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authorDiego Biurrun <diego@biurrun.de>2012-07-31 20:00:35 +0200
committerDiego Biurrun <diego@biurrun.de>2012-08-01 00:17:16 +0200
commit13a79cf84e073d0ca8489047660352eee216d059 (patch)
tree61df12e134a17060d1f6275204ec0d5130a067a8 /libavcodec/dcadec.c
parent6376a3ad24cb6a3c8ccaaa87e82846931d48045f (diff)
dca: Rename dca.c ---> dcadec.c
This will allow adding dca.c with tables used from other files.
Diffstat (limited to 'libavcodec/dcadec.c')
-rw-r--r--libavcodec/dcadec.c1971
1 files changed, 1971 insertions, 0 deletions
diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
new file mode 100644
index 0000000000..b37dc49d3f
--- /dev/null
+++ b/libavcodec/dcadec.c
@@ -0,0 +1,1971 @@
+/*
+ * DCA compatible decoder
+ * Copyright (C) 2004 Gildas Bazin
+ * Copyright (C) 2004 Benjamin Zores
+ * Copyright (C) 2006 Benjamin Larsson
+ * Copyright (C) 2007 Konstantin Shishkov
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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)
+#define DCA_SUBBANDS (32)
+#define DCA_ABITS_MAX (32) /* Should be 28 */
+#define DCA_SUBSUBFRAMES_MAX (4)
+#define DCA_SUBFRAMES_MAX (16)
+#define DCA_BLOCKS_MAX (16)
+#define DCA_LFE_MAX (3)
+
+enum DCAMode {
+ DCA_MONO = 0,
+ DCA_CHANNEL,
+ DCA_STEREO,
+ DCA_STEREO_SUMDIFF,
+ DCA_STEREO_TOTAL,
+ DCA_3F,
+ DCA_2F1R,
+ DCA_3F1R,
+ DCA_2F2R,
+ DCA_3F2R,
+ DCA_4F2R
+};
+
+/* these are unconfirmed but should be mostly correct */
+enum DCAExSSSpeakerMask {
+ DCA_EXSS_FRONT_CENTER = 0x0001,
+ DCA_EXSS_FRONT_LEFT_RIGHT = 0x0002,
+ DCA_EXSS_SIDE_REAR_LEFT_RIGHT = 0x0004,
+ DCA_EXSS_LFE = 0x0008,
+ DCA_EXSS_REAR_CENTER = 0x0010,
+ DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,
+ DCA_EXSS_REAR_LEFT_RIGHT = 0x0040,
+ DCA_EXSS_FRONT_HIGH_CENTER = 0x0080,
+ DCA_EXSS_OVERHEAD = 0x0100,
+ DCA_EXSS_CENTER_LEFT_RIGHT = 0x0200,
+ DCA_EXSS_WIDE_LEFT_RIGHT = 0x0400,
+ DCA_EXSS_SIDE_LEFT_RIGHT = 0x0800,
+ DCA_EXSS_LFE2 = 0x1000,
+ DCA_EXSS_SIDE_HIGH_LEFT_RIGHT = 0x2000,
+ DCA_EXSS_REAR_HIGH_CENTER = 0x4000,
+ DCA_EXSS_REAR_HIGH_LEFT_RIGHT = 0x8000,
+};
+
+enum DCAExtensionMask {
+ DCA_EXT_CORE = 0x001, ///< core in core substream
+ DCA_EXT_XXCH = 0x002, ///< XXCh channels extension in core substream
+ DCA_EXT_X96 = 0x004, ///< 96/24 extension in core substream
+ DCA_EXT_XCH = 0x008, ///< XCh channel extension in core substream
+ DCA_EXT_EXSS_CORE = 0x010, ///< core in ExSS (extension substream)
+ DCA_EXT_EXSS_XBR = 0x020, ///< extended bitrate extension in ExSS
+ DCA_EXT_EXSS_XXCH = 0x040, ///< XXCh channels extension in ExSS
+ DCA_EXT_EXSS_X96 = 0x080, ///< 96/24 extension in ExSS
+ DCA_EXT_EXSS_LBR = 0x100, ///< low bitrate component in ExSS
+ DCA_EXT_EXSS_XLL = 0x200, ///< lossless extension in ExSS
+};
+
+/* -1 are reserved or unknown */
+static const int dca_ext_audio_descr_mask[] = {
+ DCA_EXT_XCH,
+ -1,
+ DCA_EXT_X96,
+ DCA_EXT_XCH | DCA_EXT_X96,
+ -1,
+ -1,
+ DCA_EXT_XXCH,
+ -1,
+};
+
+/* extensions that reside in core substream */
+#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)
+
+/* Tables for mapping dts channel configurations to libavcodec multichannel api.
+ * Some compromises have been made for special configurations. Most configurations
+ * are never used so complete accuracy is not needed.
