/* * AC-3 Audio Decoder * This code was developed as part of Google Summer of Code 2006. * E-AC-3 support was added as part of Google Summer of Code 2007. * * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com). * Copyright (c) 2007-2008 Bartlomiej Wolowiec * Copyright (c) 2007 Justin Ruggles * * Portions of this code are derived from liba52 * http://liba52.sourceforge.net * Copyright (C) 2000-2003 Michel Lespinasse * Copyright (C) 1999-2000 Aaron Holtzman * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public * License as published by the Free Software Foundation; either * version 2 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 * General Public License for more details. * * You should have received a copy of the GNU 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 #include "libavutil/crc.h" #include "internal.h" #include "ac3_parser.h" #include "ac3dec.h" #include "ac3dec_data.h" /** Large enough for maximum possible frame size when the specification limit is ignored */ #define AC3_FRAME_BUFFER_SIZE 32768 /** * table for ungrouping 3 values in 7 bits. * used for exponents and bap=2 mantissas */ static uint8_t ungroup_3_in_7_bits_tab[128][3]; /** tables for ungrouping mantissas */ static int b1_mantissas[32][3]; static int b2_mantissas[128][3]; static int b3_mantissas[8]; static int b4_mantissas[128][2]; static int b5_mantissas[16]; /** * Quantization table: levels for symmetric. bits for asymmetric. * reference: Table 7.18 Mapping of bap to Quantizer */ static const uint8_t quantization_tab[16] = { 0, 3, 5, 7, 11, 15, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16 }; /** dynamic range table. converts codes to scale factors. */ static float dynamic_range_tab[256]; /** Adjustments in dB gain */ #define LEVEL_PLUS_3DB 1.4142135623730950 #define LEVEL_PLUS_1POINT5DB 1.1892071150027209 #define LEVEL_MINUS_1POINT5DB 0.8408964152537145 #define LEVEL_MINUS_3DB 0.7071067811865476 #define LEVEL_MINUS_4POINT5DB 0.5946035575013605 #define LEVEL_MINUS_6DB 0.5000000000000000 #define LEVEL_MINUS_9DB 0.3535533905932738 #define LEVEL_ZERO 0.0000000000000000 #define LEVEL_ONE 1.0000000000000000 static const float gain_levels[9] = { LEVEL_PLUS_3DB, LEVEL_PLUS_1POINT5DB, LEVEL_ONE, LEVEL_MINUS_1POINT5DB, LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB, LEVEL_ZERO, LEVEL_MINUS_9DB }; /** * Table for center mix levels * reference: Section 5.4.2.4 cmixlev */ static const uint8_t center_levels[4] = { 4, 5, 6, 5 }; /** * Table for surround mix levels * reference: Section 5.4.2.5 surmixlev */ static const uint8_t surround_levels[4] = { 4, 6, 7, 6 }; /** * Table for default stereo downmixing coefficients * reference: Section 7.8.2 Downmixing Into Two Channels */ static const uint8_t ac3_default_coeffs[8][5][2] = { { { 2, 7 }, { 7, 2 }, }, { { 4, 4 }, }, { { 2, 7 }, { 7, 2 }, }, { { 2, 7 }, { 5, 5 }, { 7, 2 }, }, { { 2, 7 }, { 7, 2 }, { 6, 6 }, }, { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 8, 8 }, }, { { 2, 7 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, }, { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, }, }; /** * Symmetrical Dequantization * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization * Tables 7.19 to 7.23 */ static inline int symmetric_dequant(int code, int levels) { return ((code - (levels >> 1)) << 24) / levels; } /* * Initialize tables at runtime. */ static av_cold void ac3_tables_init(void) { int i; /* generate