--- /dev/null
- * This file is part of Libav.
+ /*
+ * DCA compatible decoder
+ * Copyright (C) 2004 Gildas Bazin
+ * Copyright (C) 2004 Benjamin Zores
+ * Copyright (C) 2006 Benjamin Larsson
+ * Copyright (C) 2007 Konstantin Shishkov
+ *
- * Libav is free software; you can redistribute it and/or
++ * This file is part of FFmpeg.
+ *
- * Libav is distributed in the hope that it will be useful,
++ * 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.
+ *
- * License along with Libav; if not, write to the Free Software
++ * 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
-#define DCA_SUBBANDS (32)
++ * 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/float_dsp.h"
+ #include "libavutil/intmath.h"
+ #include "libavutil/intreadwrite.h"
+ #include "libavutil/mathematics.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 "dca_parser.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)
- int is_channels_set; ///< check for if the channel number is already set
++#define DCA_SUBBANDS (64)
+ #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)
++#define DCA_CHSETS_MAX (4)
++#define DCA_CHSET_CHANS_MAX (8)
+
+ 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 DCAXxchSpeakerMask {
++ DCA_XXCH_FRONT_CENTER = 0x0000001,
++ DCA_XXCH_FRONT_LEFT = 0x0000002,
++ DCA_XXCH_FRONT_RIGHT = 0x0000004,
++ DCA_XXCH_SIDE_REAR_LEFT = 0x0000008,
++ DCA_XXCH_SIDE_REAR_RIGHT = 0x0000010,
++ DCA_XXCH_LFE1 = 0x0000020,
++ DCA_XXCH_REAR_CENTER = 0x0000040,
++ DCA_XXCH_SURROUND_REAR_LEFT = 0x0000080,
++ DCA_XXCH_SURROUND_REAR_RIGHT = 0x0000100,
++ DCA_XXCH_SIDE_SURROUND_LEFT = 0x0000200,
++ DCA_XXCH_SIDE_SURROUND_RIGHT = 0x0000400,
++ DCA_XXCH_FRONT_CENTER_LEFT = 0x0000800,
++ DCA_XXCH_FRONT_CENTER_RIGHT = 0x0001000,
++ DCA_XXCH_FRONT_HIGH_LEFT = 0x0002000,
++ DCA_XXCH_FRONT_HIGH_CENTER = 0x0004000,
++ DCA_XXCH_FRONT_HIGH_RIGHT = 0x0008000,
++ DCA_XXCH_LFE2 = 0x0010000,
++ DCA_XXCH_SIDE_FRONT_LEFT = 0x0020000,
++ DCA_XXCH_SIDE_FRONT_RIGHT = 0x0040000,
++ DCA_XXCH_OVERHEAD = 0x0080000,
++ DCA_XXCH_SIDE_HIGH_LEFT = 0x0100000,
++ DCA_XXCH_SIDE_HIGH_RIGHT = 0x0200000,
++ DCA_XXCH_REAR_HIGH_CENTER = 0x0400000,
++ DCA_XXCH_REAR_HIGH_LEFT = 0x0800000,
++ DCA_XXCH_REAR_HIGH_RIGHT = 0x1000000,
++ DCA_XXCH_REAR_LOW_CENTER = 0x2000000,
++ DCA_XXCH_REAR_LOW_LEFT = 0x4000000,
++ DCA_XXCH_REAR_LOW_RIGHT = 0x8000000,
++};
++
++static const uint32_t map_xxch_to_native[28] = {
++ AV_CH_FRONT_CENTER,
++ AV_CH_FRONT_LEFT,
++ AV_CH_FRONT_RIGHT,
++ AV_CH_SIDE_LEFT,
++ AV_CH_SIDE_RIGHT,
++ AV_CH_LOW_FREQUENCY,
++ AV_CH_BACK_CENTER,
++ AV_CH_BACK_LEFT,
++ AV_CH_BACK_RIGHT,
++ AV_CH_SIDE_LEFT, /* side surround left -- dup sur side L */
++ AV_CH_SIDE_RIGHT, /* side surround right -- dup sur side R */
++ AV_CH_FRONT_LEFT_OF_CENTER,
++ AV_CH_FRONT_RIGHT_OF_CENTER,
++ AV_CH_TOP_FRONT_LEFT,
++ AV_CH_TOP_FRONT_CENTER,
++ AV_CH_TOP_FRONT_RIGHT,
++ AV_CH_LOW_FREQUENCY, /* lfe2 -- duplicate lfe1 position */
++ AV_CH_FRONT_LEFT_OF_CENTER, /* side front left -- dup front cntr L */
++ AV_CH_FRONT_RIGHT_OF_CENTER,/* side front right -- dup front cntr R */
++ AV_CH_TOP_CENTER, /* overhead */
++ AV_CH_TOP_FRONT_LEFT, /* side high left -- dup */
++ AV_CH_TOP_FRONT_RIGHT, /* side high right -- dup */
++ AV_CH_TOP_BACK_CENTER,
++ AV_CH_TOP_BACK_LEFT,
++ AV_CH_TOP_BACK_RIGHT,
++ AV_CH_BACK_CENTER, /* rear low center -- dup */
++ AV_CH_BACK_LEFT, /* rear low left -- dup */
++ AV_CH_BACK_RIGHT /* read low right -- dup */
++};
++
+ 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
-static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
+ 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
+
++ /* XXCH extension information */
++ int xxch_chset;
++ int xxch_nbits_spk_mask;
++ uint32_t xxch_core_spkmask;
++ uint32_t xxch_spk_masks[4]; /* speaker masks, last element is core mask */
++ int xxch_chset_nch[4];
++ float xxch_dmix_sf[DCA_CHSETS_MAX];
++
++ uint32_t xxch_downmix; /* downmix enabled per channel set */
++ uint32_t xxch_dmix_embedded; /* lower layer has mix pre-embedded, per chset */
++ float xxch_dmix_coeff[DCA_PRIM_CHANNELS_MAX][32]; /* worst case sizing */
++
++ int8_t xxch_order_tab[32];
++ int8_t lfe_index;
++
+ /* 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
+ AVFloatDSPContext fdsp;
+ 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);
+ }
+
- s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
++static inline int dca_xxch2index(DCAContext *s, int xxch_ch)
++{
++ int i, base, mask;
++
++ /* locate channel set containing the channel */
++ for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1);
++ i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i])
++ base += av_popcount(mask);
++
++ return base + av_popcount(mask & (xxch_ch - 1));
++}
++
++static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
++ int xxch)
+ {
+ 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 };
++ int hdr_pos = 0, hdr_size = 0;
++ float sign, mag, scale_factor;
++ int this_chans, acc_mask;
++ int embedded_downmix;
++ int nchans, mask[8];
++ int coeff, ichan;
++
++ /* xxch has arbitrary sized audio coding headers */
++ if (xxch) {