+ *
+ * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
+ * S -> side, when both rear and back are configured move one of them to the side channel
+ * OV -> center back
+ * All 2 channel configurations -> AV_CH_LAYOUT_STEREO
+ */
+static const uint64_t dca_core_channel_layout[] = {
+ AV_CH_FRONT_CENTER, ///< 1, A
+ AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
+ AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
+ AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference)
+ AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total)
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R
+ AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
+ AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
+ AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV
+
+ AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER |
+ AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
+ AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
+ AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
+ AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
+ AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
+ AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR
+};
+
+static const int8_t dca_lfe_index[] = {
+ 1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3
+};
+
+static const int8_t dca_channel_reorder_lfe[][9] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, -1, -1, -1, -1, -1},
+ { 0, 1, 3, 4, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, -1, -1, -1, -1},
+ { 3, 4, 0, 1, 5, 6, -1, -1, -1},
+ { 2, 0, 1, 4, 5, 6, -1, -1, -1},
+ { 0, 6, 4, 5, 2, 3, -1, -1, -1},
+ { 4, 2, 5, 0, 1, 6, 7, -1, -1},
+ { 5, 6, 0, 1, 7, 3, 8, 4, -1},
+ { 4, 2, 5, 0, 1, 6, 8, 7, -1},
+};
+
+static const int8_t dca_channel_reorder_lfe_xch[][9] = {
+ { 0, 2, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, -1, -1, -1, -1, -1},
+ { 0, 1, 3, 4, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, -1, -1, -1, -1},
+ { 0, 1, 4, 5, 3, -1, -1, -1, -1},
+ { 2, 0, 1, 5, 6, 4, -1, -1, -1},
+ { 3, 4, 0, 1, 6, 7, 5, -1, -1},
+ { 2, 0, 1, 4, 5, 6, 7, -1, -1},
+ { 0, 6, 4, 5, 2, 3, 7, -1, -1},
+ { 4, 2, 5, 0, 1, 7, 8, 6, -1},
+ { 5, 6, 0, 1, 8, 3, 9, 4, 7},
+ { 4, 2, 5, 0, 1, 6, 9, 8, 7},
+};
+
+static const int8_t dca_channel_reorder_nolfe[][9] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, -1, -1, -1, -1, -1},
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, 4, -1, -1, -1, -1},
+ { 2, 3, 0, 1, 4, 5, -1, -1, -1},
+ { 2, 0, 1, 3, 4, 5, -1, -1, -1},
+ { 0, 5, 3, 4, 1, 2, -1, -1, -1},
+ { 3, 2, 4, 0, 1, 5, 6, -1, -1},
+ { 4, 5, 0, 1, 6, 2, 7, 3, -1},
+ { 3, 2, 4, 0, 1, 5, 7, 6, -1},
+};
+
+static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, -1, -1, -1, -1, -1},
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, 4, -1, -1, -1, -1},
+ { 0, 1, 3, 4, 2, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, 3, -1, -1, -1},
+ { 2, 3, 0, 1, 5, 6, 4, -1, -1},
+ { 2, 0, 1, 3, 4, 5, 6, -1, -1},
+ { 0, 5, 3, 4, 1, 2, 6, -1, -1},
+ { 3, 2, 4, 0, 1, 6, 7, 5, -1},
+ { 4, 5, 0, 1, 7, 2, 8, 3, 6},
+ { 3, 2, 4, 0, 1, 5, 8, 7, 6},
+};
+
+#define DCA_DOLBY 101 /* FIXME */
+
+#define DCA_CHANNEL_BITS 6
+#define DCA_CHANNEL_MASK 0x3F
+
+#define DCA_LFE 0x80
+
+#define HEADER_SIZE 14
+
+#define DCA_MAX_FRAME_SIZE 16384
+#define DCA_MAX_EXSS_HEADER_SIZE 4096
+
+#define DCA_BUFFER_PADDING_SIZE 1024
+
+/** Bit allocation */
+typedef struct {
+ int offset; ///< code values offset
+ int maxbits[8]; ///< max bits in VLC
+ int wrap; ///< wrap for get_vlc2()
+ VLC vlc[8]; ///< actual codes
+} BitAlloc;
+
+static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
+static BitAlloc dca_tmode; ///< transition mode VLCs
+static BitAlloc dca_scalefactor; ///< scalefactor VLCs
+static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
+
+static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
+ int idx)
+{
+ return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
+ ba->offset;
+}
+
+typedef struct {
+ AVCodecContext *avctx;
+ AVFrame frame;
+ /* Frame header */
+ int frame_type; ///< type of the current frame
+ int samples_deficit; ///< deficit sample count
+ int crc_present; ///< crc is present in the bitstream
+ int sample_blocks; ///< number of PCM sample blocks
+ int frame_size; ///< primary frame byte size
+ int amode; ///< audio channels arrangement
+ int sample_rate; ///< audio sampling rate
+ int bit_rate; ///< transmission bit rate
+ int bit_rate_index; ///< transmission bit rate index
+
+ int downmix; ///< embedded downmix enabled
+ int dynrange; ///< embedded dynamic range flag
+ int timestamp; ///< embedded time stamp flag
+ int aux_data; ///< auxiliary data flag
+ int hdcd; ///< source material is mastered in HDCD
+ int ext_descr; ///< extension audio descriptor flag
+ int ext_coding; ///< extended coding flag
+ int aspf; ///< audio sync word insertion flag
+ int lfe; ///< low frequency effects flag
+ int predictor_history; ///< predictor history flag
+ int header_crc; ///< header crc check bytes
+ int multirate_inter; ///< multirate interpolator switch
+ int version; ///< encoder software revision
+ int copy_history; ///< copy history
+ int source_pcm_res; ///< source pcm resolution
+ int front_sum; ///< front sum/difference flag
+ int surround_sum; ///< surround sum/difference flag
+ int dialog_norm; ///< dialog normalisation parameter
+
+ /* Primary audio coding header */
+ int subframes; ///< number of subframes
+ int is_channels_set; ///< check for if the channel number is already set
+ int total_channels; ///< number of channels including extensions
+ int prim_channels; ///< number of primary audio channels
+ int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
+ int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
+ int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
+ int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
+ int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
+ int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
+ int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
+ float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
+
+ /* Primary audio coding side information */
+ int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
+ int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
+ int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
+ int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
+ int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
+ int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
+ int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
+ int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
+ int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
+ int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
+ int dynrange_coef; ///< dynamic range coefficient