table for ungrouping 3 values in 7 bits reference: Section 7.1.3 Exponent Decoding */ for(i=0; i<128; i++) { ungroup_3_in_7_bits_tab[i][0] = i / 25; ungroup_3_in_7_bits_tab[i][1] = (i % 25) / 5; ungroup_3_in_7_bits_tab[i][2] = (i % 25) % 5; } /* generate grouped mantissa tables reference: Section 7.3.5 Ungrouping of Mantissas */ for(i=0; i<32; i++) { /* bap=1 mantissas */ b1_mantissas[i][0] = symmetric_dequant(ff_ac3_ungroup_3_in_5_bits_tab[i][0], 3); b1_mantissas[i][1] = symmetric_dequant(ff_ac3_ungroup_3_in_5_bits_tab[i][1], 3); b1_mantissas[i][2] = symmetric_dequant(ff_ac3_ungroup_3_in_5_bits_tab[i][2], 3); } for(i=0; i<128; i++) { /* bap=2 mantissas */ b2_mantissas[i][0] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][0], 5); b2_mantissas[i][1] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][1], 5); b2_mantissas[i][2] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][2], 5); /* bap=4 mantissas */ b4_mantissas[i][0] = symmetric_dequant(i / 11, 11); b4_mantissas[i][1] = symmetric_dequant(i % 11, 11); } /* generate ungrouped mantissa tables reference: Tables 7.21 and 7.23 */ for(i=0; i<7; i++) { /* bap=3 mantissas */ b3_mantissas[i] = symmetric_dequant(i, 7); } for(i=0; i<15; i++) { /* bap=5 mantissas */ b5_mantissas[i] = symmetric_dequant(i, 15); } /* generate dynamic range table reference: Section 7.7.1 Dynamic Range Control */ for(i=0; i<256; i++) { int v = (i >> 5) - ((i >> 7) << 3) - 5; dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20); } } /** * AVCodec initialization */ static av_cold int ac3_decode_init(AVCodecContext *avctx) { AC3DecodeContext *s = avctx->priv_data; s->avctx = avctx; ac3_common_init(); ac3_tables_init(); ff_mdct_init(&s->imdct_256, 8, 1); ff_mdct_init(&s->imdct_512, 9, 1); ff_kbd_window_init(s->window, 5.0, 256); dsputil_init(&s->dsp, avctx); av_lfg_init(&s->dith_state, 0); /* set bias values for float to int16 conversion */ if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { s->add_bias = 385.0f; s->mul_bias = 1.0f; } else { s->add_bias = 0.0f; s->mul_bias = 32767.0f; } /* allow downmixing to stereo or mono */ if (avctx->channels > 0 && avctx->request_channels > 0 && avctx->request_channels < avctx->channels && avctx->request_channels <= 2) { avctx->channels = avctx->request_channels; } s->downmixed = 1; /* allocate context input buffer */ if (avctx->error_recognition >= FF_ER_CAREFUL) { s->input_buffer = av_mallocz(AC3_FRAME_BUFFER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); if (!s->input_buffer) return AVERROR_NOMEM; } avctx->sample_fmt = SAMPLE_FMT_S16; return 0; } /** * Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream. * GetBitContext within AC3DecodeContext must point to * the start of the synchronized AC-3 bitstream. */ static int ac3_parse_header(AC3DecodeContext *s) { GetBitContext *gbc = &s->gbc; int i; /* read the rest of the bsi. read twice for dual mono mode. */ i = !(s->channel_mode); do { skip_bits(gbc, 5); // skip dialog normalization if (get_bits1(gbc)) skip_bits(gbc, 8); //skip compression if (get_bits1(gbc)) skip_bits(gbc, 8); //skip language code if (get_bits1(gbc)) skip_bits(gbc, 7); //skip audio production information } while (i--); skip_bits(gbc, 2); //skip copyright bit and original bitstream bit /* skip the timecodes (or extra bitstream information for Alternate Syntax) TODO: read & use the xbsi1 downmix levels */ if (get_bits1(gbc)) skip_bits(gbc, 14); //skip timecode1 / xbsi1 if (get_bits1(gbc)) skip_bits(gbc, 14); //skip timecode2 / xbsi2 /* skip additional bitstream info */ if (get_bits1(gbc)) { i = get_bits(gbc, 6); do { skip_bits(gbc, 8); } while(i--); } return 0; } /** * Common function to parse AC-3 or E-AC-3 frame header */ static int parse_frame_header(AC3DecodeContext *s) { AC3HeaderInfo hdr; int err; err = ff_ac3_parse_header(&s->gbc, &hdr); if(err) return err; /* get decoding parameters from header info */ s->bit_alloc_params.sr_code = hdr.sr_code; s->channel_mode = hdr.channel_mode; s->lfe_on = hdr.lfe_on; s->bit_alloc_params.sr_shift = hdr.sr_shift; s->sample_rate = hdr.sample_rate; s->bit_rate = hdr.bit_rate; s->channels = hdr.channels; s->fbw_channels = s->channels - s->lfe_on; s->lfe_ch = s->fbw_channels + 1; s->frame_size = hdr.frame_size; s->center_mix_level = hdr.center_mix_level; s->surround_mix_level = hdr.surround_mix_level; s->num_blocks = hdr.num_blocks; s->frame_type = hdr.frame_type; s->substreamid = hdr.substreamid; if(s->lfe_on) { s->start_freq[s->lfe_ch] = 0; s->end_freq[s->lfe_ch] = 7; s->num_exp_groups[s->lfe_ch] = 2; s->channel_in_cpl[s->lfe_ch] = 0; } if (hdr.bitstream_id <= 10) { s->eac3 = 0; s->snr_offset_strategy = 2; s->block_switch_syntax = 1; s->dither_flag_syntax = 1; s->bit_allocation_syntax = 1; s->fast_gain_syntax = 0; s->first_cpl_leak = 0; s->dba_syntax = 1; s->skip_syntax = 1; memset(s->channel_uses_aht, 0, sizeof(s->channel_uses_aht)); return ac3_parse_header(s); } else { s->eac3 = 1; return ff_eac3_parse_header(s); } } /** * Set stereo downmixing coefficients based on frame header info. * reference: Section 7.8.2 Downmixing Into Two Channels */ static void set_downmix_coeffs(AC3DecodeContext *s) { int i; float cmix = gain_levels[center_levels[s->center_mix_level]]; float smix = gain_levels[surround_levels[s->surround_mix_level]]; float norm0, norm1; for(i=0; ifbw_channels; i++) { s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]]; s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]]; } if(s->channel_mode > 1 && s->channel_mode & 1) { s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix; } if(s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) { int nf = s->channel_mode - 2; s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB; } if(s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) { int nf = s->channel_mode - 4; s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix; } /* renormalize */ norm0 = norm1 = 0.0; for(i=0; ifbw_channels; i++) { norm0 += s->downmix_coeffs[i][0]; norm1 += s->downmix_coeffs[i][1]; } norm0 = 1.0f / norm0; norm1 = 1.0f / norm1; for(i=0; ifbw_channels; i++) { s->downmix_coeffs[i][0] *= norm0; s->downmix_coeffs[i][1] *= norm1; } if(s->output_mode == AC3_CHMODE_MONO) { for(i=0; ifbw_channels; i++) s->downmix_coeffs[i][0] = (s->downmix_coeffs[i][0] + s->downmix_coeffs[i][1]) * LEVEL_MINUS_3DB; } } /** * Decode the grouped exponents according to exponent strategy. * reference: Section 7.1.3 Exponent Decoding */ static int decode_exponents(GetBitContext *gbc, int exp_strategy, int ngrps, uint8_t absexp, int8_t *dexps) { int i, j, grp, group_size; int dexp[256]; int expacc, prevexp; /* unpack groups */ group_size = exp_strategy + (exp_strategy == EXP_D45); for(grp=0,i=0; grp 24U) return -1; switch (group_size) { case 4: dexps[j++] = prevexp; dexps[j++] = prevexp; case 2: dexps[j++] = prevexp; case 1: dexps[j++] = prevexp; } } return 0; } /** * Generate