/* * 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 FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include #include #include #include "libavutil/common.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 "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 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 DSPContext dsp; 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; } /** * Convert bitstream to one representation based on sync marker */ static int dca_convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst, int max_size) { uint32_t mrk; int i, tmp; const uint16_t *ssrc = (const uint16_t *) src; uint16_t *sdst = (uint16_t *) dst; PutBitContext pb; if ((unsigned) src_size > (unsigned) max_size) { // av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n"); // return -1; src_size = max_size; } mrk = AV_RB32(src); switch (mrk) { case DCA_MARKER_RAW_BE: memcpy(dst, src, src_size); return src_size; case DCA_MARKER_RAW_LE: for (i = 0; i < (src_size + 1) >> 1; i++) *sdst++ = av_bswap16(*ssrc++); return src_size; case DCA_MARKER_14B_BE: case DCA_MARKER_14B_LE: init_put_bits(&pb, dst, max_size); for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) { tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF; put_bits(&pb, 14, tmp); } flush_put_bits(&pb); return (put_bits_count(&pb) + 7) >> 3; default: return AVERROR_INVALIDDATA; } } /** * 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 = 0; int embedded_stereo = 0; int embedded_6ch = 0; int drc_code_present; int av_uninit(extensions_mask); int i, j; if (get_bits_left(&s->gb) < 16) return -1; /* We will parse just enough to get to the extensions bitmask with which * we can set the profile value. */ header_size = get_bits(&s->gb, 9) + 1; skip_bits(&s->gb, 3); // asset index if (s->static_fields) { if (get_bits1(&s->gb)) skip_bits(&s->gb, 4); // asset type descriptor if (get_bits1(&s->gb)) skip_bits_long(&s->gb, 24); // language descriptor if (get_bits1(&s->gb)) { /* How can one fit 1024 bytes of text here if the maximum value * for the asset header size field above was 512 bytes? */ int text_length = get_bits(&s->gb, 10) + 1; if (get_bits_left(&s->gb) < text_length * 8) return -1; skip_bits_long(&s->gb, text_length * 8); // info text } skip_bits(&s->gb, 5); // bit resolution - 1 skip_bits(&s->gb, 4); // max sample rate code channels = get_bits(&s->gb, 8) + 1; if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers int spkr_remap_sets; int spkr_mask_size = 16; int num_spkrs[7]; if (channels > 2) embedded_stereo = get_bits1(&s->gb); if (channels > 6) embedded_6ch = get_bits1(&s->gb); if (get_bits1(&s->gb)) { spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2; skip_bits(&s->gb, spkr_mask_size); // spkr activity mask } spkr_remap_sets = get_bits(&s->gb, 3); for (i = 0; i < spkr_remap_sets; i++) { /* std layout mask for each remap set */ num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size)); } for (i = 0; i < spkr_remap_sets; i++) { int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1; if (get_bits_left(&s->gb) < 0) return -1; for (j = 0; j < num_spkrs[i]; j++) { int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps); int num_dec_ch = av_popcount(remap_dec_ch_mask); skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes } } } else { skip_bits(&s->gb, 3); // representation type } } drc_code_present = get_bits1(&s->gb); if (drc_code_present) get_bits(&s->gb, 8); // drc code if (get_bits1(&s->gb)) skip_bits(&s->gb, 5); // dialog normalization code if (drc_code_present && embedded_stereo) get_bits(&s->gb, 8); // drc stereo code if (s->mix_metadata && get_bits1(&s->gb)) { skip_bits(&s->gb, 1); // external mix skip_bits(&s->gb, 6); // post mix gain code if (get_bits(&s->gb, 2) != 3) // mixer drc code skip_bits(&s->gb, 3); // drc limit else skip_bits(&s->gb, 8); // custom drc code if (get_bits1(&s->gb)) // channel specific scaling for (i = 0; i < s->num_mix_configs; i++) skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes else skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes for (i = 0; i < s->num_mix_configs; i++) { if (get_bits_left(&s->gb) < 0) return -1; dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]); if (embedded_6ch) dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]); if (embedded_stereo) dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]); } } switch (get_bits(&s->gb, 2)) { case 0: extensions_mask = get_bits(&s->gb, 12); break; case 1: extensions_mask = DCA_EXT_EXSS_XLL; break; case 2: extensions_mask = DCA_EXT_EXSS_LBR; break; case 3: extensions_mask = 0; /* aux coding */ break; } /* not parsed further, we were only interested in the extensions mask */ if (get_bits_left(&s->gb) < 0) return -1; if (get_bits_count(&s->gb) - header_pos > header_size * 8) { av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n"); return -1; } skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb)); if (extensions_mask & DCA_EXT_EXSS_XLL) s->profile = FF_PROFILE_DTS_HD_MA; else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 | DCA_EXT_EXSS_XXCH)) s->profile = FF_PROFILE_DTS_HD_HRA; if (!(extensions_mask & DCA_EXT_CORE)) av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n"); if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask) av_log(s->avctx, AV_LOG_WARNING, "DTS extensions detection mismatch (%d, %d)\n", extensions_mask & DCA_CORE_EXTS, s->core_ext_mask); return 0; } /** * 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 = 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; avctx->frame_size = s->sample_blocks * 32; 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 if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) { static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 }; s->channel_order_tab = dca_channel_order_native; } } else { av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode); return AVERROR_INVALIDDATA; } if (avctx->channels != channels) { if (avctx->channels) av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels); avctx->channels = channels; } /* get output buffer */ s->frame.nb_samples = 256 * (s->sample_blocks / 8); if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } samples_flt = (float *) s->frame.data[0]; samples_s16 = (int16_t *) s->frame.data[0]; /* filter to get final output */ for (i = 0; i < (s->sample_blocks / 8); i++) { dca_filter_channels(s, i); /* If this was marked as a DTS-ES stream we need to subtract back- */ /* channel from SL & SR to remove matrixed back-channel signal */ if ((s->source_pcm_res & 1) && s->xch_present) { float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256; float *lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256; float *rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256; s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256); s->dsp.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(); dsputil_init(&s->dsp, avctx); 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), };