+
+ int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
+
+ float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
+ int lfe_scale_factor;
+
+ /* Subband samples history (for ADPCM) */
+ DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
+ DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
+ DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
+ int hist_index[DCA_PRIM_CHANNELS_MAX];
+ DECLARE_ALIGNED(32, float, raXin)[32];
+
+ int output; ///< type of output
+ float scale_bias; ///< output scale
+
+ DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX + 1) * 256];
+ const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
+
+ uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
+ int dca_buffer_size; ///< how much data is in the dca_buffer
+
+ const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
+ GetBitContext gb;
+ /* Current position in DCA frame */
+ int current_subframe;
+ int current_subsubframe;
+
+ int core_ext_mask; ///< present extensions in the core substream
+
+ /* XCh extension information */
+ int xch_present; ///< XCh extension present and valid
+ int xch_base_channel; ///< index of first (only) channel containing XCH data
+
+ /* ExSS header parser */
+ int static_fields; ///< static fields present
+ int mix_metadata; ///< mixing metadata present
+ int num_mix_configs; ///< number of mix out configurations
+ int mix_config_num_ch[4]; ///< number of channels in each mix out configuration
+
+ int profile;
+
+ int debug_flag; ///< used for suppressing repeated error messages output
+ 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);
+}
+
+static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
+{
+ int i, j;
+ static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
+ static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
+ static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
+
+ s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
+ s->prim_channels = s->total_channels;
+
+ if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
+ s->prim_channels = DCA_PRIM_CHANNELS_MAX;
+
+
+ for (i = base_channel; i < s->prim_channels; i++) {
+ s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
+ if (s->subband_activity[i] > DCA_SUBBANDS)
+ s->subband_activity[i] = DCA_SUBBANDS;
+ }
+ for (i = base_channel; i < s->prim_channels; i++) {
+ s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
+ if (s->vq_start_subband[i] > DCA_SUBBANDS)
+ s->vq_start_subband[i] = DCA_SUBBANDS;
+ }
+ get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3);
+ get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2);
+ get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
+ get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3);
+
+ /* Get codebooks quantization indexes */
+ if (!base_channel)
+ memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
+ for (j = 1; j < 11; j++)
+ for (i = base_channel; i < s->prim_channels; i++)
+ s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
+
+ /* Get scale factor adjustment */
+ for (j = 0; j < 11; j++)
+ for (i = base_channel; i < s->prim_channels; i++)
+ s->scalefactor_adj[i][j] = 1;
+
+ for (j = 1; j < 11; j++)
+ for (i = base_channel; i < s->prim_channels; i++)
+ if (s->quant_index_huffman[i][j] < thr[j])
+ s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
+
+ if (s->crc_present) {
+ /* Audio header CRC check */
+ get_bits(&s->gb, 16);
+ }
+
+ s->current_subframe = 0;
+ s->current_subsubframe = 0;
+
+#ifdef TRACE
+ av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
+ av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
+ for (i = base_channel; i < s->prim_channels; i++) {
+ av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n",
+ s->subband_activity[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n",
+ s->vq_start_subband[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n",
+ s->joint_intensity[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n",
+ s->transient_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n",
+ s->scalefactor_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n",
+ s->bitalloc_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
+ for (j = 0; j < 11; j++)
+ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->quant_index_huffman[i][j]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
+ for (j = 0; j < 11; j++)
+ av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+#endif
+
+ return 0;
+}
+
+static int dca_parse_frame_header(DCAContext *s)
+{
+ init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
+
+ /* Sync code */
+ skip_bits_long(&s->gb, 32);
+
+ /* Frame header */
+ s->frame_type = get_bits(&s->gb, 1);
+ s->samples_deficit = get_bits(&s->gb, 5) + 1;
+ s->crc_present = get_bits(&s->gb, 1);
+ s->sample_blocks = get_bits(&s->gb, 7) + 1;
+ s->frame_size = get_bits(&s->gb, 14) + 1;
+ if (s->frame_size < 95)
+ return AVERROR_INVALIDDATA;
+ s->amode = get_bits(&s->gb, 6);
+ s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
+ if (!s->sample_rate)
+ return AVERROR_INVALIDDATA;
+ s->bit_rate_index = get_bits(&s->gb, 5);
+ s->bit_rate = dca_bit_rates[s->bit_rate_index];
+ if (!s->bit_rate)
+ return AVERROR_INVALIDDATA;
+
+ s->downmix = get_bits(&s->gb, 1);
+ s->dynrange = get_bits(&s->gb, 1);
+ s->timestamp = get_bits(&s->gb, 1);
+ s->aux_data = get_bits(&s->gb, 1);
+ s->hdcd = get_bits(&s->gb, 1);
+ s->ext_descr = get_bits(&s->gb, 3);
+ s->ext_coding = get_bits(&s->gb, 1);
+ s->aspf = get_bits(&s->gb, 1);
+ s->lfe = get_bits(&s->gb, 2);
+ s->predictor_history = get_bits(&s->gb, 1);
+
+ /* TODO: check CRC */
+ if (s->crc_present)
+ s->header_crc = get_bits(&s->gb, 16);
+
+ s->multirate_inter = get_bits(&s->gb, 1);
+ s->version = get_bits(&s->gb, 4);
+ s->copy_history = get_bits(&s->gb, 2);
+ s->source_pcm_res = get_bits(&s->gb, 3);
+ s->front_sum = get_bits(&s->gb, 1);
+ s->surround_sum = get_bits(&s->gb, 1);
+ s->dialog_norm = get_bits(&s->gb, 4);
+
+ /* FIXME: channels mixing levels */
+ s->output = s->amode;
+ if (s->lfe)
+ s->output |= DCA_LFE;
+
+#ifdef TRACE
+ av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
+ av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
+ av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
+ av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
+ s->sample_blocks, s->sample_blocks * 32);
+ av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
+ av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
+ s->amode, dca_channels[s->amode]);
+ av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n",
+ s->sample_rate);
+ av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",
+ s->bit_rate);
+ av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
+ av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
+ av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