++ hdr_pos = get_bits_count(&s->gb);
++ hdr_size = get_bits(&s->gb, 7) + 1;
++ }
+
- if (s->crc_present) {
- /* Audio header CRC check */
- get_bits(&s->gb, 16);
++ nchans = get_bits(&s->gb, 3) + 1;
++ s->total_channels = nchans + base_channel;
+ s->prim_channels = s->total_channels;
+
++ /* obtain speaker layout mask & downmix coefficients for XXCH */
++ if (xxch) {
++ acc_mask = s->xxch_core_spkmask;
++
++ this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6;
++ s->xxch_spk_masks[s->xxch_chset] = this_chans;
++ s->xxch_chset_nch[s->xxch_chset] = nchans;
++
++ for (i = 0; i <= s->xxch_chset; i++)
++ acc_mask |= s->xxch_spk_masks[i];
++
++ /* check for downmixing information */
++ if (get_bits1(&s->gb)) {
++ embedded_downmix = get_bits1(&s->gb);
++ scale_factor =
++ 1.0f / dca_downmix_scale_factors[(get_bits(&s->gb, 6) - 1) << 2];
++
++ s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
++
++ for (i = base_channel; i < s->prim_channels; i++) {
++ s->xxch_downmix |= (1 << i);
++ mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask);
++ }
++
++ for (j = base_channel; j < s->prim_channels; j++) {
++ memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0]));
++ s->xxch_dmix_embedded |= (embedded_downmix << j);
++ for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
++ if (mask[j] & (1 << i)) {
++ if ((1 << i) == DCA_XXCH_LFE1) {
++ av_log(s->avctx, AV_LOG_WARNING,
++ "DCA-XXCH: dmix to LFE1 not supported.\n");
++ continue;
++ }
++
++ coeff = get_bits(&s->gb, 7);
++ sign = (coeff & 64) ? 1.0 : -1.0;
++ mag = dca_downmix_scale_factors[((coeff & 63) - 1) << 2];
++ ichan = dca_xxch2index(s, 1 << i);
++ s->xxch_dmix_coeff[j][ichan] = sign * mag;
++ }
++ }
++ }
++ }
++ }
++
+ 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)];
+
- s->downmix = get_bits(&s->gb, 1);
++ if (!xxch) {
++ if (s->crc_present) {
++ /* Audio header CRC check */
++ get_bits(&s->gb, 16);
++ }
++ } else {
++ /* Skip to the end of the header, also ignore CRC if present */
++ i = get_bits_count(&s->gb);
++ if (hdr_pos + 8 * hdr_size > i)
++ skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i);
+ }
+
+ 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 = avpriv_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;
+
- return dca_parse_audio_coding_header(s, 0);
++ s->downmix = get_bits(&s->gb, 1); /* note: this is FixedBit == 0 */
+ 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;
+
- for (j = base_channel; j < s->prim_channels; j++) {
++ return dca_parse_audio_coding_header(s, 0, 0);
+ }
+
+
+ static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
+ {
+ if (level < 5) {
+ /* huffman encoded */
+ value += get_bitalloc(gb, &dca_scalefactor, level);
+ value = av_clip(value, 0, (1 << log2range) - 1);
+ } else if (level < 8) {
+ if (level + 1 > log2range) {
+ skip_bits(gb, level + 1 - log2range);
+ value = get_bits(gb, log2range);
+ } else {
+ value = get_bits(gb, level + 1);
+ }
+ }
+ 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, log_size;
+
+ 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;
+ log_size = 7;
+ } else {
+ scale_table = scale_factor_quant6;
+ log_size = 6;
+ }
+
+ /* 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, log_size);
+ 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, log_size);
+ 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], 64 /* bias */, 7);
+ 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;
+ if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid channel mode %d\n", am);
+ return AVERROR_INVALIDDATA;
+ }
- skip_bits(&s->gb, 1);
- s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 7)];
++ for (j = base_channel; j < FFMIN(s->prim_channels, FF_ARRAY_ELEMS(dca_default_coeffs[am])); 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) {
++ int quant7;
+ /* 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->samples[256 * dca_lfe_index[s->amode]],
++ quant7 = get_bits(&s->gb, 8);
++ if (quant7 > 127) {
++ av_log_ask_for_sample(s->avctx, "LFEScaleIndex larger than 127\n");
++ return AVERROR_INVALIDDATA;
++ }
++ s->lfe_scale_factor = scale_factor_quant7[quant7];
+
+ /* 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),
- int channels;
++ &s->samples[256 * s->lfe_index],
+ (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;
+ }
+
+ /**
+ * 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 extensions_mask;
++ int channels = 0;
+ int embedded_stereo = 0;
+ int embedded_6ch = 0;
+ int drc_code_present;
- skip_bits(&s->gb, 8 + 4 * blownup); // header_size
++ int av_uninit(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;
+ }
+
++static int dca_xbr_parse_frame(DCAContext *s)
++{
++ int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2];
++ int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX];
++ int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS];
++ int anctemp[DCA_CHSET_CHANS_MAX];
++ int chset_fsize[DCA_CHSETS_MAX];
++ int n_xbr_ch[DCA_CHSETS_MAX];
++ int hdr_size, num_chsets, xbr_tmode, hdr_pos;
++ int i, j, k, l, chset, chan_base;
++
++ av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n");
