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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/dca.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/dca.c')
-rw-r--r--libavcodec/dca.c1971
1 files changed, 0 insertions, 1971 deletions
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
deleted file mode 100644
index b37dc49d3f..0000000000
--- a/libavcodec/dca.c
+++ /dev/null
@@ -1,1971 +0,0 @@
-/*
- * 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 2024-05-18 04:07:25 +0000