+ av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
+ av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
+ av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
+ av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
+ av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
+ av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
+ av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
+ s->predictor_history);
+ av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
+ av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
+ s->multirate_inter);
+ av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
+ av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
+ av_log(s->avctx, AV_LOG_DEBUG,
+ "source pcm resolution: %i (%i bits/sample)\n",
+ s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
+ av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
+ av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
+ av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+#endif
+
+ /* Primary audio coding header */
+ s->subframes = get_bits(&s->gb, 4) + 1;
+
+ return dca_parse_audio_coding_header(s, 0);
+}
+
+
+static inline int get_scale(GetBitContext *gb, int level, int value, 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;
+ }
+ for (j = base_channel; j < s->prim_channels; j++) {
+ s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
+ s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
+ }
+ }
+ }
+
+ /* Dynamic range coefficient */
+ if (!base_channel && s->dynrange)
+ s->dynrange_coef = get_bits(&s->gb, 8);
+
+ /* Side information CRC check word */
+ if (s->crc_present) {
+ get_bits(&s->gb, 16);
+ }
+
+ /*
+ * Primary audio data arrays
+ */
+
+ /* VQ encoded high frequency subbands */
+ for (j = base_channel; j < s->prim_channels; j++)
+ for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
+ /* 1 vector -> 32 samples */
+ s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
+
+ /* Low frequency effect data */
+ if (!base_channel && s->lfe) {
+ /* LFE samples */
+ int lfe_samples = 2 * s->lfe * (4 + block_index);
+ int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
+ float lfe_scale;
+
+ for (j = lfe_samples; j < lfe_end_sample; j++) {
+ /* Signed 8 bits int */
+ s->lfe_data[j] = get_sbits(&s->gb, 8);
+ }
+
+ /* Scale factor index */
+ skip_bits(&s->gb, 1);
+ s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 7)];
+
+ /* Quantization step size * scale factor */
+ lfe_scale = 0.035 * s->lfe_scale_factor;
+
+ for (j = lfe_samples; j < lfe_end_sample; j++)
+ s->lfe_data[j] *= lfe_scale;
+ }
+
+#ifdef TRACE
+ av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n",
+ s->subsubframes[s->current_subframe]);
+ av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
+ s->partial_samples[s->current_subframe]);
+
+ for (j = base_channel; j < s->prim_channels; j++) {
+ av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
+ for (k = 0; k < s->subband_activity[j]; k++)
+ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+ for (j = base_channel; j < s->prim_channels; j++) {
+ for (k = 0; k < s->subband_activity[j]; k++)
+ av_log(s->avctx, AV_LOG_DEBUG,
+ "prediction coefs: %f, %f, %f, %f\n",
+ (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
+ }
+ for (j = base_channel; j < s->prim_channels; j++) {
+ av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
+ for (k = 0; k < s->vq_start_subband[j]; k++)
+ av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+ for (j = base_channel; j < s->prim_channels; j++) {
+ av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
+ for (k = 0; k < s->subband_activity[j]; k++)
+ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+ for (j = base_channel; j < s->prim_channels; j++) {
+ av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
+ for (k = 0; k < s->subband_activity[j]; k++) {
+ if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
+ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
+ if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
+ av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
+ }
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+ for (j = base_channel; j < s->prim_channels; j++) {
+ if (s->joint_intensity[j] > 0) {
+ int source_channel = s->joint_intensity[j] - 1;
+ av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
+ for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
+ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+ }
+ if (!base_channel && s->prim_channels > 2 && s->downmix) {
+ av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
+ for (j = 0; j < s->prim_channels; j++) {
+ av_log(s->avctx, AV_LOG_DEBUG, "Channel 0, %d = %f\n", j,
+ dca_downmix_coeffs[s->downmix_coef[j][0]]);
+ av_log(s->avctx, AV_LOG_DEBUG, "Channel 1, %d = %f\n", j,
+ dca_downmix_coeffs[s->downmix_coef[j][1]]);
+ }
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+ for (j = base_channel; j < s->prim_channels; j++)
+ for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
+ av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
+ if (!base_channel && s->lfe) {
+ int lfe_samples = 2 * s->lfe * (4 + block_index);
+ int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
+
+ av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
+ for (j = lfe_samples; j < lfe_end_sample; j++)
+ av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
+ av_log(s->avctx, AV_LOG_DEBUG, "\n");
+ }
+#endif
+
+ return 0;
+}
+
+static void qmf_32_subbands(DCAContext *s, int chans,
+ float samples_in[32][8], float *samples_out,
+ float scale)
+{
+ const float *prCoeff;
+ int i;
+
+ int sb_act = s->subband_activity[chans];
+ int subindex;
+
+ scale *= sqrt(1 / 8.0);
+
+ /* Select filter */
+ if (!s->multirate_inter) /* Non-perfect reconstruction */
+ prCoeff = fir_32bands_nonperfect;
+ else /* Perfect reconstruction */
+ prCoeff = fir_32bands_perfect;
+
+ for (i = sb_act; i < 32; i++)
+ s->raXin[i] = 0.0;
+
+ /* Reconstructed channel sample index */
+ for (subindex = 0; subindex < 8; subindex++) {
+ /* Load in one sample from each subband and clear inactive subbands */
+ for (i = 0; i < sb_act; i++) {
+ unsigned sign = (i - 1) & 2;
+ uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
+ AV_WN32A(&s->raXin[i], v);
+ }
+
+ s->synth.synth_filter_float(&s->imdct,
+ s->subband_fir_hist[chans],
+ &s->hist_index[chans],
+ s->subband_fir_noidea[chans], prCoeff,
+ samples_out, s->raXin, scale);
+ samples_out += 32;
+ }
+}
+
+static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
+ int num_deci_sample, float *samples_in,
+ float *samples_out, float scale)
+{
+ /* samples_in: An array holding decimated samples.
+ * Samples in current subframe starts from samples_in[0],
+ * while samples_in[-1], samples_in[-2], ..., stores samples
+ * from last subframe as history.