++
++ /* get bit position of sync header */
++ hdr_pos = get_bits_count(&s->gb) - 32;
++
++ hdr_size = get_bits(&s->gb, 6) + 1;
++ num_chsets = get_bits(&s->gb, 2) + 1;
++
++ for(i = 0; i < num_chsets; i++)
++ chset_fsize[i] = get_bits(&s->gb, 14) + 1;
++
++ xbr_tmode = get_bits1(&s->gb);
++
++ for(i = 0; i < num_chsets; i++) {
++ n_xbr_ch[i] = get_bits(&s->gb, 3) + 1;
++ k = get_bits(&s->gb, 2) + 5;
++ for(j = 0; j < n_xbr_ch[i]; j++)
++ active_bands[i][j] = get_bits(&s->gb, k) + 1;
++ }
++
++ /* skip to the end of the header */
++ i = get_bits_count(&s->gb);
++ if(hdr_pos + hdr_size * 8 > i)
++ skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
++
++ /* loop over the channel data sets */
++ /* only decode as many channels as we've decoded base data for */
++ for(chset = 0, chan_base = 0;
++ chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->prim_channels;
++ chan_base += n_xbr_ch[chset++]) {
++ int start_posn = get_bits_count(&s->gb);
++ int subsubframe = 0;
++ int subframe = 0;
++
++ /* loop over subframes */
++ for (k = 0; k < (s->sample_blocks / 8); k++) {
++ /* parse header if we're on first subsubframe of a block */
++ if(subsubframe == 0) {
++ /* Parse subframe header */
++ for(i = 0; i < n_xbr_ch[chset]; i++) {
++ anctemp[i] = get_bits(&s->gb, 2) + 2;
++ }
++
++ for(i = 0; i < n_xbr_ch[chset]; i++) {
++ get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]);
++ }
++
++ for(i = 0; i < n_xbr_ch[chset]; i++) {
++ anctemp[i] = get_bits(&s->gb, 3);
++ if(anctemp[i] < 1) {
++ av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n");
++ return AVERROR_INVALIDDATA;
++ }
++ }
++
++ /* generate scale factors */
++ for(i = 0; i < n_xbr_ch[chset]; i++) {
++ const uint32_t *scale_table;
++ int nbits;
++
++ if (s->scalefactor_huffman[chan_base+i] == 6) {
++ scale_table = scale_factor_quant7;
++ } else {
++ scale_table = scale_factor_quant6;
++ }
++
++ nbits = anctemp[i];
++
++ for(j = 0; j < active_bands[chset][i]; j++) {
++ if(abits_high[i][j] > 0) {
++ scale_table_high[i][j][0] =
++ scale_table[get_bits(&s->gb, nbits)];
++
++ if(xbr_tmode && s->transition_mode[i][j]) {
++ scale_table_high[i][j][1] =
++ scale_table[get_bits(&s->gb, nbits)];
++ }
++ }
++ }
++ }
++ }
++
++ /* decode audio array for this block */
++ for(i = 0; i < n_xbr_ch[chset]; i++) {
++ for(j = 0; j < active_bands[chset][i]; j++) {
++ const int xbr_abits = abits_high[i][j];
++ const float quant_step_size = lossless_quant_d[xbr_abits];
++ const int sfi = xbr_tmode && s->transition_mode[i][j] && subsubframe >= s->transition_mode[i][j];
++ const float rscale = quant_step_size * scale_table_high[i][j][sfi];
++ float *subband_samples = s->subband_samples[k][chan_base+i][j];
++ int block[8];
++
++ if(xbr_abits <= 0)
++ continue;
++
++ if(xbr_abits > 7) {
++ get_array(&s->gb, block, 8, xbr_abits - 3);
++ } else {
++ int block_code1, block_code2, size, levels, err;
++
++ size = abits_sizes[xbr_abits - 1];
++ levels = abits_levels[xbr_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: DTS-XBR: block code look-up failed\n");
++ return AVERROR_INVALIDDATA;
++ }
++ }
++
++ /* scale & sum into subband */
++ for(l = 0; l < 8; l++)
++ subband_samples[l] += (float)block[l] * rscale;
++ }
++ }
++
++ /* check DSYNC marker */
++ if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) {
++ if(get_bits(&s->gb, 16) != 0xffff) {
++ av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n");
++ return AVERROR_INVALIDDATA;
++ }
++ }
++
++ /* advance sub-sub-frame index */
++ if(++subsubframe >= s->subsubframes[subframe]) {
++ subsubframe = 0;
++ subframe++;
++ }
++ }
++
++ /* skip to next channel set */
++ i = get_bits_count(&s->gb);
++ if(start_posn + chset_fsize[chset] * 8 != i) {
++ j = start_posn + chset_fsize[chset] * 8 - i;
++ if(j < 0 || j >= 8)
++ av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set,"
++ " skipping further than expected (%d bits)\n", j);
++ skip_bits_long(&s->gb, j);
++ }
++ }
++
++ return 0;
++}
++
++/* parse initial header for XXCH and dump details */
++static int dca_xxch_decode_frame(DCAContext *s)
++{
++ int hdr_size, chhdr_crc, spkmsk_bits, num_chsets, core_spk, hdr_pos;
++ int i, chset, base_channel, chstart, fsize[8];
++
++ /* assume header word has already been parsed */
++ hdr_pos = get_bits_count(&s->gb) - 32;
++ hdr_size = get_bits(&s->gb, 6) + 1;
++ chhdr_crc = get_bits1(&s->gb);
++ spkmsk_bits = get_bits(&s->gb, 5) + 1;
++ num_chsets = get_bits(&s->gb, 2) + 1;
++
++ for (i = 0; i < num_chsets; i++)
++ fsize[i] = get_bits(&s->gb, 14) + 1;
++
++ core_spk = get_bits(&s->gb, spkmsk_bits);
++ s->xxch_core_spkmask = core_spk;
++ s->xxch_nbits_spk_mask = spkmsk_bits;
++ s->xxch_downmix = 0;
++ s->xxch_dmix_embedded = 0;
++
++ /* skip to the end of the header */
++ i = get_bits_count(&s->gb);
++ if (hdr_pos + hdr_size * 8 > i)
++ skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
++
++ for (chset = 0; chset < num_chsets; chset++) {
++ chstart = get_bits_count(&s->gb);
++ base_channel = s->prim_channels;
++ s->xxch_chset = chset;
++
++ /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs.
