From 13a79cf84e073d0ca8489047660352eee216d059 Mon Sep 17 00:00:00 2001 From: Diego Biurrun Date: Tue, 31 Jul 2012 20:00:35 +0200 Subject: dca: Rename dca.c ---> dcadec.c This will allow adding dca.c with tables used from other files. --- libavcodec/dca.c | 1971 ------------------------------------------------------ 1 file changed, 1971 deletions(-) delete mode 100644 libavcodec/dca.c (limited to 'libavcodec/dca.c') 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 -#include -#include - -#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), -}; -- cgit v1.2.3