/* * DCA compatible decoder * Copyright (C) 2004 Gildas Bazin * Copyright (C) 2004 Benjamin Zores * Copyright (C) 2006 Benjamin Larsson * Copyright (C) 2007 Konstantin Shishkov * Copyright (C) 2012 Paul B Mahol * Copyright (C) 2014 Niels Möller * * 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/attributes.h" #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "libavutil/float_dsp.h" #include "libavutil/internal.h" #include "libavutil/intreadwrite.h" #include "libavutil/mathematics.h" #include "libavutil/opt.h" #include "libavutil/samplefmt.h" #include "avcodec.h" #include "dca.h" #include "dca_syncwords.h" #include "dcadata.h" #include "dcadsp.h" #include "dcahuff.h" #include "fft.h" #include "fmtconvert.h" #include "get_bits.h" #include "internal.h" #include "mathops.h" #include "synth_filter.h" #if ARCH_ARM # include "arm/dca.h" #endif 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 }; enum DCAXxchSpeakerMask { DCA_XXCH_FRONT_CENTER = 0x0000001, DCA_XXCH_FRONT_LEFT = 0x0000002, DCA_XXCH_FRONT_RIGHT = 0x0000004, DCA_XXCH_SIDE_REAR_LEFT = 0x0000008, DCA_XXCH_SIDE_REAR_RIGHT = 0x0000010, DCA_XXCH_LFE1 = 0x0000020, DCA_XXCH_REAR_CENTER = 0x0000040, DCA_XXCH_SURROUND_REAR_LEFT = 0x0000080, DCA_XXCH_SURROUND_REAR_RIGHT = 0x0000100, DCA_XXCH_SIDE_SURROUND_LEFT = 0x0000200, DCA_XXCH_SIDE_SURROUND_RIGHT = 0x0000400, DCA_XXCH_FRONT_CENTER_LEFT = 0x0000800, DCA_XXCH_FRONT_CENTER_RIGHT = 0x0001000, DCA_XXCH_FRONT_HIGH_LEFT = 0x0002000, DCA_XXCH_FRONT_HIGH_CENTER = 0x0004000, DCA_XXCH_FRONT_HIGH_RIGHT = 0x0008000, DCA_XXCH_LFE2 = 0x0010000, DCA_XXCH_SIDE_FRONT_LEFT = 0x0020000, DCA_XXCH_SIDE_FRONT_RIGHT = 0x0040000, DCA_XXCH_OVERHEAD = 0x0080000, DCA_XXCH_SIDE_HIGH_LEFT = 0x0100000, DCA_XXCH_SIDE_HIGH_RIGHT = 0x0200000, DCA_XXCH_REAR_HIGH_CENTER = 0x0400000, DCA_XXCH_REAR_HIGH_LEFT = 0x0800000, DCA_XXCH_REAR_HIGH_RIGHT = 0x1000000, DCA_XXCH_REAR_LOW_CENTER = 0x2000000, DCA_XXCH_REAR_LOW_LEFT = 0x4000000, DCA_XXCH_REAR_LOW_RIGHT = 0x8000000, }; #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_NSYNCAUX 0x9A1105A0 #define SAMPLES_PER_SUBBAND 8 // number of samples per subband per subsubframe /** Bit allocation */ typedef struct BitAlloc { 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; } static float dca_dmix_code(unsigned code); 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[ff_dca_vlc_offs[i]]; dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_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[ff_dca_vlc_offs[i + 5]]; dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_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[ff_dca_vlc_offs[i + 10]]; dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_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[ff_dca_vlc_offs[c]]; dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_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 inline int dca_xxch2index(DCAContext *s, int xxch_ch) { int i, base, mask; /* locate channel set containing the channel */ for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1); i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i]) base += av_popcount(mask); return base + av_popcount(mask & (xxch_ch - 1)); } static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, int xxch) { 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 }; int hdr_pos = 0, hdr_size = 0; float scale_factor; int this_chans, acc_mask; int embedded_downmix; int nchans, mask[8]; int coeff, ichan; /* xxch has arbitrary sized audio coding headers */ if (xxch) { hdr_pos = get_bits_count(&s->gb); hdr_size = get_bits(&s->gb, 7) + 1; } nchans = get_bits(&s->gb, 3) + 1; if (xxch && nchans >= 3) { av_log(s->avctx, AV_LOG_ERROR, "nchans %d is too large\n", nchans); return AVERROR_INVALIDDATA; } else if (nchans + base_channel > DCA_PRIM_CHANNELS_MAX) { av_log(s->avctx, AV_LOG_ERROR, "channel sum %d + %d is too large\n", nchans, base_channel); return AVERROR_INVALIDDATA; } s->audio_header.total_channels = nchans + base_channel; s->audio_header.prim_channels = s->audio_header.total_channels; /* obtain speaker layout mask & downmix coefficients for XXCH */ if (xxch) { acc_mask = s->xxch_core_spkmask; this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6; s->xxch_spk_masks[s->xxch_chset] = this_chans; s->xxch_chset_nch[s->xxch_chset] = nchans; for (i = 0; i <= s->xxch_chset; i++) acc_mask |= s->xxch_spk_masks[i]; /* check for downmixing information */ if (get_bits1(&s->gb)) { embedded_downmix = get_bits1(&s->gb); coeff = get_bits(&s->gb, 6); if (coeff<1 || coeff>61) { av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff); return AVERROR_INVALIDDATA; } scale_factor = -1.0f / dca_dmix_code((coeff<<2)-3); s->xxch_dmix_sf[s->xxch_chset] = scale_factor; for (i = base_channel; i < s->audio_header.prim_channels; i++) { mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask); } for (j = base_channel; j < s->audio_header.prim_channels; j++) { memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0])); s->xxch_dmix_embedded |= (embedded_downmix << j); for (i = 0; i < s->xxch_nbits_spk_mask; i++) { if (mask[j] & (1 << i)) { if ((1 << i) == DCA_XXCH_LFE1) { av_log(s->avctx, AV_LOG_WARNING, "DCA-XXCH: dmix to LFE1 not supported.