+ *
+ * samples_out: An array holding interpolated samples
+ */
+
+ int decifactor;
+ const float *prCoeff;
+ int deciindex;
+
+ /* Select decimation filter */
+ if (decimation_select == 1) {
+ decifactor = 64;
+ prCoeff = lfe_fir_128;
+ } else {
+ decifactor = 32;
+ prCoeff = lfe_fir_64;
+ }
+ /* Interpolation */
+ for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
+ s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale);
+ samples_in++;
+ samples_out += 2 * decifactor;
+ }
+}
+
+/* downmixing routines */
+#define MIX_REAR1(samples, si1, rs, coef) \
+ samples[i] += samples[si1] * coef[rs][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1];
+
+#define MIX_REAR2(samples, si1, si2, rs, coef) \
+ samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs + 1][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs + 1][1];
+
+#define MIX_FRONT3(samples, coef) \
+ t = samples[i + c]; \
+ u = samples[i + l]; \
+ v = samples[i + r]; \
+ samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
+ samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
+
+#define DOWNMIX_TO_STEREO(op1, op2) \
+ for (i = 0; i < 256; i++) { \
+ op1 \
+ op2 \
+ }
+
+static void dca_downmix(float *samples, int srcfmt,
+ int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
+ const int8_t *channel_mapping)
+{
+ int c, l, r, sl, sr, s;
+ int i;
+ float t, u, v;
+ float coef[DCA_PRIM_CHANNELS_MAX][2];
+
+ for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {
+ coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
+ coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
+ }
+
+ switch (srcfmt) {
+ case DCA_MONO:
+ case DCA_CHANNEL:
+ case DCA_STEREO_TOTAL:
+ case DCA_STEREO_SUMDIFF:
+ case DCA_4F2R:
+ av_log(NULL, 0, "Not implemented!\n");
+ break;
+ case DCA_STEREO:
+ break;
+ case DCA_3F:
+ c = channel_mapping[0] * 256;
+ l = channel_mapping[1] * 256;
+ r = channel_mapping[2] * 256;
+ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
+ break;
+ case DCA_2F1R:
+ s = channel_mapping[2] * 256;
+ DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );
+ break;
+ case DCA_3F1R:
+ c = channel_mapping[0] * 256;
+ l = channel_mapping[1] * 256;
+ r = channel_mapping[2] * 256;
+ s = channel_mapping[3] * 256;
+ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
+ MIX_REAR1(samples, i + s, 3, coef));
+ break;
+ case DCA_2F2R:
+ sl = channel_mapping[2] * 256;
+ sr = channel_mapping[3] * 256;
+ DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );
+ break;
+ case DCA_3F2R:
+ c = channel_mapping[0] * 256;
+ l = channel_mapping[1] * 256;
+ r = channel_mapping[2] * 256;
+ sl = channel_mapping[3] * 256;
+ sr = channel_mapping[4] * 256;
+ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
+ MIX_REAR2(samples, i + sl, i + sr, 3, coef));
+ break;
+ }
+}
+
+
+#ifndef decode_blockcodes
+/* Very compact version of the block code decoder that does not use table
+ * look-up but is slightly slower */
+static int decode_blockcode(int code, int levels, int *values)
+{
+ int i;
+ int offset = (levels - 1) >> 1;
+
+ for (i = 0; i < 4; i++) {
+ int div = FASTDIV(code, levels);
+ values[i] = code - offset - div * levels;
+ code = div;
+ }
+
+ return code;
+}
+
+static int decode_blockcodes(int code1, int code2, int levels, int *values)
+{
+ return decode_blockcode(code1, levels, values) |
+ decode_blockcode(code2, levels, values + 4);
+}
+#endif
+
+static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
+static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
+
+#ifndef int8x8_fmul_int32
+static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
+{
+ float fscale = scale / 16.0;
+ int i;
+ for (i = 0; i < 8; i++)
+ dst[i] = src[i] * fscale;
+}
+#endif
+
+static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
+{
+ int k, l;
+ int subsubframe = s->current_subsubframe;
+
+ const float *quant_step_table;
+
+ /* FIXME */
+ float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
+ LOCAL_ALIGNED_16(int, block, [8]);
+
+ /*
+ * Audio data
+ */
+
+ /* Select quantization step size table */
+ if (s->bit_rate_index == 0x1f)
+ quant_step_table = lossless_quant_d;
+ else
+ quant_step_table = lossy_quant_d;
+
+ for (k = base_channel; k < s->prim_channels; k++) {
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ for (l = 0; l < s->vq_start_subband[k]; l++) {
+ int m;
+
+ /* Select the mid-tread linear quantizer */
+ int abits = s->bitalloc[k][l];
+
+ float quant_step_size = quant_step_table[abits];
+
+ /*
+ * Determine quantization index code book and its type
+ */
+
+ /* Select quantization index code book */
+ int sel = s->quant_index_huffman[k][abits];
+
+ /*
+ * Extract bits from the bit stream
+ */
+ if (!abits) {
+ memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
+ } else {
+ /* Deal with transients */
+ int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
+ float rscale = quant_step_size * s->scale_factor[k][l][sfi] *
+ s->scalefactor_adj[k][sel];
+
+ if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
+ if (abits <= 7) {
+ /* Block code */
+ int block_code1, block_code2, size, levels, err;
+
+ size = abits_sizes[abits - 1];
+ levels = abits_levels[abits - 1];
+
+ block_code1 = get_bits(&s->gb, size);
+ block_code2 = get_bits(&s->gb, size);
+ err = decode_blockcodes(block_code1, block_code2,
+ levels, block);
+ if (err) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "ERROR: block code look-up failed\n");
+ return AVERROR_INVALIDDATA;
+ }