++ 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */
++ dca_parse_audio_coding_header(s, base_channel, 1);
++
++ /* decode channel data */
++ for (i = 0; i < (s->sample_blocks / 8); i++) {
++ if (dca_decode_block(s, base_channel, i)) {
++ av_log(s->avctx, AV_LOG_ERROR,
++ "Error decoding DTS-XXCH extension\n");
++ continue;
++ }
++ }
++
++ /* skip to end of this section */
++ i = get_bits_count(&s->gb);
++ if (chstart + fsize[chset] * 8 > i)
++ skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i);
++ }
++ s->xxch_chset = num_chsets;
++
++ return 0;
++}
++
+ /**
+ * Parse extension substream header (HD)
+ */
+ static void dca_exss_parse_header(DCAContext *s)
+ {
++ int asset_size[8];
+ int ss_index;
+ int blownup;
+ int num_audiop = 1;
+ int num_assets = 1;
+ int active_ss_mask[8];
+ int i, j;
++ int start_posn;
++ int hdrsize;
++ uint32_t mkr;
+
+ if (get_bits_left(&s->gb) < 52)
+ return;
+
++ start_posn = get_bits_count(&s->gb) - 32;
++
+ skip_bits(&s->gb, 8); // user data
+ ss_index = get_bits(&s->gb, 2);
+
+ blownup = get_bits1(&s->gb);
- skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size
++ hdrsize = get_bits(&s->gb, 8 + 4 * blownup) + 1; // 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++)
-
++ asset_size[i] = get_bits_long(&s->gb, 16 + 4 * blownup);
+
+ 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 */
++
++ if (num_assets > 0) {
++ j = get_bits_count(&s->gb);
++ if (start_posn + hdrsize * 8 > j)
++ skip_bits_long(&s->gb, start_posn + hdrsize * 8 - j);
++
++ for (i = 0; i < num_assets; i++) {
++ start_posn = get_bits_count(&s->gb);
++ mkr = get_bits_long(&s->gb, 32);
++
++ /* parse extensions that we know about */
++ if (mkr == 0x655e315e) {
++ dca_xbr_parse_frame(s);
++ } else if (mkr == 0x47004a03) {
++ dca_xxch_decode_frame(s);
++ s->core_ext_mask |= DCA_EXT_XXCH; /* xxx use for chan reordering */
++ } else {
++ av_log(s->avctx, AV_LOG_DEBUG,
++ "DTS-ExSS: unknown marker = 0x%08x\n", mkr);
++ }
++
++ /* skip to end of block */
++ j = get_bits_count(&s->gb);
++ if (start_posn + asset_size[i] * 8 > j)
++ skip_bits_long(&s->gb, start_posn + asset_size[i] * 8 - j);
++ }
++ }
+ }
+
+ /**
+ * 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;
- float *samples_flt;
++ int channel_mask;
++ int channel_layout;
+ int lfe_samples;
+ int num_core_channels = 0;
+ int i, ret;
- int channels;
++ float *samples_flt;
++ float *src_chan;
++ float *dst_chan;
+ int16_t *samples_s16;
+ DCAContext *s = avctx->priv_data;
-
+ int core_ss_end;
- if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
++ int channels;
++ float scale;
++ int achan;
++ int chset;
++ int mask;
++ int lavc;
++ int posn;
++ int j, k;
++ int ch;
++ int endch;
+
+ s->xch_present = 0;
+
+ s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
+ DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
+ if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
+ av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ 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;
+
+ 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 */
- dca_parse_audio_coding_header(s, s->xch_base_channel);
++ if (s->core_ext_mask < 0 || s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) {
+
+ /* 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(number of channels in extension) should be 1 */
+ /* AFAIK XCh is not used for more channels */
+ if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
+ av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
+ " supported!\n", ext_amode);
+ continue;
+ }
+
+ /* much like core primary audio coding header */
- 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];
++ dca_parse_audio_coding_header(s, s->xch_base_channel, 0);
+
+ 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;
++ dca_xxch_decode_frame(s);
+ 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;
+
- s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
++ /* If we have XXCH then the channel layout is managed differently */
++ /* note that XLL will also have another way to do things */
++ if (!(s->core_ext_mask & DCA_EXT_XXCH)
++ || (s->core_ext_mask & DCA_EXT_XXCH && avctx->request_channels > 0
++ && avctx->request_channels
++ < num_core_channels + !!s->lfe + s->xxch_chset_nch[0]))
++ { /* xxx should also do MA extensions */
++ 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 {
- 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];
++ 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;
++ }
++ s->lfe_index = dca_lfe_index[s->amode];
+ } else {
++ av_log(avctx, AV_LOG_ERROR,
++ "Non standard configuration %d !\n", s->amode);
++ return AVERROR_INVALIDDATA;
++ }
++
++ s->xxch_downmix = 0;
++ } else {
++ /* we only get here if an XXCH channel set can be added to the mix */
++ channel_mask = s->xxch_core_spkmask;
++
++ if (avctx->request_channels > 0
++ && avctx->request_channels < s->prim_channels) {
+ channels = num_core_channels + !!s->lfe;
- if (channels > !!s->lfe &&
- s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
- return AVERROR_INVALIDDATA;
