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/dcadec.c | 1971 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 1971 insertions(+) create mode 100644 libavcodec/dcadec.c (limited to 'libavcodec/dcadec.c') diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c new file mode 100644 index 0000000000..b37dc49d3f --- /dev/null +++ b/libavcodec/dcadec.c @@ -0,0 +1,1971 @@ +/* + * DCA compatible decoder + * Copyright (C) 2004 Gildas Bazin + * Copyright (C) 2004 Benjamin Zores + * Copyright (C) 2006 Benjamin Larsson + * Copyright (C) 2007 Konstantin Shishkov + * + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * Libav is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include +#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 From 9e4bca16f89bc12c58b58f4611d580a30d5f9638 Mon Sep 17 00:00:00 2001 From: Diego Biurrun Date: Tue, 31 Jul 2012 20:09:23 +0200 Subject: dca: Move tables used outside of dcadec.c to a separate file. --- libavcodec/Makefile | 5 +++-- libavcodec/dca.c | 29 +++++++++++++++++++++++++++++ libavcodec/dca.h | 4 ++++ libavcodec/dca_parser.c | 3 +-- libavcodec/dcadata.h | 6 ------ libavcodec/dcadec.c | 2 +- libavformat/spdifenc.c | 3 +-- 7 files changed, 39 insertions(+), 13 deletions(-) create mode 100644 libavcodec/dca.c (limited to 'libavcodec/dcadec.c') diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 2d0006f74b..7fc50594ff 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -119,7 +119,7 @@ OBJS-$(CONFIG_CLJR_ENCODER) += cljr.o OBJS-$(CONFIG_COOK_DECODER) += cook.o OBJS-$(CONFIG_CSCD_DECODER) += cscd.o OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o -OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dcadsp.o \ +OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadsp.o \ dca_parser.o synth_filter.o OBJS-$(CONFIG_DFA_DECODER) += dfa.o OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o @@ -596,6 +596,7 @@ OBJS-$(CONFIG_OGG_DEMUXER) += xiph.o flac.o flacdata.o \ OBJS-$(CONFIG_OGG_MUXER) += xiph.o flac.o flacdata.o OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o mpegvideo.o xiph.o OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o +OBJS-$(CONFIG_SPDIF_MUXER) += dca.o OBJS-$(CONFIG_WEBM_MUXER) += mpeg4audio.o mpegaudiodata.o \ xiph.o flac.o flacdata.o OBJS-$(CONFIG_WTV_DEMUXER) += mpeg4audio.o mpegaudiodata.o @@ -641,7 +642,7 @@ OBJS-$(CONFIG_AC3_PARSER) += ac3_parser.o ac3tab.o \ OBJS-$(CONFIG_ADX_PARSER) += adx_parser.o adx.o OBJS-$(CONFIG_CAVSVIDEO_PARSER) += cavs_parser.o OBJS-$(CONFIG_COOK_PARSER) += cook_parser.o -OBJS-$(CONFIG_DCA_PARSER) += dca_parser.o +OBJS-$(CONFIG_DCA_PARSER) += dca_parser.o dca.o OBJS-$(CONFIG_DIRAC_PARSER) += dirac_parser.o OBJS-$(CONFIG_DNXHD_PARSER) += dnxhd_parser.o OBJS-$(CONFIG_DVBSUB_PARSER) += dvbsub_parser.o diff --git a/libavcodec/dca.c b/libavcodec/dca.c new file mode 100644 index 0000000000..4194f58aa9 --- /dev/null +++ b/libavcodec/dca.c @@ -0,0 +1,29 @@ +/* + * DCA compatible decoder data + * + * 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 "dca.h" + +const uint32_t ff_dca_sample_rates[16] = +{ + 0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0, + 12000, 24000, 48000, 96000, 192000 +}; diff --git a/libavcodec/dca.h b/libavcodec/dca.h index 8ea6049e0d..9235fa4f0b 100644 --- a/libavcodec/dca.h +++ b/libavcodec/dca.h @@ -25,6 +25,8 @@ #ifndef AVCODEC_DCA_H #define AVCODEC_DCA_H +#include + /** DCA syncwords, also used for bitstream type detection */ #define DCA_MARKER_RAW_BE 0x7FFE8001 #define DCA_MARKER_RAW_LE 0xFE7F0180 @@ -34,4 +36,6 @@ /** DCA-HD specific block starts with this marker. */ #define DCA_HD_MARKER 0x64582025 +extern const uint32_t ff_dca_sample_rates[16]; + #endif /* AVCODEC_DCA_H */ diff --git a/libavcodec/dca_parser.c b/libavcodec/dca_parser.c index e7b2ce42cc..553e69c41c 100644 --- a/libavcodec/dca_parser.c +++ b/libavcodec/dca_parser.c @@ -24,7 +24,6 @@ #include "parser.h" #include "dca.h" -#include "dcadata.h" #include "dca_parser.h" #include "get_bits.h" #include "put_bits.h" @@ -162,7 +161,7 @@ static int dca_parse_params(const uint8_t *buf, int buf_size, int *duration, skip_bits(&gb, 20); sr_code = get_bits(&gb, 4); - *sample_rate = dca_sample_rates[sr_code]; + *sample_rate = ff_dca_sample_rates[sr_code]; if (*sample_rate == 0) return AVERROR_INVALIDDATA; diff --git a/libavcodec/dcadata.h b/libavcodec/dcadata.h index 4b58ef7c38..324e40f104 100644 --- a/libavcodec/dcadata.h +++ b/libavcodec/dcadata.h @@ -28,12 +28,6 @@ /* Generic tables */ -static const uint32_t dca_sample_rates[16] = -{ - 0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0, - 12000, 24000, 48000, 96000, 192000 -}; - static const uint32_t dca_bit_rates[32] = { 32000, 56000, 64000, 96000, 112000, 128000, diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c index b37dc49d3f..f488da6d3f 100644 --- a/libavcodec/dcadec.c +++ b/libavcodec/dcadec.c @@ -561,7 +561,7 @@ static int dca_parse_frame_header(DCAContext *s) 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)]; + s->sample_rate = ff_dca_sample_rates[get_bits(&s->gb, 4)]; if (!s->sample_rate) return AVERROR_INVALIDDATA; s->bit_rate_index = get_bits(&s->gb, 5); diff --git a/libavformat/spdifenc.c b/libavformat/spdifenc.c index b25c7fa722..c563008b2e 100644 --- a/libavformat/spdifenc.c +++ b/libavformat/spdifenc.c @@ -49,7 +49,6 @@ #include "spdif.h" #include "libavcodec/ac3.h" #include "libavcodec/dca.h" -#include "libavcodec/dcadata.h" #include "libavcodec/aacadtsdec.h" #include "libavutil/opt.h" @@ -253,7 +252,7 @@ static int spdif_header_dts(AVFormatContext *s, AVPacket *pkt) case DCA_MARKER_RAW_BE: blocks = (AV_RB16(pkt->data + 4) >> 2) & 0x7f; core_size = ((AV_RB24(pkt->data + 5) >> 4) & 0x3fff) + 1; - sample_rate = dca_sample_rates[(pkt->data[8] >> 2) & 0x0f]; + sample_rate = ff_dca_sample_rates[(pkt->data[8] >> 2) & 0x0f]; break; case DCA_MARKER_RAW_LE: blocks = (AV_RL16(pkt->data + 4) >> 2) & 0x7f; -- cgit v1.2.3 From 19cf7163c1576e7b03ea33d7bf633e14d7516db8 Mon Sep 17 00:00:00 2001 From: Diego Biurrun Date: Wed, 1 Aug 2012 11:12:08 +0200 Subject: dca: Switch dca_sample_rates to avpriv_ prefix; it is used across libs --- libavcodec/dca.c | 2 +- libavcodec/dca.h | 2 +- libavcodec/dca_parser.c | 2 +- libavcodec/dcadec.c | 2 +- libavformat/spdifenc.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) (limited to 'libavcodec/dcadec.c') diff --git a/libavcodec/dca.c b/libavcodec/dca.c index 4194f58aa9..0f1eeecf7b 100644 --- a/libavcodec/dca.c +++ b/libavcodec/dca.c @@ -22,7 +22,7 @@ #include "dca.h" -const uint32_t ff_dca_sample_rates[16] = +const uint32_t avpriv_dca_sample_rates[16] = { 0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000 diff --git a/libavcodec/dca.h b/libavcodec/dca.h index 9235fa4f0b..1515270471 100644 --- a/libavcodec/dca.h +++ b/libavcodec/dca.h @@ -36,6 +36,6 @@ /** DCA-HD specific block starts with this marker. */ #define DCA_HD_MARKER 0x64582025 -extern const uint32_t ff_dca_sample_rates[16]; +extern const uint32_t avpriv_dca_sample_rates[16]; #endif /* AVCODEC_DCA_H */ diff --git a/libavcodec/dca_parser.c b/libavcodec/dca_parser.c index 553e69c41c..73611e0233 100644 --- a/libavcodec/dca_parser.c +++ b/libavcodec/dca_parser.c @@ -161,7 +161,7 @@ static int dca_parse_params(const uint8_t *buf, int buf_size, int *duration, skip_bits(&gb, 20); sr_code = get_bits(&gb, 4); - *sample_rate = ff_dca_sample_rates[sr_code]; + *sample_rate = avpriv_dca_sample_rates[sr_code]; if (*sample_rate == 0) return AVERROR_INVALIDDATA; diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c index f488da6d3f..d4fd23e215 100644 --- a/libavcodec/dcadec.c +++ b/libavcodec/dcadec.c @@ -561,7 +561,7 @@ static int dca_parse_frame_header(DCAContext *s) if (s->frame_size < 95) return AVERROR_INVALIDDATA; s->amode = get_bits(&s->gb, 6); - s->sample_rate = ff_dca_sample_rates[get_bits(&s->gb, 4)]; + s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)]; if (!s->sample_rate) return AVERROR_INVALIDDATA; s->bit_rate_index = get_bits(&s->gb, 5); diff --git a/libavformat/spdifenc.c b/libavformat/spdifenc.c index c563008b2e..f8c38c44ab 100644 --- a/libavformat/spdifenc.c +++ b/libavformat/spdifenc.c @@ -252,7 +252,7 @@ static int spdif_header_dts(AVFormatContext *s, AVPacket *pkt) case DCA_MARKER_RAW_BE: blocks = (AV_RB16(pkt->data + 4) >> 2) & 0x7f; core_size = ((AV_RB24(pkt->data + 5) >> 4) & 0x3fff) + 1; - sample_rate = ff_dca_sample_rates[(pkt->data[8] >> 2) & 0x0f]; + sample_rate = avpriv_dca_sample_rates[(pkt->data[8] >> 2) & 0x0f]; break; case DCA_MARKER_RAW_LE: blocks = (AV_RL16(pkt->data + 4) >> 2) & 0x7f; -- cgit v1.2.3