\n"); continue; } coeff = get_bits(&s->gb, 7); ichan = dca_xxch2index(s, 1 << i); if ((coeff&63)<1 || (coeff&63)>61) { av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff); return AVERROR_INVALIDDATA; } s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3); } } } } } if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX) s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX; for (i = base_channel; i < s->audio_header.prim_channels; i++) { s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2; if (s->audio_header.subband_activity[i] > DCA_SUBBANDS) s->audio_header.subband_activity[i] = DCA_SUBBANDS; } for (i = base_channel; i < s->audio_header.prim_channels; i++) { s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1; if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS) s->audio_header.vq_start_subband[i] = DCA_SUBBANDS; } get_array(&s->gb, s->audio_header.joint_intensity + base_channel, s->audio_header.prim_channels - base_channel, 3); get_array(&s->gb, s->audio_header.transient_huffman + base_channel, s->audio_header.prim_channels - base_channel, 2); get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel, s->audio_header.prim_channels - base_channel, 3); get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel, s->audio_header.prim_channels - base_channel, 3); /* Get codebooks quantization indexes */ if (!base_channel) memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.quant_index_huffman)); for (j = 1; j < 11; j++) for (i = base_channel; i < s->audio_header.prim_channels; i++) s->audio_header.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->audio_header.prim_channels; i++) s->audio_header.scalefactor_adj[i][j] = 1; for (j = 1; j < 11; j++) for (i = base_channel; i < s->audio_header.prim_channels; i++) if (s->audio_header.quant_index_huffman[i][j] < thr[j]) s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; if (!xxch) { if (s->crc_present) { /* Audio header CRC check */ get_bits(&s->gb, 16); } } else { /* Skip to the end of the header, also ignore CRC if present */ i = get_bits_count(&s->gb); if (hdr_pos + 8 * hdr_size > i) skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i); } s->current_subframe = 0; s->current_subsubframe = 0; 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 = 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); s->bit_rate = ff_dca_bit_rates[s->bit_rate_index]; if (!s->bit_rate) return AVERROR_INVALIDDATA; skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1) 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); if (s->lfe > 2) { s->lfe = 0; av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe); return AVERROR_INVALIDDATA; } /* 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; /* Primary audio coding header */ s->audio_header.subframes = get_bits(&s->gb, 4) + 1; return dca_parse_audio_coding_header(s, 0, 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; if (block_index + s->subsubframes[s->current_subframe] > (s->sample_blocks / SAMPLES_PER_SUBBAND)) { s->subsubframes[s->current_subframe] = 1; return AVERROR_INVALIDDATA; } s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3); } for (j = base_channel; j < s->audio_header.prim_channels; j++) { for (k = 0; k < s->audio_header.subband_activity[j]; k++) s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1); } /* Get prediction codebook */ for (j = base_channel; j < s->audio_header.prim_channels; j++) { for (k = 0; k < s->audio_header.subband_activity[j]; k++) { if (s->dca_chan[j].prediction_mode[k] > 0) { /* (Prediction coefficient VQ address) */ s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12); } } } /* Bit allocation index */ for (j = base_channel; j < s->audio_header.prim_channels; j++) { for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) { if (s->audio_header.bitalloc_huffman[j] == 6) s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5); else if (s->audio_header.bitalloc_huffman[j] == 5) s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4); else if (s->audio_header.bitalloc_huffman[j] == 7) { av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation index\n"); return AVERROR_INVALIDDATA; } else { s->dca_chan[j].bitalloc[k] = get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]); } if (s->dca_chan[j].bitalloc[k] > 26) { ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n", j, k, s->dca_chan[j].bitalloc[k]); return AVERROR_INVALIDDATA; } } } /* Transition mode */ for (j = base_channel; j < s->audio_header.prim_channels; j++) { for (k = 0; k < s->audio_header.subband_activity[j]; k++) { s->dca_chan[j].transition_mode[k] = 0; if (s->subsubframes[s->current_subframe] > 1 && k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) { s->dca_chan[j].transition_mode[k] = get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]); } } } if (get_bits_left(&s->gb) < 0) return AVERROR_INVALIDDATA; for (j = base_channel; j < s->audio_header.prim_channels; j++) { const uint32_t *scale_table; int scale_sum, log_size; memset(s->dca_chan[j].scale_factor, 0, s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2); if (s->audio_header.scalefactor_huffman[j] == 6) { scale_table = ff_dca_scale_factor_quant7; log_size = 7; } else { scale_table = ff_dca_scale_factor_quant6; log_size = 6; } /* When huffman coded, only the difference is encoded */ scale_sum = 0; for (k = 0; k < s->audio_header.subband_activity[j]; k++) { if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) { scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size); s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum]; } if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) { /* Get second scale factor */ scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size); s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum]; } } } /* Joint subband scale factor codebook select */ for (j = base_channel; j < s->audio_header.prim_channels; j++) { /* Transmitted only if joint subband coding enabled */ if (s->audio_header.joint_intensity[j] > 0) s->dca_chan[j].joint_huff = 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->audio_header.prim_channels; j++) { int source_channel; /* Transmitted only if joint subband coding