++ for (i = 0; i < s->xxch_chset && channels + s->xxch_chset_nch[i]
++ <= avctx->request_channels; i++) {
++ channels += s->xxch_chset_nch[i];
++ channel_mask |= s->xxch_spk_masks[i];
++ }
++ } else {
++ channels = s->prim_channels + !!s->lfe;
++ for (i = 0; i < s->xxch_chset; i++) {
++ channel_mask |= s->xxch_spk_masks[i];
++ }
+ }
+
- if (avctx->request_channels == 2 && s->prim_channels > 2) {
- channels = 2;
- s->output = DCA_STEREO;
- avctx->channel_layout = AV_CH_LAYOUT_STEREO;
++ /* Given the DTS spec'ed channel mask, generate an avcodec version */
++ channel_layout = 0;
++ for (i = 0; i < s->xxch_nbits_spk_mask; ++i) {
++ if (channel_mask & (1 << i)) {
++ channel_layout |= map_xxch_to_native[i];
++ }
++ }
+
- } else {
- av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
- return AVERROR_INVALIDDATA;
- }
++ /* make sure that we have managed to get equivelant dts/avcodec channel
++ * masks in some sense -- unfortunately some channels could overlap */
++ if (av_popcount(channel_mask) != av_popcount(channel_layout)) {
++ av_log(avctx, AV_LOG_DEBUG,
++ "DTS-XXCH: Inconsistant avcodec/dts channel layouts\n");
++ return AVERROR_INVALIDDATA;
+ }
- /* There is nothing that prevents a dts frame to change channel configuration
- but Libav doesn't support that so only set the channels if it is previously
- unset. Ideally during the first probe for channels the crc should be checked
- and only set avctx->channels when the crc is ok. Right now the decoder could
- set the channels based on a broken first frame.*/
- if (s->is_channels_set == 0) {
- s->is_channels_set = 1;
- avctx->channels = channels;
+
++ avctx->channel_layout = channel_layout;
++
++ if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) {
++ /* Estimate DTS --> avcodec ordering table */
++ for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) {
++ mask = chset >= 0 ? s->xxch_spk_masks[chset]
++ : s->xxch_core_spkmask;
++ for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
++ if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) {
++ lavc = map_xxch_to_native[i];
++ posn = av_popcount(channel_layout & (lavc - 1));
++ s->xxch_order_tab[j++] = posn;
++ }
++ }
++ }
+
- av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "
- "channels changing in stream. Skipping frame.\n");
- return AVERROR_PATCHWELCOME;
++ s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1));
++ } else { /* native ordering */
++ for (i = 0; i < channels; i++)
++ s->xxch_order_tab[i] = i;
++
++ s->lfe_index = channels - 1;
++ }
++
++ s->channel_order_tab = s->xxch_order_tab;
+ }
++
+ 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->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
+ s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
+ }
++
++ /* If stream contains XXCH, we might need to undo an embedded downmix */
++ if (s->xxch_dmix_embedded) {
++ /* Loop over channel sets in turn */
++ ch = num_core_channels;
++ for (chset = 0; chset < s->xxch_chset; chset++) {
++ endch = ch + s->xxch_chset_nch[chset];
++ mask = s->xxch_dmix_embedded;
++
++ /* undo downmix */
++ for (j = ch; j < endch; j++) {
++ if (mask & (1 << j)) { /* this channel has been mixed-out */
++ src_chan = s->samples + s->channel_order_tab[j] * 256;
++ for (k = 0; k < endch; k++) {
++ achan = s->channel_order_tab[k];
++ scale = s->xxch_dmix_coeff[j][k];
++ if (scale != 0.0) {
++ dst_chan = s->samples + achan * 256;
++ s->fdsp.vector_fmac_scalar(dst_chan, src_chan,
++ -scale, 256);
++ }
++ }
++ }
++ }
++
++ /* if a downmix has been embedded then undo the pre-scaling */
++ if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) {
++ scale = s->xxch_dmix_sf[chset];
++
++ for (j = 0; j < ch; j++) {
++ src_chan = s->samples + s->channel_order_tab[j] * 256;
++ for (k = 0; k < 256; k++)
++ src_chan[k] *= scale;
++ }
++
++ /* LFE channel is always part of core, scale if it exists */
++ if (s->lfe) {
++ src_chan = s->samples + s->lfe_index * 256;
++ for (k = 0; k < 256; k++)
++ src_chan[k] *= scale;
++ }
++ }
++
++ ch = endch;
++ }
++
++ }
+
+ 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();
+
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+ 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),
+ };
--- /dev/null
- if (dca_sample_rates[i] && (dca_sample_rates[i] == avctx->sample_rate))
+/*
+ * DCA encoder
+ * Copyright (C) 2008 Alexander E. Patrakov
+ * 2010 Benjamin Larsson
+ * 2011 Xiang Wang
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/common.h"
+#include "libavutil/avassert.h"
+#include "libavutil/audioconvert.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "internal.h"
+#include "put_bits.h"
+#include "dcaenc.h"
+#include "dcadata.h"
++#include "dca.h"
+
+#undef NDEBUG
+
+#define MAX_CHANNELS 6
+#define DCA_SUBBANDS_32 32
+#define DCA_MAX_FRAME_SIZE 16383
+#define DCA_HEADER_SIZE 13
+
+#define DCA_SUBBANDS 32 ///< Subband activity count
+#define QUANTIZER_BITS 16
+#define SUBFRAMES 1
+#define SUBSUBFRAMES 4
+#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
+#define LFE_BITS 8