+ } else {
+ /* no coding */
+ for (m = 0; m < 8; m++)
+ block[m] = get_sbits(&s->gb, abits - 3);
+ }
+ } else {
+ /* Huffman coded */
+ for (m = 0; m < 8; m++)
+ block[m] = get_bitalloc(&s->gb,
+ &dca_smpl_bitalloc[abits], sel);
+ }
+
+ s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
+ block, rscale, 8);
+ }
+
+ /*
+ * Inverse ADPCM if in prediction mode
+ */
+ if (s->prediction_mode[k][l]) {
+ int n;
+ for (m = 0; m < 8; m++) {
+ for (n = 1; n <= 4; n++)
+ if (m >= n)
+ subband_samples[k][l][m] +=
+ (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
+ subband_samples[k][l][m - n] / 8192);
+ else if (s->predictor_history)
+ subband_samples[k][l][m] +=
+ (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
+ s->subband_samples_hist[k][l][m - n + 4] / 8192);
+ }
+ }
+ }
+
+ /*
+ * Decode VQ encoded high frequencies
+ */
+ for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
+ /* 1 vector -> 32 samples but we only need the 8 samples
+ * for this subsubframe. */
+ int hfvq = s->high_freq_vq[k][l];
+
+ if (!s->debug_flag & 0x01) {
+ av_log(s->avctx, AV_LOG_DEBUG,
+ "Stream with high frequencies VQ coding\n");
+ s->debug_flag |= 0x01;
+ }
+
+ int8x8_fmul_int32(subband_samples[k][l],
+ &high_freq_vq[hfvq][subsubframe * 8],
+ s->scale_factor[k][l][0]);
+ }
+ }
+
+ /* Check for DSYNC after subsubframe */
+ if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
+ if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
+#ifdef TRACE
+ av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
+#endif
+ } else {
+ av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
+ }
+ }
+
+ /* Backup predictor history for adpcm */
+ for (k = base_channel; k < s->prim_channels; k++)
+ for (l = 0; l < s->vq_start_subband[k]; l++)
+ memcpy(s->subband_samples_hist[k][l],
+ &subband_samples[k][l][4],
+ 4 * sizeof(subband_samples[0][0][0]));
+
+ return 0;
+}
+
+static int dca_filter_channels(DCAContext *s, int block_index)
+{
+ float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
+ int k;
+
+ /* 32 subbands QMF */
+ for (k = 0; k < s->prim_channels; k++) {
+/* static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,
+ 0, 8388608.0, 8388608.0 };*/
+ qmf_32_subbands(s, k, subband_samples[k],
+ &s->samples[256 * s->channel_order_tab[k]],
+ M_SQRT1_2 * s->scale_bias /* pcm_to_double[s->source_pcm_res] */);
+ }
+
+ /* Down mixing */
+ if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
+ dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
+ }
+
+ /* Generate LFE samples for this subsubframe FIXME!!! */
+ if (s->output & DCA_LFE) {
+ lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
+ s->lfe_data + 2 * s->lfe * (block_index + 4),
+ &s->samples[256 * dca_lfe_index[s->amode]],
+ (1.0 / 256.0) * s->scale_bias);
+ /* Outputs 20bits pcm samples */
+ }
+
+ return 0;
+}
+
+
+static int dca_subframe_footer(DCAContext *s, int base_channel)
+{
+ int aux_data_count = 0, i;
+
+ /*
+ * Unpack optional information
+ */
+
+ /* presumably optional information only appears in the core? */
+ if (!base_channel) {
+ if (s->timestamp)
+ skip_bits_long(&s->gb, 32);
+
+ if (s->aux_data)
+ aux_data_count = get_bits(&s->gb, 6);
+
+ for (i = 0; i < aux_data_count; i++)
+ get_bits(&s->gb, 8);
+
+ if (s->crc_present && (s->downmix || s->dynrange))
+ get_bits(&s->gb, 16);
+ }
+
+ return 0;
+}
+
+/**
+ * Decode a dca frame block
+ *
+ * @param s pointer to the DCAContext
+ */
+
+static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
+{
+ int ret;
+
+ /* Sanity check */
+ if (s->current_subframe >= s->subframes) {
+ av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
+ s->current_subframe, s->subframes);
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (!s->current_subsubframe) {
+#ifdef TRACE
+ av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
+#endif
+ /* Read subframe header */
+ if ((ret = dca_subframe_header(s, base_channel, block_index)))
+ return ret;
+ }
+
+ /* Read subsubframe */
+#ifdef TRACE
+ av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
+#endif
+ if ((ret = dca_subsubframe(s, base_channel, block_index)))
+ return ret;
+
+ /* Update state */
+ s->current_subsubframe++;
+ if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
+ s->current_subsubframe = 0;
+ s->current_subframe++;
+ }
+ if (s->current_subframe >= s->subframes) {
+#ifdef TRACE
+ av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
+#endif
+ /* Read subframe footer */
+ if ((ret = dca_subframe_footer(s, base_channel)))
+ return ret;
+ }
+
+ return 0;
+}
+
+/**
+ * Return the number of channels in an ExSS speaker mask (HD)
+ */
+static int dca_exss_mask2count(int mask)
+{
+ /* count bits that mean speaker pairs twice */
+ return av_popcount(mask) +
+ av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT |
+ DCA_EXSS_FRONT_LEFT_RIGHT |
+ DCA_EXSS_FRONT_HIGH_LEFT_RIGHT |
+ DCA_EXSS_WIDE_LEFT_RIGHT |
+ DCA_EXSS_SIDE_LEFT_RIGHT |
+ DCA_EXSS_SIDE_HIGH_LEFT_RIGHT |
+ DCA_EXSS_SIDE_REAR_LEFT_RIGHT |
+ DCA_EXSS_REAR_LEFT_RIGHT |
+ DCA_EXSS_REAR_HIGH_LEFT_RIGHT));
+}
+
+/**
+ * Skip mixing coefficients of a single mix out configuration (HD)
+ */
+static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
+{
+ int i;
+
+ for (i = 0; i < channels; i++) {
+ int mix_map_mask = get_bits(gb, out_ch);
+ int num_coeffs = av_popcount(mix_map_mask);
+ skip_bits_long(gb, num_coeffs * 6);
+ }
+}
+
+/**
+ * Parse extension substream asset header (HD)
+ */