enabled */ if (s->audio_header.joint_intensity[j] > 0) { int scale = 0; source_channel = s->audio_header.joint_intensity[j] - 1; /* When huffman coded, only the difference is encoded * (is this valid as well for joint scales ???) */ for (k = s->audio_header.subband_activity[j]; k < s->audio_header.subband_activity[source_channel]; k++) { scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7); s->dca_chan[j].joint_scale_factor[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; } } } /* 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->audio_header.prim_channels; j++) for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++) /* 1 vector -> 32 samples */ s->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10); /* Low frequency effect data */ if (!base_channel && s->lfe) { int quant7; /* 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 */ quant7 = get_bits(&s->gb, 8); if (quant7 > 127) { avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127"); return AVERROR_INVALIDDATA; } s->lfe_scale_factor = ff_dca_scale_factor_quant7[quant7]; /* 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; } return 0; } static void qmf_32_subbands(DCAContext *s, int chans, float samples_in[32][SAMPLES_PER_SUBBAND], float *samples_out, float scale) { const float *prCoeff; int sb_act = s->audio_header.subband_activity[chans]; scale *= sqrt(1 / 8.0); /* Select filter */ if (!s->multirate_inter) /* Non-perfect reconstruction */ prCoeff = ff_dca_fir_32bands_nonperfect; else /* Perfect reconstruction */ prCoeff = ff_dca_fir_32bands_perfect; s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct, s->dca_chan[chans].subband_fir_hist, &s->dca_chan[chans].hist_index, s->dca_chan[chans].subband_fir_noidea, prCoeff, samples_out, s->raXin, scale); } static QMF64_table *qmf64_precompute(void) { unsigned i, j; QMF64_table *table = av_malloc(sizeof(*table)); if (!table) return NULL; for (i = 0; i < 32; i++) for (j = 0; j < 32; j++) table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128); for (i = 0; i < 32; i++) for (j = 0; j < 32; j++) table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64); /* FIXME: Is the factor 0.125 = 1/8 right? */ for (i = 0; i < 32; i++) table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256); for (i = 0; i < 32; i++) table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256); return table; } /* FIXME: Totally unoptimized. Based on the reference code and * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks * for doubling the size. */ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPLES_PER_SUBBAND], float *samples_out, float scale) { float raXin[64]; float A[32], B[32]; float *raX = s->dca_chan[chans].subband_fir_hist; float *raZ = s->dca_chan[chans].subband_fir_noidea; unsigned i, j, k, subindex; for (i = s->audio_header.subband_activity[chans]; i < 64; i++) raXin[i] = 0.0; for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) { for (i = 0; i < s->audio_header.subband_activity[chans]; i++) raXin[i] = samples_in[i][subindex]; for (k = 0; k < 32; k++) { A[k] = 0.0; for (i = 0; i < 32; i++) A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i]; } for (k = 0; k < 32; k++) { B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0]; for (i = 1; i < 32; i++) B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i]; } for (k = 0; k < 32; k++) { raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]); raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]); } for (i = 0; i < 64; i++) { float out = raZ[i]; for (j = 0; j < 1024; j += 128) out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]); *samples_out++ = out * scale; } for (i = 0; i < 64; i++) { float hist = 0.0; for (j = 0; j < 1024; j += 128) hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]); raZ[i] = hist; } /* FIXME: Make buffer circular, to avoid this move. */ memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX)); } } static void lfe_interpolation_fir(DCAContext *s, const float *samples_in, float *samples_out) { /* 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 idx; const float *prCoeff; int deciindex; /* Select decimation filter */ if (s->lfe == 1) { idx = 1; prCoeff = ff_dca_lfe_fir_128; } else { idx = 0; if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) prCoeff = ff_dca_lfe_xll_fir_64; else prCoeff = ff_dca_lfe_fir_64; } /* Interpolation */ for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) { s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff); samples_in++; samples_out += 2 * 32 * (1 + idx); } } /* downmixing routines */ #define MIX_REAR1(samples, s1, rs, coef) \ samples[0][i] += samples[s1][i] * coef[rs][0]; \ samples[1][i] += samples[s1][i] * coef[rs][1]; #define MIX_REAR2(samples, s1, s2, rs, coef) \ samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \ samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1]; #define MIX_FRONT3(samples, coef) \ t = samples[c][i]; \ u = samples[l][i]; \ v = samples[r][i]; \ samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \ samples[1][i] = 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 lfe_present, float coef[DCA_PRIM_CHANNELS_MAX + 1][2], const int8_t *channel_mapping) { int c, l, r, sl, sr, s; int i; float t, u, v; switch (srcfmt) { case DCA_MONO: case DCA_4F2R: av_log(NULL, AV_LOG_ERROR, "Not implemented!