+#define LFE_INTERPOLATION 64
+#define LFE_PRESENT 2
+#define LFE_MISSING 0
+
+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 },
+ { 1, 2, 0, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, -1, 2, -1, -1, -1, -1, -1 },
+ { 1, 2, 0, -1, 3, -1, -1, -1, -1 },
+ { 0, 1, -1, 2, 3, -1, -1, -1, -1 },
+ { 1, 2, 0, -1, 3, 4, -1, -1, -1 },
+ { 2, 3, -1, 0, 1, 4, 5, -1, -1 },
+ { 1, 2, 0, -1, 3, 4, 5, -1, -1 },
+ { 0, -1, 4, 5, 2, 3, 1, -1, -1 },
+ { 3, 4, 1, -1, 0, 2, 5, 6, -1 },
+ { 2, 3, -1, 5, 7, 0, 1, 4, 6 },
+ { 3, 4, 1, -1, 0, 2, 5, 7, 6 },
+};
+
+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 },
+ { 1, 2, 0, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
+ { 1, 2, 0, 3, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
+ { 1, 2, 0, 3, 4, -1, -1, -1, -1 },
+ { 2, 3, 0, 1, 4, 5, -1, -1, -1 },
+ { 1, 2, 0, 3, 4, 5, -1, -1, -1 },
+ { 0, 4, 5, 2, 3, 1, -1, -1, -1 },
+ { 3, 4, 1, 0, 2, 5, 6, -1, -1 },
+ { 2, 3, 5, 7, 0, 1, 4, 6, -1 },
+ { 3, 4, 1, 0, 2, 5, 7, 6, -1 },
+};
+
+typedef struct {
+ PutBitContext pb;
+ int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
+ int start[MAX_CHANNELS];
+ int frame_size;
+ int prim_channels;
+ int lfe_channel;
+ int sample_rate_code;
+ int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
+ int lfe_scale_factor;
+ int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
+
+ int a_mode; ///< audio channels arrangement
+ int num_channel;
+ int lfe_state;
+ int lfe_offset;
+ const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
+
+ int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
+ int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
+} DCAContext;
+
+static int32_t cos_table[128];
+
+static inline int32_t mul32(int32_t a, int32_t b)
+{
+ int64_t r = (int64_t) a * b;
+ /* round the result before truncating - improves accuracy */
+ return (r + 0x80000000) >> 32;
+}
+
+/* Integer version of the cosine modulated Pseudo QMF */
+
+static void qmf_init(void)
+{
+ int i;
+ int32_t c[17], s[17];
+ s[0] = 0; /* sin(index * PI / 64) * 0x7fffffff */
+ c[0] = 0x7fffffff; /* cos(index * PI / 64) * 0x7fffffff */
+
+ for (i = 1; i <= 16; i++) {
+ s[i] = 2 * (mul32(c[i - 1], 105372028) + mul32(s[i - 1], 2144896908));
+ c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
+ }
+
+ for (i = 0; i < 16; i++) {
+ cos_table[i ] = c[i] >> 3; /* avoid output overflow */
+ cos_table[i + 16] = s[16 - i] >> 3;
+ cos_table[i + 32] = -s[i] >> 3;
+ cos_table[i + 48] = -c[16 - i] >> 3;
+ cos_table[i + 64] = -c[i] >> 3;
+ cos_table[i + 80] = -s[16 - i] >> 3;
+ cos_table[i + 96] = s[i] >> 3;
+ cos_table[i + 112] = c[16 - i] >> 3;
+ }
+}
+
+static int32_t band_delta_factor(int band, int sample_num)
+{
+ int index = band * (2 * sample_num + 1);
+ if (band == 0)
+ return 0x07ffffff;
+ else
+ return cos_table[index & 127];
+}
+
+static void add_new_samples(DCAContext *c, const int32_t *in,
+ int count, int channel)
+{
+ int i;
+
+ /* Place new samples into the history buffer */
+ for (i = 0; i < count; i++) {
+ c->history[channel][c->start[channel] + i] = in[i];
+ av_assert0(c->start[channel] + i < 512);
+ }
+ c->start[channel] += count;
+ if (c->start[channel] == 512)
+ c->start[channel] = 0;
+ av_assert0(c->start[channel] < 512);
+}
+
+static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
+ int channel)
+{
+ int band, i, j, k;
+ int32_t resp;
+ int32_t accum[DCA_SUBBANDS_32] = {0};
+
+ add_new_samples(c, in, DCA_SUBBANDS_32, channel);
+
+ /* Calculate the dot product of the signal with the (possibly inverted)
+ reference decoder's response to this vector:
+ (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
+ so that -1.0 cancels 1.0 from the previous step */
+
+ for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
+ accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
+ for (i = 0; i < c->start[channel]; k++, j++, i++)
+ accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
+
+ resp = 0;
+ /* TODO: implement FFT instead of this naive calculation */
+ for (band = 0; band < DCA_SUBBANDS_32; band++) {
+ for (j = 0; j < 32; j++)
+ resp += mul32(accum[j], band_delta_factor(band, j));
+
+ out[band] = (band & 2) ? (-resp) : resp;
+ }
+}
+
+static int32_t lfe_fir_64i[512];
+static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
+{
+ int i, j;
+ int channel = c->prim_channels;
+ int32_t accum = 0;
+
+ add_new_samples(c, in, LFE_INTERPOLATION, channel);
+ for (i = c->start[channel], j = 0; i < 512; i++, j++)
+ accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
+ for (i = 0; i < c->start[channel]; i++, j++)
+ accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
+ return accum;
+}
+
+static void init_lfe_fir(void)
+{
+ static int initialized = 0;
+ int i;
+ if (initialized)
+ return;
+
+ for (i = 0; i < 512; i++)
+ lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
+ initialized = 1;
+}
+
+static void put_frame_header(DCAContext *c)
+{
+ /* SYNC */
+ put_bits(&c->pb, 16, 0x7ffe);
+ put_bits(&c->pb, 16, 0x8001);
+
+ /* Frame type: normal */
+ put_bits(&c->pb, 1, 1);
+