+static int dca_exss_parse_asset_header(DCAContext *s)
+{
+ int header_pos = get_bits_count(&s->gb);
+ int header_size;
+ int channels;
+ int embedded_stereo = 0;
+ int embedded_6ch = 0;
+ int drc_code_present;
+ int extensions_mask;
+ int i, j;
+
+ if (get_bits_left(&s->gb) < 16)
+ return -1;
+
+ /* We will parse just enough to get to the extensions bitmask with which
+ * we can set the profile value. */
+
+ header_size = get_bits(&s->gb, 9) + 1;
+ skip_bits(&s->gb, 3); // asset index
+
+ if (s->static_fields) {
+ if (get_bits1(&s->gb))
+ skip_bits(&s->gb, 4); // asset type descriptor
+ if (get_bits1(&s->gb))
+ skip_bits_long(&s->gb, 24); // language descriptor
+
+ if (get_bits1(&s->gb)) {
+ /* How can one fit 1024 bytes of text here if the maximum value
+ * for the asset header size field above was 512 bytes? */
+ int text_length = get_bits(&s->gb, 10) + 1;
+ if (get_bits_left(&s->gb) < text_length * 8)
+ return -1;
+ skip_bits_long(&s->gb, text_length * 8); // info text
+ }
+
+ skip_bits(&s->gb, 5); // bit resolution - 1
+ skip_bits(&s->gb, 4); // max sample rate code
+ channels = get_bits(&s->gb, 8) + 1;
+
+ if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers
+ int spkr_remap_sets;
+ int spkr_mask_size = 16;
+ int num_spkrs[7];
+
+ if (channels > 2)
+ embedded_stereo = get_bits1(&s->gb);
+ if (channels > 6)
+ embedded_6ch = get_bits1(&s->gb);
+
+ if (get_bits1(&s->gb)) {
+ spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
+ skip_bits(&s->gb, spkr_mask_size); // spkr activity mask
+ }
+
+ spkr_remap_sets = get_bits(&s->gb, 3);
+
+ for (i = 0; i < spkr_remap_sets; i++) {
+ /* std layout mask for each remap set */
+ num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));
+ }
+
+ for (i = 0; i < spkr_remap_sets; i++) {
+ int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;
+ if (get_bits_left(&s->gb) < 0)
+ return -1;
+
+ for (j = 0; j < num_spkrs[i]; j++) {
+ int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);
+ int num_dec_ch = av_popcount(remap_dec_ch_mask);
+ skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes
+ }
+ }
+
+ } else {
+ skip_bits(&s->gb, 3); // representation type
+ }
+ }
+
+ drc_code_present = get_bits1(&s->gb);
+ if (drc_code_present)
+ get_bits(&s->gb, 8); // drc code
+
+ if (get_bits1(&s->gb))
+ skip_bits(&s->gb, 5); // dialog normalization code
+
+ if (drc_code_present && embedded_stereo)
+ get_bits(&s->gb, 8); // drc stereo code
+
+ if (s->mix_metadata && get_bits1(&s->gb)) {
+ skip_bits(&s->gb, 1); // external mix
+ skip_bits(&s->gb, 6); // post mix gain code
+
+ if (get_bits(&s->gb, 2) != 3) // mixer drc code
+ skip_bits(&s->gb, 3); // drc limit
+ else
+ skip_bits(&s->gb, 8); // custom drc code
+
+ if (get_bits1(&s->gb)) // channel specific scaling
+ for (i = 0; i < s->num_mix_configs; i++)
+ skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes
+ else
+ skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes
+
+ for (i = 0; i < s->num_mix_configs; i++) {
+ if (get_bits_left(&s->gb) < 0)
+ return -1;
+ dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);
+ if (embedded_6ch)
+ dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);
+ if (embedded_stereo)
+ dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);
+ }
+ }
+
+ switch (get_bits(&s->gb, 2)) {
+ case 0: extensions_mask = get_bits(&s->gb, 12); break;
+ case 1: extensions_mask = DCA_EXT_EXSS_XLL; break;
+ case 2: extensions_mask = DCA_EXT_EXSS_LBR; break;
+ case 3: extensions_mask = 0; /* aux coding */ break;
+ }
+
+ /* not parsed further, we were only interested in the extensions mask */
+
+ if (get_bits_left(&s->gb) < 0)
+ return -1;
+
+ if (get_bits_count(&s->gb) - header_pos > header_size * 8) {
+ av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");
+ return -1;
+ }
+ skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));
+
+ if (extensions_mask & DCA_EXT_EXSS_XLL)
+ s->profile = FF_PROFILE_DTS_HD_MA;
+ else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |
+ DCA_EXT_EXSS_XXCH))
+ s->profile = FF_PROFILE_DTS_HD_HRA;
+
+ if (!(extensions_mask & DCA_EXT_CORE))
+ av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
+ if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)
+ av_log(s->avctx, AV_LOG_WARNING,
+ "DTS extensions detection mismatch (%d, %d)\n",
+ extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);
+
+ return 0;
+}
+
+/**
+ * Parse extension substream header (HD)
+ */
+static void dca_exss_parse_header(DCAContext *s)
+{
+ int ss_index;
+ int blownup;
+ int num_audiop = 1;
+ int num_assets = 1;
+ int active_ss_mask[8];
+ int i, j;
+
+ if (get_bits_left(&s->gb) < 52)
+ return;
+
+ skip_bits(&s->gb, 8); // user data
+ ss_index = get_bits(&s->gb, 2);
+
+ blownup = get_bits1(&s->gb);
+ skip_bits(&s->gb, 8 + 4 * blownup); // header_size
+ skip_bits(&s->gb, 16 + 4 * blownup); // hd_size
+
+ s->static_fields = get_bits1(&s->gb);
+ if (s->static_fields) {
+ skip_bits(&s->gb, 2); // reference clock code
+ skip_bits(&s->gb, 3); // frame duration code
+
+ if (get_bits1(&s->gb))
+ skip_bits_long(&s->gb, 36); // timestamp
+
+ /* a single stream can contain multiple audio assets that can be
+ * combined to form multiple audio presentations */
+
+ num_audiop = get_bits(&s->gb, 3) + 1;
+ if (num_audiop > 1) {
+ av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");
+ /* ignore such streams for now */
+ return;
+ }
+
+ num_assets = get_bits(&s->gb, 3) + 1;
+ if (num_assets > 1) {
+ av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");