\n"); break; case DCA_CHANNEL: case DCA_STEREO: case DCA_STEREO_TOTAL: case DCA_STEREO_SUMDIFF: break; case DCA_3F: c = channel_mapping[0]; l = channel_mapping[1]; r = channel_mapping[2]; DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), ); break; case DCA_2F1R: s = channel_mapping[2]; DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), ); break; case DCA_3F1R: c = channel_mapping[0]; l = channel_mapping[1]; r = channel_mapping[2]; s = channel_mapping[3]; DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), MIX_REAR1(samples, s, 3, coef)); break; case DCA_2F2R: sl = channel_mapping[2]; sr = channel_mapping[3]; DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), ); break; case DCA_3F2R: c = channel_mapping[0]; l = channel_mapping[1]; r = channel_mapping[2]; sl = channel_mapping[3]; sr = channel_mapping[4]; DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), MIX_REAR2(samples, sl, sr, 3, coef)); break; } if (lfe_present) { int lf_buf = ff_dca_lfe_index[srcfmt]; int lf_idx = ff_dca_channels[srcfmt]; for (i = 0; i < 256; i++) { samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0]; samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1]; } } } #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, int32_t *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, int32_t *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 }; 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; LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]); /* * Audio data */ /* Select quantization step size table */ if (s->bit_rate_index == 0x1f) quant_step_table = ff_dca_lossless_quant_d; else quant_step_table = ff_dca_lossy_quant_d; for (k = base_channel; k < s->audio_header.prim_channels; k++) { float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index]; float rscale[DCA_SUBBANDS]; if (get_bits_left(&s->gb) < 0) return AVERROR_INVALIDDATA; for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) { int m; /* Select the mid-tread linear quantizer */ int abits = s->dca_chan[k].bitalloc[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->audio_header.quant_index_huffman[k][abits]; /* * Extract bits from the bit stream */ if (!abits) { rscale[l] = 0; memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0])); } else { /* Deal with transients */ int sfi = s->dca_chan[k].transition_mode[l] && subsubframe >= s->dca_chan[k].transition_mode[l]; rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] * s->audio_header.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 + SAMPLES_PER_SUBBAND * l); 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 < SAMPLES_PER_SUBBAND; m++) block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3); } } else { /* Huffman coded */ for (m = 0; m < SAMPLES_PER_SUBBAND; m++) block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel); } } } s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0], block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]); for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) { int m; /* * Inverse ADPCM if in prediction mode */ if (s->dca_chan[k].prediction_mode[l]) { int n; if (s->predictor_history) subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * s->dca_chan[k].subband_samples_hist[l][3] + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] * s->dca_chan[k].subband_samples_hist[l][2] + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] * s->dca_chan[k].subband_samples_hist[l][1] + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] * s->dca_chan[k].subband_samples_hist[l][0]) * (1.0f / 8192); for (m = 1; m < SAMPLES_PER_SUBBAND; m++) { float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * subband_samples[l][m - 1]; for (n = 2; n <= 4; n++) if (m >= n) sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * subband_samples[l][m - n]; else if (s->predictor_history) sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * s->dca_chan[k].subband_samples_hist[l][m - n + 4]; subband_samples[l][m] += sum * (1.0f / 8192); } } } /* Backup predictor history for adpcm */ for (l = 0; l < DCA_SUBBANDS; l++) AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]); /* * Decode VQ encoded high frequencies */ if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) { if (!(s->debug_flag & 0x01)) { av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); s->debug_flag |= 0x01; } s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq, ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND, s->dca_chan[k].scale_factor, s->audio_header.vq_start_subband[k], s->audio_header.subband_activity[k]); } } /* Check for DSYNC after subsubframe */ if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) { if (get_bits(&s->gb, 16) != 0xFFFF) { av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n"); return AVERROR_INVALIDDATA; } } return 0; } static int dca_filter_channels(DCAContext *s, int block_index, int upsample) { int k; if (upsample) { if (!s->qmf64_table) { s->qmf64_table = qmf64_precompute(); if (!s->qmf64_table) return AVERROR(ENOMEM); } /* 64 subbands QMF */ for (k = 0; k < s->audio_header.prim_channels; k++) { float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index]; if (s->channel_order_tab[k] >= 0) qmf_64_subbands(s, k, subband_samples, s->samples_chanptr[s->channel_order_tab[k]], /* Upsampling needs a factor 2 here. */ M_SQRT2 / 32768.0); } } else { /* 32 subbands QMF */ for (k = 0; k < s->audio_header.prim_channels; k++) { float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index]; if (s->channel_order_tab[k] >= 0) qmf_32_subbands(s, k, subband_samples, s->samples_chanptr[s->channel_order_tab[k]], M_SQRT1_2 / 32768.0); } } /* Generate LFE samples for this subsubframe FIXME!!! */ if (s->lfe) { float *samples = s->samples_chanptr[s->lfe_index]; lfe_interpolation_fir(s, s->lfe_data + 2 * s->lfe * (block_index + 4), samples); if (upsample) { unsigned i; /* Should apply the filter in Table 6-11 when upsampling. For * now, just duplicate. */ for (i = 255; i > 0; i--) { samples[2 * i] = samples[2 * i + 1] = samples[i]; } samples[1] = samples[0]; } } /* FIXME: This downmixing is probably broken with upsample. * Probably totally broken also with XLL in general. */ /* Downmixing to Stereo */ if (s->audio_header.prim_channels + !!s->lfe > 2 && s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef, s->channel_order_tab); } return 0; } static int dca_subframe_footer(DCAContext *s, int base_channel) { int in, out, aux_data_count, aux_data_end, reserved; uint32_t nsyncaux; /* * 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); // align (32-bit) skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31); aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb); if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) { av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n", nsyncaux); return AVERROR_INVALIDDATA; } if (get_bits1(&s->gb)) { // bAUXTimeStampFlag avpriv_request_sample(s->avctx, "Auxiliary Decode Time Stamp Flag"); // align (4-bit) skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4); // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4) skip_bits_long(&s->gb, 44); } if ((s->core_downmix = get_bits1(&s->gb))) { int am = get_bits(&s->gb, 3); switch (am) { case 0: s->core_downmix_amode = DCA_MONO; break; case 1: s->core_downmix_amode = DCA_STEREO; break; case 2: s->core_downmix_amode = DCA_STEREO_TOTAL; break; case 3: s->core_downmix_amode = DCA_3F; break; case 4: s->core_downmix_amode = DCA_2F1R; break; case 5: s->core_downmix_amode = DCA_2F2R; break; case 6: s->core_downmix_amode = DCA_3F1R; break; default: av_log(s->avctx, AV_LOG_ERROR, "Invalid mode %d for embedded downmix coefficients\n", am); return AVERROR_INVALIDDATA; } for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) { for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) { uint16_t tmp = get_bits(&s->gb, 9); if ((tmp & 0xFF) > 241) { av_log(s->avctx, AV_LOG_ERROR, "Invalid downmix coefficient code %"PRIu16"\n", tmp); return AVERROR_INVALIDDATA; } s->core_downmix_codes[in][out] = tmp; } } } align_get_bits(&s->gb); // byte align skip_bits(&s->gb, 16); // nAUXCRC16 // additional data (reserved, cf. ETSI TS 102 114 V1.4.1) if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) { av_log(s->avctx, AV_LOG_ERROR, "Overread auxiliary data by %d bits\n", -reserved); return AVERROR_INVALIDDATA; } else if (reserved) { avpriv_request_sample(s->avctx, "Core auxiliary data reserved content"); skip_bits_long(&s->gb, reserved); } } if (s->crc_present && 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->audio_header.subframes) { av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", s->current_subframe, s->audio_header.subframes); return AVERROR_INVALIDDATA; } if (!s->current_subsubframe) { /* Read subframe header */ if ((ret = dca_subframe_header(s, base_channel, block_index))) return ret; } /* Read subsubframe */ 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->audio_header.subframes) { /* Read subframe footer */ if ((ret = dca_subframe_footer(s, base_channel))) return ret; } return 0; } int ff_dca_xbr_parse_frame(DCAContext *s) { int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2]; int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX]; int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS]; int anctemp[DCA_CHSET_CHANS_MAX]; int chset_fsize[DCA_CHSETS_MAX]; int n_xbr_ch[DCA_CHSETS_MAX]; int hdr_size, num_chsets, xbr_tmode, hdr_pos; int i, j, k, l, chset, chan_base; av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n"); /* get bit position of sync header */ hdr_pos = get_bits_count(&s->gb) - 32; hdr_size = get_bits(&s->gb, 6) + 1; num_chsets = get_bits(&s->gb, 2) + 1; for(i = 0; i < num_chsets; i++) chset_fsize[i] = get_bits(&s->gb, 14) + 1; xbr_tmode = get_bits1(&s->gb); for(i = 0; i < num_chsets; i++) { n_xbr_ch[i] = get_bits(&s->gb, 3) + 1; k = get_bits(&s->gb, 2) + 5; for(j = 0; j < n_xbr_ch[i]; j++) { active_bands[i][j] = get_bits(&s->gb, k) + 1; if (active_bands[i][j] > DCA_SUBBANDS) { av_log(s->avctx, AV_LOG_ERROR, "too many active subbands (%d)\n", active_bands[i][j]); return AVERROR_INVALIDDATA; } } } /* skip to the end of the header */ i = get_bits_count(&s->gb); if(hdr_pos + hdr_size * 8 > i) skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i); /* loop over the channel data sets */ /* only decode as many channels as we've decoded base data for */ for(chset = 0, chan_base = 0; chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->audio_header.prim_channels; chan_base += n_xbr_ch[chset++]) { int start_posn = get_bits_count(&s->gb); int subsubframe = 0; int subframe = 0; /* loop over subframes */ for (k = 0; k < (s->sample_blocks / 8); k++) { /* parse header if we're on first subsubframe of a block */ if(subsubframe == 0) { /* Parse subframe header */ for(i = 0; i < n_xbr_ch[chset]; i++) { anctemp[i] = get_bits(&s->gb, 2) + 2; } for(i = 0; i < n_xbr_ch[chset]; i++) { get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]); } for(i = 0; i < n_xbr_ch[chset]; i++) { anctemp[i] = get_bits(&s->gb, 3); if(anctemp[i] < 1) { av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n"); return AVERROR_INVALIDDATA; } } /* generate scale factors */ for(i = 0; i < n_xbr_ch[chset]; i++) { const uint32_t *scale_table; int nbits; int scale_table_size; if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) { scale_table = ff_dca_scale_factor_quant7; scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7); } else { scale_table = ff_dca_scale_factor_quant6; scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6); } nbits = anctemp[i]; for(j = 0; j < active_bands[chset][i]; j++) { if(abits_high[i][j] > 0) { int index = get_bits(&s->gb, nbits); if (index >= scale_table_size) { av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index); return