+ /* Deficit sample count: none */
+ put_bits(&c->pb, 5, 31);
+
+ /* CRC is not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* Number of PCM sample blocks */
+ put_bits(&c->pb, 7, PCM_SAMPLES-1);
+
+ /* Primary frame byte size */
+ put_bits(&c->pb, 14, c->frame_size-1);
+
+ /* Audio channel arrangement: L + R (stereo) */
+ put_bits(&c->pb, 6, c->num_channel);
+
+ /* Core audio sampling frequency */
+ put_bits(&c->pb, 4, c->sample_rate_code);
+
+ /* Transmission bit rate: 1411.2 kbps */
+ put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
+
+ /* Embedded down mix: disabled */
+ put_bits(&c->pb, 1, 0);
+
+ /* Embedded dynamic range flag: not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* Embedded time stamp flag: not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* Auxiliary data flag: not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* HDCD source: no */
+ put_bits(&c->pb, 1, 0);
+
+ /* Extension audio ID: N/A */
+ put_bits(&c->pb, 3, 0);
+
+ /* Extended audio data: not present */
+ put_bits(&c->pb, 1, 0);
+
+ /* Audio sync word insertion flag: after each sub-frame */
+ put_bits(&c->pb, 1, 0);
+
+ /* Low frequency effects flag: not present or interpolation factor=64 */
+ put_bits(&c->pb, 2, c->lfe_state);
+
+ /* Predictor history switch flag: on */
+ put_bits(&c->pb, 1, 1);
+
+ /* No CRC */
+ /* Multirate interpolator switch: non-perfect reconstruction */
+ put_bits(&c->pb, 1, 0);
+
+ /* Encoder software revision: 7 */
+ put_bits(&c->pb, 4, 7);
+
+ /* Copy history: 0 */
+ put_bits(&c->pb, 2, 0);
+
+ /* Source PCM resolution: 16 bits, not DTS ES */
+ put_bits(&c->pb, 3, 0);
+
+ /* Front sum/difference coding: no */
+ put_bits(&c->pb, 1, 0);
+
+ /* Surrounds sum/difference coding: no */
+ put_bits(&c->pb, 1, 0);
+
+ /* Dialog normalization: 0 dB */
+ put_bits(&c->pb, 4, 0);
+}
+
+static void put_primary_audio_header(DCAContext *c)
+{
+ 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 };
+
+ int ch, i;
+ /* Number of subframes */
+ put_bits(&c->pb, 4, SUBFRAMES - 1);
+
+ /* Number of primary audio channels */
+ put_bits(&c->pb, 3, c->prim_channels - 1);
+
+ /* Subband activity count */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
+
+ /* High frequency VQ start subband */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
+
+ /* Joint intensity coding index: 0, 0 */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ put_bits(&c->pb, 3, 0);
+
+ /* Transient mode codebook: A4, A4 (arbitrary) */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ put_bits(&c->pb, 2, 0);
+
+ /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ put_bits(&c->pb, 3, 6);
+
+ /* Bit allocation quantizer select: linear 5-bit */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ put_bits(&c->pb, 3, 6);
+
+ /* Quantization index codebook select: dummy data
+ to avoid transmission of scale factor adjustment */
+
+ for (i = 1; i < 11; i++)
+ for (ch = 0; ch < c->prim_channels; ch++)
+ put_bits(&c->pb, bitlen[i], thr[i]);
+
+ /* Scale factor adjustment index: not transmitted */
+}
+
+/**
+ * 8-23 bits quantization
+ * @param sample
+ * @param bits
+ */
+static inline uint32_t quantize(int32_t sample, int bits)
+{
+ av_assert0(sample < 1 << (bits - 1));
+ av_assert0(sample >= -(1 << (bits - 1)));
+ return sample & ((1 << bits) - 1);
+}
+
+static inline int find_scale_factor7(int64_t max_value, int bits)
+{
+ int i = 0, j = 128, q;
+ max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
+ while (i < j) {
+ q = (i + j) >> 1;
+ if (max_value < scale_factor_quant7[q])
+ j = q;
+ else
+ i = q + 1;
+ }
+ av_assert1(i < 128);
+ return i;
+}
+
+static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
+ int scale_factor)
+{
+ sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
+ put_bits(&c->pb, bits, quantize((int) sample, bits));
+}
+
+static void put_subframe(DCAContext *c,
+ int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
+ int subframe)
+{
+ int i, sub, ss, ch, max_value;
+ int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
+
+ /* Subsubframes count */
+ put_bits(&c->pb, 2, SUBSUBFRAMES -1);
+
+ /* Partial subsubframe sample count: dummy */
+ put_bits(&c->pb, 3, 0);
+
+ /* Prediction mode: no ADPCM, in each channel and subband */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ for (sub = 0; sub < DCA_SUBBANDS; sub++)
+ put_bits(&c->pb, 1, 0);
+
+ /* Prediction VQ addres: not transmitted */
+ /* Bit allocation index */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ for (sub = 0; sub < DCA_SUBBANDS; sub++)
+ put_bits(&c->pb, 5, QUANTIZER_BITS+3);
+
+ if (SUBSUBFRAMES > 1) {
+ /* Transition mode: none for each channel and subband */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ for (sub = 0; sub < DCA_SUBBANDS; sub++)
+ put_bits(&c->pb, 1, 0); /* codebook A4 */
+ }
+
+ /* Determine scale_factor */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ for (sub = 0; sub < DCA_SUBBANDS; sub++) {
+ max_value = 0;
+ for (i = 0; i < 8 * SUBSUBFRAMES; i++)
+ max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
