXCH))
s->profile = FF_PROFILE_DTS_HD_HRA;
if (!(extensions_mask & DCA_EXT_CORE))
av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)
av_log(s->avctx, AV_LOG_WARNING, "DTS extensions detection mismatch (%d, %d)\n",
extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);
return 0;
}
/**
* Parse extension substream header (HD)
*/
static void dca_exss_parse_header(DCAContext *s)
{
int ss_index;
int blownup;
int num_audiop = 1;
int num_assets = 1;
int active_ss_mask[8];
int i, j;
if (get_bits_left(&s->gb) < 52)
return;
skip_bits(&s->gb, 8); // user data
ss_index = get_bits(&s->gb, 2);
blownup = get_bits1(&s->gb);
skip_bits(&s->gb, 8 + 4 * blownup); // header_size
skip_bits(&s->gb, 16 + 4 * blownup); // hd_size
s->static_fields = get_bits1(&s->gb);
if (s->static_fields) {
skip_bits(&s->gb, 2); // reference clock code
skip_bits(&s->gb, 3); // frame duration code
if (get_bits1(&s->gb))
skip_bits_long(&s->gb, 36); // timestamp
/* a single stream can contain multiple audio assets that can be
* combined to form multiple audio presentations */
num_audiop = get_bits(&s->gb, 3) + 1;
if (num_audiop > 1) {
av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");
/* ignore such streams for now */
return;
}
num_assets = get_bits(&s->gb, 3) + 1;
if (num_assets > 1) {
av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");
/* ignore such streams for now */
return;
}
for (i = 0; i < num_audiop; i++)
active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);
for (i = 0; i < num_audiop; i++)
for (j = 0; j <= ss_index; j++)
if (active_ss_mask[i] & (1 << j))
skip_bits(&s->gb, 8); // active asset mask
s->mix_metadata = get_bits1(&s->gb);
if (s->mix_metadata) {
int mix_out_mask_size;
skip_bits(&s->gb, 2); // adjustment level
mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
s->num_mix_configs = get_bits(&s->gb, 2) + 1;
for (i = 0; i < s->num_mix_configs; i++) {
int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);
s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);
}
}
}
for (i = 0; i < num_assets; i++)
skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size
for (i = 0; i < num_assets; i++) {
if (dca_exss_parse_asset_header(s))
return;
}
/* not parsed further, we were only interested in the extensions mask
* from the asset header */
}
/**
* Main frame decoding function
* FIXME add arguments
*/
static int dca_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
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;
int16_t *samples_s16;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
s->xch_present = 0;
s->dca_buffer_size = 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;
}
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
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;
avctx->frame_size = s->sample_blocks * 32;
s->profile = FF_PROFILE_DTS;
for (i = 0; i < (s->sample_blocks / 8); i++) {
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 */
num_core_channels = s->prim_channels;
if (s->ext_coding)
s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
else
s->core_ext_mask = 0;
core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
/* 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);
while(core_ss_end - get_bits_count(&s->gb) >= 32) {
uint32_t bits = get_bits_long(&s->gb, 32);
switch(bits) {
case 0x5a5a5a5a: {
int ext_amode, xch_fsize;
s->xch_base_channel = s->prim_channels;
/* 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;
/* skip length-to-end-of-frame field for the moment */
skip_bits(&s->gb, 10);
s->core_ext_mask |= DCA_EXT_XCH;
/* extension amode should == 1, number of channels in extension */
/* AFAIK XCh is not used for more channels */
if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
" supported!\n",ext_amode);
continue;
}
/* 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;
}
case 0x47004a03:
/* 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;
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));
s->core_ext_mask |= DCA_EXT_X96;
break;
}
}
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
}
} 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;
/* check for ExSS (HD part) */
if (s->dca_buffer_size - s->frame_size > 32
&& get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
dca_exss_parse_header(s);
avctx->profile = s->profile;
channels = s->prim_channels + !!s->lfe;
if (s->amode<16) {
avctx->channel_layout = dca_core_channel_layout[s->amode];
if (s->xch_present && (!avctx->request_channels ||
avctx->request_channels > num_core_channels + !!s->lfe)) {
avctx->channel_layout |= AV_CH_BACK_CENTER;
if (s->lfe) {
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
} else {
s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
}
} else {
channels = num_core_channels + !!s->lfe;
s->xch_present = 0; /* disable further xch processing */
if (s->lfe) {
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
s->channel_order_tab = dca_channel_reorder_lfe[s->amode];
} else
s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
}
if (channels > !!s->lfe &&
s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
return AVERROR_INVALIDDATA;
if (avctx->request_channels == 2 && s->prim_channels > 2) {
channels = 2;
s->output = DCA_STEREO;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
}
else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };
s->channel_order_tab = dca_channel_order_native;
}
} else {
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
return AVERROR_INVALIDDATA;
}
if (avctx->channels != channels) {
if (avctx->channels)
av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);
avctx->channels = channels;
}
/* get output buffer */
s->frame.nb_samples = 256 * (s->sample_blocks / 8);
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples_flt = (float *)s->frame.data[0];
samples_s16 = (int16_t *)s->frame.data[0];
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
dca_filter_channels(s, i);
/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
if((s->source_pcm_res & 1) && s->xch_present) {
float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
float* lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
channels);
samples_flt += 256 * channels;
} else {
s->fmt_conv.float_to_int16_interleave(samples_s16,
s->samples_chanptr, 256,
channels);
samples_s16 += 256 * channels;
}
}
/* update lfe history */
lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
for (i = 0; i < 2 * s->lfe * 4; i++) {
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
}
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
/**
* DCA initialization
*
* @param avctx pointer to the AVCodecContext
*/
static av_cold int dca_decode_init(AVCodecContext * avctx)
{
DCAContext *s = avctx->priv_data;
int i;
s->avctx = avctx;
dca_init_vlcs();
dsputil_init(&s->dsp, avctx);
ff_mdct_init(&s->imdct, 6, 1, 1.0);
ff_synth_filter_init(&s->synth);
ff_dcadsp_init(&s->dcadsp);
ff_fmt_convert_init(&s->fmt_conv, avctx);
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
s->scale_bias = 1.0 / 32768.0;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
s->scale_bias = 1.0;
}
/* allow downmixing to stereo */
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
avctx->request_channels == 2) {
avctx->channels = avctx->request_channels;
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static av_cold int dca_decode_end(AVCodecContext * avctx)
{
DCAContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct);
return 0;
}
static const AVProfile profiles[] = {
{ FF_PROFILE_DTS, "DTS" },
{ FF_PROFILE_DTS_ES, "DTS-ES" },
{ FF_PROFILE_DTS_96_24, "DTS 96/24" },
{ FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
{ FF_PROFILE_DTS_HD_MA, "DTS-HD MA" },
{ FF_PROFILE_UNKNOWN },
};
AVCodec ff_dca_decoder = {
.name = "dca",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = dca_decode_init,
.decode = dca_decode_frame,
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.profiles = NULL_IF_CONFIG_SMALL(profiles),
};
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