AVERROR_INVALIDDATA; } scale_table_high[i][j][0] = scale_table[index]; if(xbr_tmode && s->dca_chan[i].transition_mode[j]) { int index = get_bits(&s->gb, nbits); if (index >= scale_table_size) { av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index); return AVERROR_INVALIDDATA; } scale_table_high[i][j][1] = scale_table[index]; } } } } } /* decode audio array for this block */ for(i = 0; i < n_xbr_ch[chset]; i++) { for(j = 0; j < active_bands[chset][i]; j++) { const int xbr_abits = abits_high[i][j]; const float quant_step_size = ff_dca_lossless_quant_d[xbr_abits]; const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j]; const float rscale = quant_step_size * scale_table_high[i][j][sfi]; float *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j]; int block[8]; if(xbr_abits <= 0) continue; if(xbr_abits > 7) { get_array(&s->gb, block, 8, xbr_abits - 3); } else { int block_code1, block_code2, size, levels, err; size = abits_sizes[xbr_abits - 1]; levels = abits_levels[xbr_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: DTS-XBR: block code look-up failed\n"); return AVERROR_INVALIDDATA; } } /* scale & sum into subband */ for(l = 0; l < 8; l++) subband_samples[l] += (float)block[l] * rscale; } } /* check DSYNC marker */ if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) { if(get_bits(&s->gb, 16) != 0xffff) { av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n"); return AVERROR_INVALIDDATA; } } /* advance sub-sub-frame index */ if(++subsubframe >= s->subsubframes[subframe]) { subsubframe = 0; subframe++; } } /* skip to next channel set */ i = get_bits_count(&s->gb); if(start_posn + chset_fsize[chset] * 8 != i) { j = start_posn + chset_fsize[chset] * 8 - i; if(j < 0 || j >= 8) av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set," " skipping further than expected (%d bits)\n", j); skip_bits_long(&s->gb, j); } } return 0; } /* parse initial header for XXCH and dump details */ int ff_dca_xxch_decode_frame(DCAContext *s) { int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos; int i, chset, base_channel, chstart, fsize[8]; /* assume header word has already been parsed */ hdr_pos = get_bits_count(&s->gb) - 32; hdr_size = get_bits(&s->gb, 6) + 1; /*chhdr_crc =*/ skip_bits1(&s->gb); spkmsk_bits = get_bits(&s->gb, 5) + 1; num_chsets = get_bits(&s->gb, 2) + 1; for (i = 0; i < num_chsets; i++) fsize[i] = get_bits(&s->gb, 14) + 1; core_spk = get_bits(&s->gb, spkmsk_bits); s->xxch_core_spkmask = core_spk; s->xxch_nbits_spk_mask = spkmsk_bits; s->xxch_dmix_embedded = 0; /* skip to the end of the header */ i = get_bits_count(&s->gb); if (hdr_pos + hdr_size * 8 > i) skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i); for (chset = 0; chset < num_chsets; chset++) { chstart = get_bits_count(&s->gb); base_channel = s->audio_header.prim_channels; s->xxch_chset = chset; /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs. 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */ dca_parse_audio_coding_header(s, base_channel, 1); /* decode channel data */ for (i = 0; i < (s->sample_blocks / 8); i++) { if (dca_decode_block(s, base_channel, i)) { av_log(s->avctx, AV_LOG_ERROR, "Error decoding DTS-XXCH extension\n"); continue; } } /* skip to end of this section */ i = get_bits_count(&s->gb); if (chstart + fsize[chset] * 8 > i) skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i); } s->xxch_chset = num_chsets; return 0; } static float dca_dmix_code(unsigned code) { int sign = (code >> 8) - 1; code &= 0xff; return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15)); } static int scan_for_extensions(AVCodecContext *avctx) { DCAContext *s = avctx->priv_data; int core_ss_end, ret = 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 | DCA_EXT_XXCH)) { /* 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); int i; switch (bits) { case DCA_SYNCWORD_XCH: { int ext_amode, xch_fsize; s->xch_base_channel = s->audio_header.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; } if (s->xch_base_channel < 2) { avpriv_request_sample(avctx, "XCh with fewer than 2 base channels"); continue; } /* much like core primary audio coding header */ dca_parse_audio_coding_header(s, s->xch_base_channel, 0); 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 DCA_SYNCWORD_XXCH: /* 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; ff_dca_xxch_decode_frame(s); 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_SYNCWORD_SUBSTREAM) ff_dca_exss_parse_header(s); return ret; } static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core_channels) { DCAContext *s = avctx->priv_data; int i, j, chset, mask; int channel_layout, channel_mask; int posn, lavc; /* If we have XXCH then the channel layout is managed differently */ /* note that XLL will also have another way to do things */ if (!(s->core_ext_mask & DCA_EXT_XXCH)) { /* xxx should also do MA extensions */ if (s->amode < 16) { avctx->channel_layout = ff_dca_core_channel_layout[s->amode]; if (s->audio_header.prim_channels + !!s->lfe > 2 && avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { /* * Neither the core's auxiliary data nor our default tables contain * downmix coefficients for the additional channel coded in the XCh * extension, so when we're doing a Stereo downmix, don't decode it. */ s->xch_disable = 1; } if (s->xch_present && !s->xch_disable) { if (avctx->channel_layout & AV_CH_BACK_CENTER) { avpriv_request_sample(avctx, "XCh with Back center channel"); return AVERROR_INVALIDDATA; } avctx->channel_layout |= AV_CH_BACK_CENTER; if (s->lfe) { avctx->channel_layout |= AV_CH_LOW_FREQUENCY; s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode]; } else { s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode]; } if (s->channel_order_tab[s->xch_base_channel] < 0) return AVERROR_INVALIDDATA; } 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 = ff_dca_channel_reorder_lfe[s->amode]; } else s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode]; } if (*channels > !!s->lfe && s->channel_order_tab[*channels - 1 - !!s->lfe] < 0) return AVERROR_INVALIDDATA; if (av_get_channel_layout_nb_channels(avctx->channel_layout) != *channels) { av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", *channels, av_get_channel_layout_nb_channels(avctx->channel_layout)); return AVERROR_INVALIDDATA; } if (num_core_channels + !!s->lfe > 2 && avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { *channels = 2; s->output = s->audio_header.prim_channels == 2 ? s->amode : 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; } s->lfe_index = ff_dca_lfe_index[s->amode]; } else { av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode); return AVERROR_INVALIDDATA; } s->xxch_dmix_embedded = 0; } else { /* we only get here if an XXCH channel set can be added to the mix */ channel_mask = s->xxch_core_spkmask; { *channels = s->audio_header.prim_channels + !!s->lfe; for (i = 0; i < s->xxch_chset; i++) { channel_mask |= s->xxch_spk_masks[i]; } } /* Given the DTS spec'ed channel mask, generate an avcodec version */ channel_layout = 0; for (i = 0; i < s->xxch_nbits_spk_mask; ++i) { if (channel_mask & (1 << i)) { channel_layout |= ff_dca_map_xxch_to_native[i]; } } /* make sure that we have managed to get equivalent dts/avcodec channel * masks in some sense -- unfortunately some channels could overlap */ if (av_popcount(channel_mask) != av_popcount(channel_layout)) { av_log(avctx, AV_LOG_DEBUG, "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n"); return AVERROR_INVALIDDATA; } avctx->channel_layout = channel_layout; if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) { /* Estimate DTS --> avcodec ordering table */ for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) { mask = chset >= 0 ? s->xxch_spk_masks[chset] : s->xxch_core_spkmask; for (i = 0; i < s->xxch_nbits_spk_mask; i++) { if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) { lavc = ff_dca_map_xxch_to_native[i]; posn = av_popcount(channel_layout & (lavc - 1)); s->xxch_order_tab[j++] = posn; } } } s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1)); } else { /* native ordering */ for (i = 0; i < *channels; i++) s->xxch_order_tab[i] = i; s->lfe_index = *channels - 1; } s->channel_order_tab = s->xxch_order_tab; } return 0; } /** * Main frame decoding function * FIXME add arguments */ static int dca_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { AVFrame *frame = data; 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; float *src_chan; float *dst_chan; DCAContext *s = avctx->priv_data; int channels, full_channels; float scale; int achan; int chset; int mask; int j, k; int endch; int upsample = 0; s->exss_ext_mask = 0; s->xch_present = 0; s->dca_buffer_size = AVERROR_INVALIDDATA; for (i = 0; i < buf_size - 3 && s->dca_buffer_size == AVERROR_INVALIDDATA; i++) s->dca_buffer_size = avpriv_dca_convert_bitstream(buf + i, buf_size - i, 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; } 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; s->profile = FF_PROFILE_DTS; for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); 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->audio_header.prim_channels; if (s->audio_header.prim_channels + !!s->lfe > 2 && avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { /* Stereo downmix coefficients * * The decoder can only downmix to 2-channel, so we need to ensure * embedded downmix coefficients are actually targeting 2-channel. */ if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO || s->core_downmix_amode == DCA_STEREO_TOTAL)) { for (i = 0; i < num_core_channels + !!s->lfe; i++) { /* Range checked earlier */ s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]); s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]); } s->output = s->core_downmix_amode; } else { int am = s->amode & DCA_CHANNEL_MASK; if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) { av_log(s->avctx, AV_LOG_ERROR, "Invalid channel mode %d\n", am); return AVERROR_INVALIDDATA; } if (num_core_channels + !!s->lfe > FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) { avpriv_request_sample(s->avctx, "Downmixing %d channels", s->audio_header.prim_channels + !!s->lfe); return AVERROR_PATCHWELCOME; } for (i = 0; i < num_core_channels + !!s->lfe; i++) { s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0]; s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1]; } } ff_dlog(s->avctx, "Stereo downmix coeffs:\n"); for (i = 0; i < num_core_channels + !!s->lfe; i++) { ff_dlog(s->avctx, "L, input channel %d = %f\n", i, s->downmix_coef[i][0]); ff_dlog(s->avctx, "R, input channel %d = %f\n", i, s->downmix_coef[i][1]); } ff_dlog(s->avctx, "\n"); } if (s->ext_coding) s->core_ext_mask = ff_dca_ext_audio_descr_mask[s->ext_descr]; else s->core_ext_mask = 0; ret = scan_for_extensions(avctx); avctx->profile = s->profile; full_channels = channels = s->audio_header.prim_channels + !!s->lfe; ret = set_channel_layout(avctx, &channels, num_core_channels); if (ret < 0) return ret; /* get output buffer */ frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND); if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) { int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg; /* Check for invalid/unsupported conditions first */ if (s->xll_residual_channels > channels) { av_log(s->avctx, AV_LOG_WARNING, "DCA: too many residual channels (%d, core channels %d). Disabling