+ c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
+ }
+
+ if (c->lfe_channel) {
+ max_value = 0;
+ for (i = 0; i < 4 * SUBSUBFRAMES; i++)
+ max_value = FFMAX(max_value, FFABS(lfe_data[i]));
+ c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
+ }
+
+ /* Scale factors: the same for each channel and subband,
+ encoded according to Table D.1.2 */
+ for (ch = 0; ch < c->prim_channels; ch++)
+ for (sub = 0; sub < DCA_SUBBANDS; sub++)
+ put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
+
+ /* Joint subband scale factor codebook select: not transmitted */
+ /* Scale factors for joint subband coding: not transmitted */
+ /* Stereo down-mix coefficients: not transmitted */
+ /* Dynamic range coefficient: not transmitted */
+ /* Stde information CRC check word: not transmitted */
+ /* VQ encoded high frequency subbands: not transmitted */
+
+ /* LFE data */
+ if (c->lfe_channel) {
+ for (i = 0; i < 4 * SUBSUBFRAMES; i++)
+ put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
+ put_bits(&c->pb, 8, c->lfe_scale_factor);
+ }
+
+ /* Audio data (subsubframes) */
+
+ for (ss = 0; ss < SUBSUBFRAMES ; ss++)
+ for (ch = 0; ch < c->prim_channels; ch++)
+ for (sub = 0; sub < DCA_SUBBANDS; sub++)
+ for (i = 0; i < 8; i++)
+ put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
+
+ /* DSYNC */
+ put_bits(&c->pb, 16, 0xffff);
+}
+
+static void put_frame(DCAContext *c,
+ int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
+ uint8_t *frame)
+{
+ int i;
+ init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
+
+ put_primary_audio_header(c);
+ for (i = 0; i < SUBFRAMES; i++)
+ put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
+
+ flush_put_bits(&c->pb);
+ c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
+
+ init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
+ put_frame_header(c);
+ flush_put_bits(&c->pb);
+}
+
+static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
+{
+ int i, k, channel;
+ DCAContext *c = avctx->priv_data;
+ const int16_t *samples;
+ int ret, real_channel = 0;
+
+ if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)))
+ return ret;
+
+ samples = (const int16_t *)frame->data[0];
+ for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
+ for (channel = 0; channel < c->prim_channels + 1; channel++) {
+ real_channel = c->channel_order_tab[channel];
+ if (real_channel >= 0) {
+ /* Get 32 PCM samples */
+ for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
+ c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
+ }
+ /* Put subband samples into the proper place */
+ qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
+ }
+ }
+ }
+
+ if (c->lfe_channel) {
+ for (i = 0; i < PCM_SAMPLES / 2; i++) {
+ for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
+ c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
+ c->lfe_data[i] = lfe_downsample(c, c->pcm);
+ }
+ }
+
+ put_frame(c, c->subband, avpkt->data);
+
+ avpkt->size = c->frame_size;
+ *got_packet_ptr = 1;
+ return 0;
+}
+
+static int encode_init(AVCodecContext *avctx)
+{
+ DCAContext *c = avctx->priv_data;
+ int i;
+ uint64_t layout = avctx->channel_layout;
+
+ c->prim_channels = avctx->channels;
+ c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
+
+ if (!layout) {
+ av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
+ "encoder will guess the layout, but it "
+ "might be incorrect.\n");
+ layout = av_get_default_channel_layout(avctx->channels);
+ }
+ switch (layout) {
+ case AV_CH_LAYOUT_STEREO: c->a_mode = 2; c->num_channel = 2; break;
+ case AV_CH_LAYOUT_5POINT0: c->a_mode = 9; c->num_channel = 9; break;
+ case AV_CH_LAYOUT_5POINT1: c->a_mode = 9; c->num_channel = 9; break;
+ case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
+ case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
+ default:
+ av_log(avctx, AV_LOG_ERROR,
+ "Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ if (c->lfe_channel) {
+ init_lfe_fir();
+ c->prim_channels--;
+ c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
+ c->lfe_state = LFE_PRESENT;
+ c->lfe_offset = dca_lfe_index[c->a_mode];
+ } else {
+ c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
+ c->lfe_state = LFE_MISSING;
+ }
+
+ for (i = 0; i < 16; i++) {
- av_log(avctx, AV_LOG_ERROR, "%d, ", dca_sample_rates[i]);
++ if (avpriv_dca_sample_rates[i] && (avpriv_dca_sample_rates[i] == avctx->sample_rate))
+ break;
+ }
+ if (i == 16) {
+ av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
+ for (i = 0; i < 16; i++)
++ av_log(avctx, AV_LOG_ERROR, "%d, ", avpriv_dca_sample_rates[i]);
+ av_log(avctx, AV_LOG_ERROR, "supported.\n");
+ return -1;
+ }
+ c->sample_rate_code = i;
+
+ avctx->frame_size = 32 * PCM_SAMPLES;
+
+ if (!cos_table[127])
+ qmf_init();
+ return 0;
+}
+
+AVCodec ff_dca_encoder = {
+ .name = "dca",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_DTS,
+ .priv_data_size = sizeof(DCAContext),
+ .init = encode_init,
+ .encode2 = encode_frame,
+ .capabilities = CODEC_CAP_EXPERIMENTAL,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
+};
projects / ffmpeg.git / commitdiff