+ /* ignore such streams for now */
+ return;
+ }
+
+ for (i = 0; i < num_audiop; i++)
+ active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);
+
+ for (i = 0; i < num_audiop; i++)
+ for (j = 0; j <= ss_index; j++)
+ if (active_ss_mask[i] & (1 << j))
+ skip_bits(&s->gb, 8); // active asset mask
+
+ s->mix_metadata = get_bits1(&s->gb);
+ if (s->mix_metadata) {
+ int mix_out_mask_size;
+
+ skip_bits(&s->gb, 2); // adjustment level
+ mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
+ s->num_mix_configs = get_bits(&s->gb, 2) + 1;
+
+ for (i = 0; i < s->num_mix_configs; i++) {
+ int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);
+ s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);
+ }
+ }
+ }
+
+ for (i = 0; i < num_assets; i++)
+ skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size
+
+ for (i = 0; i < num_assets; i++) {
+ if (dca_exss_parse_asset_header(s))
+ return;
+ }
+
+ /* not parsed further, we were only interested in the extensions mask
+ * from the asset header */
+}
+
+/**
+ * Main frame decoding function
+ * FIXME add arguments
+ */
+static int dca_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+
+ int lfe_samples;
+ int num_core_channels = 0;
+ int i, ret;
+ float *samples_flt;
+ int16_t *samples_s16;
+ DCAContext *s = avctx->priv_data;
+ int channels;
+ int core_ss_end;
+
+
+ s->xch_present = 0;
+
+ s->dca_buffer_size = 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 */
+ if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
+
+ /* if ext_descr was unknown, clear s->core_ext_mask so that the
+ * extensions scan can fill it up */
+ s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
+
+ /* extensions start at 32-bit boundaries into bitstream */
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+
+ while (core_ss_end - get_bits_count(&s->gb) >= 32) {
+ uint32_t bits = get_bits_long(&s->gb, 32);
+
+ switch (bits) {
+ case 0x5a5a5a5a: {
+ int ext_amode, xch_fsize;
+
+ s->xch_base_channel = s->prim_channels;
+
+ /* validate sync word using XCHFSIZE field */
+ xch_fsize = show_bits(&s->gb, 10);
+ if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
+ (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
+ continue;
+
+ /* skip length-to-end-of-frame field for the moment */
+ skip_bits(&s->gb, 10);
+
+ s->core_ext_mask |= DCA_EXT_XCH;
+
+ /* extension amode(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 */
+ dca_parse_audio_coding_header(s, s->xch_base_channel);
+
+ for (i = 0; i < (s->sample_blocks / 8); i++)
+ if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
+ continue;
+ }
+
+ s->xch_present = 1;
+ break;
+ }
+ case 0x47004a03:
+ /* XXCh: extended channels */
+ /* usually found either in core or HD part in DTS-HD HRA streams,
+ * but not in DTS-ES which contains XCh extensions instead */
+ s->core_ext_mask |= DCA_EXT_XXCH;
+ break;
+
+ case 0x1d95f262: {
+ int fsize96 = show_bits(&s->gb, 12) + 1;
+ if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
+ continue;
+
+ av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
+ get_bits_count(&s->gb));
+ skip_bits(&s->gb, 12);
+ av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
+ av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
+
+ s->core_ext_mask |= DCA_EXT_X96;
+ break;
+ }
+ }
+
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+ }
+ } else {
+ /* no supported extensions, skip the rest of the core substream */
+ skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
+ }
+
+ if (s->core_ext_mask & DCA_EXT_X96)
+ s->profile = FF_PROFILE_DTS_96_24;
+ else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
+ s->profile = FF_PROFILE_DTS_ES;
+
+ /* check for ExSS (HD part) */
+ if (s->dca_buffer_size - s->frame_size > 32 &&
+ get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
+ dca_exss_parse_header(s);
+
+ avctx->profile = s->profile;
+
+ channels = s->prim_channels + !!s->lfe;
+
+ if (s->amode < 16) {
+ avctx->channel_layout = dca_core_channel_layout[s->amode];
+
+ if (s->xch_present && (!avctx->request_channels ||
+ avctx->request_channels > num_core_channels + !!s->lfe)) {
+ avctx->channel_layout |= AV_CH_BACK_CENTER;
+ if (s->lfe) {
+ avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
+ s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
+ } else {
+ s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
+ }
+ } else {
+ channels = num_core_channels + !!s->lfe;
+ s->xch_present = 0; /* disable further xch processing */
+ if (s->lfe) {
+ avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
+ s->channel_order_tab = dca_channel_reorder_lfe[s->amode];
+ } else
+ s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
+ }
+
+ if (channels > !!s->lfe &&
+ s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (avctx->request_channels == 2 && s->prim_channels > 2) {
+ channels = 2;
+ s->output = DCA_STEREO;
+ avctx->channel_layout = AV_CH_LAYOUT_STEREO;
+ }
+ } else {
+ av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
+ 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;
+ }
+ if (avctx->channels != channels) {
+ av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "
+ "channels changing in stream. Skipping frame.\n");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ /* 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 (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),
+};
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-08 14:52:07 +0000