XLL\n", s->xll_residual_channels, channels); s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL; } else if (xll_nb_samples != frame->nb_samples && 2 * frame->nb_samples != xll_nb_samples) { av_log(s->avctx, AV_LOG_WARNING, "DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n", xll_nb_samples, frame->nb_samples); s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL; } else { if (2 * frame->nb_samples == xll_nb_samples) { av_log(s->avctx, AV_LOG_INFO, "XLL: upsampling core channels by a factor of 2\n"); upsample = 1; frame->nb_samples = xll_nb_samples; // FIXME: Is it good enough to copy from the first channel set? avctx->sample_rate = s->xll_chsets[0].sampling_frequency; } /* If downmixing to stereo, don't decode additional channels. * FIXME: Using the xch_disable flag for this doesn't seem right. */ if (!s->xch_disable) channels = s->xll_channels; } } 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; } /* FIXME: This is an ugly hack, to just revert to the default * layout if we have additional channels. Need to convert the XLL * channel masks to ffmpeg channel_layout mask. */ if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels) avctx->channel_layout = 0; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; samples_flt = (float **) frame->extended_data; /* allocate buffer for extra channels if downmixing */ if (avctx->channels < full_channels) { ret = av_samples_get_buffer_size(NULL, full_channels - channels, frame->nb_samples, avctx->sample_fmt, 0); if (ret < 0) return ret; av_fast_malloc(&s->extra_channels_buffer, &s->extra_channels_buffer_size, ret); if (!s->extra_channels_buffer) return AVERROR(ENOMEM); ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL, s->extra_channels_buffer, full_channels - channels, frame->nb_samples, avctx->sample_fmt, 0); if (ret < 0) return ret; } /* filter to get final output */ for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) { int ch; unsigned block = upsample ? 512 : 256; for (ch = 0; ch < channels; ch++) s->samples_chanptr[ch] = samples_flt[ch] + i * block; for (; ch < full_channels; ch++) s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block; dca_filter_channels(s, i, upsample); /* 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_chanptr[s->channel_order_tab[s->xch_base_channel]]; float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]]; float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]]; 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 stream contains XXCH, we might need to undo an embedded downmix */ if (s->xxch_dmix_embedded) { /* Loop over channel sets in turn */ ch = num_core_channels; for (chset = 0; chset < s->xxch_chset; chset++) { endch = ch + s->xxch_chset_nch[chset]; mask = s->xxch_dmix_embedded; /* undo downmix */ for (j = ch; j < endch; j++) { if (mask & (1 << j)) { /* this channel has been mixed-out */ src_chan = s->samples_chanptr[s->channel_order_tab[j]]; for (k = 0; k < endch; k++) { achan = s->channel_order_tab[k]; scale = s->xxch_dmix_coeff[j][k]; if (scale != 0.0) { dst_chan = s->samples_chanptr[achan]; s->fdsp->vector_fmac_scalar(dst_chan, src_chan, -scale, 256); } } } } /* if a downmix has been embedded then undo the pre-scaling */ if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) { scale = s->xxch_dmix_sf[chset]; for (j = 0; j < ch; j++) { src_chan = s->samples_chanptr[s->channel_order_tab[j]]; for (k = 0; k < 256; k++) src_chan[k] *= scale; } /* LFE channel is always part of core, scale if it exists */ if (s->lfe) { src_chan = s->samples_chanptr[s->lfe_index]; for (k = 0; k < 256; k++) src_chan[k] *= scale; } } ch = endch; } } } /* update lfe history */ lfe_samples = 2 * s->lfe * (s->sample_blocks / SAMPLES_PER_SUBBAND); for (i = 0; i < 2 * s->lfe * 4; i++) s->lfe_data[i] = s->lfe_data[i + lfe_samples]; if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) { ret = ff_dca_xll_decode_audio(s, frame); if (ret < 0) return ret; } /* AVMatrixEncoding * * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */ ret = ff_side_data_update_matrix_encoding(frame, (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ? AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE); if (ret < 0) return ret; if ( avctx->profile != FF_PROFILE_DTS_HD_MA && avctx->profile != FF_PROFILE_DTS_HD_HRA) avctx->bit_rate = s->bit_rate; *got_frame_ptr = 1; 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; s->avctx = avctx; dca_init_vlcs(); s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); if (!s->fdsp) return AVERROR(ENOMEM); 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); avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; /* allow downmixing to stereo */ if (avctx->channels > 2 && avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) avctx->channels = 2; return 0; } static av_cold int dca_decode_end(AVCodecContext *avctx) { DCAContext *s = avctx->priv_data; ff_mdct_end(&s->imdct); av_freep(&s->extra_channels_buffer); av_freep(&s->fdsp); av_freep(&s->xll_sample_buf); av_freep(&s->qmf64_table); 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 }, }; static const AVOption options[] = { { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, { NULL }, }; static const AVClass dca_decoder_class = { .class_name = "DCA decoder", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, .category = AV_CLASS_CATEGORY_DECODER, }; AVCodec ff_dca_decoder = { .name = "dca", .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_DTS, .priv_data_size = sizeof(DCAContext), .init = dca_decode_init, .decode = dca_decode_frame, .close = dca_decode_end, .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .profiles = NULL_IF_CONFIG_SMALL(profiles), .priv_class = &dca_decoder_class, };