transform coefficients for each coupled channel in the coupling * range using the coupling coefficients and coupling coordinates. * reference: Section 7.4.3 Coupling Coordinate Format */ static void calc_transform_coeffs_cpl(AC3DecodeContext *s) { int i, j, ch, bnd, subbnd; subbnd = -1; i = s->start_freq[CPL_CH]; for(bnd=0; bndnum_cpl_bands; bnd++) { do { subbnd++; for(j=0; j<12; j++) { for(ch=1; ch<=s->fbw_channels; ch++) { if(s->channel_in_cpl[ch]) { s->fixed_coeffs[ch][i] = ((int64_t)s->fixed_coeffs[CPL_CH][i] * (int64_t)s->cpl_coords[ch][bnd]) >> 23; if (ch == 2 && s->phase_flags[bnd]) s->fixed_coeffs[ch][i] = -s->fixed_coeffs[ch][i]; } } i++; } } while(s->cpl_band_struct[subbnd]); } } /** * Grouped mantissas for 3-level 5-level and 11-level quantization */ typedef struct { int b1_mant[3]; int b2_mant[3]; int b4_mant[2]; int b1ptr; int b2ptr; int b4ptr; } mant_groups; /** * Decode the transform coefficients for a particular channel * reference: Section 7.3 Quantization and Decoding of Mantissas */ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m) { GetBitContext *gbc = &s->gbc; int i, gcode, tbap, start, end; uint8_t *exps; uint8_t *bap; int *coeffs; exps = s->dexps[ch_index]; bap = s->bap[ch_index]; coeffs = s->fixed_coeffs[ch_index]; start = s->start_freq[ch_index]; end = s->end_freq[ch_index]; for (i = start; i < end; i++) { tbap = bap[i]; switch (tbap) { case 0: coeffs[i] = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000; break; case 1: if(m->b1ptr > 2) { gcode = get_bits(gbc, 5); m->b1_mant[0] = b1_mantissas[gcode][0]; m->b1_mant[1] = b1_mantissas[gcode][1]; m->b1_mant[2] = b1_mantissas[gcode][2]; m->b1ptr = 0; } coeffs[i] = m->b1_mant[m->b1ptr++]; break; case 2: if(m->b2ptr > 2) { gcode = get_bits(gbc, 7); m->b2_mant[0] = b2_mantissas[gcode][0]; m->b2_mant[1] = b2_mantissas[gcode][1]; m->b2_mant[2] = b2_mantissas[gcode][2]; m->b2ptr = 0; } coeffs[i] = m->b2_mant[m->b2ptr++]; break; case 3: coeffs[i] = b3_mantissas[get_bits(gbc, 3)]; break; case 4: if(m->b4ptr > 1) { gcode = get_bits(gbc, 7); m->b4_mant[0] = b4_mantissas[gcode][0]; m->b4_mant[1] = b4_mantissas[gcode][1]; m->b4ptr = 0; } coeffs[i] = m->b4_mant[m->b4ptr++]; break; case 5: coeffs[i] = b5_mantissas[get_bits(gbc, 4)]; break; default: { /* asymmetric dequantization */ int qlevel = quantization_tab[tbap]; coeffs[i] = get_sbits(gbc, qlevel) << (24 - qlevel); break; } } coeffs[i] >>= exps[i]; } } /** * Remove random dithering from coefficients with zero-bit mantissas * reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0) */ static void remove_dithering(AC3DecodeContext *s) { int ch, i; int end=0; int *coeffs; uint8_t *bap; for(ch=1; ch<=s->fbw_channels; ch++) { if(!s->dither_flag[ch]) { coeffs = s->fixed_coeffs[ch]; bap = s->bap[ch]; if(s->channel_in_cpl[ch]) end = s->start_freq[CPL_CH]; else end = s->end_freq[ch]; for(i=0; ichannel_in_cpl[ch]) { bap = s->bap[CPL_CH]; for(; iend_freq[CPL_CH]; i++) { if(!bap[i]) coeffs[i] = 0; } } } } } static void decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch, mant_groups *m) { if (!s->channel_uses_aht[ch]) { ac3_decode_transform_coeffs_ch(s, ch, m); } else { /* if AHT is used, mantissas for all blocks are encoded in the first block of the frame. */ int bin; if (!blk) ff_eac3_decode_transform_coeffs_aht_ch(s, ch); for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) { s->fixed_coeffs[ch][bin] = s->pre_mantissa[ch][bin][blk] >> s->dexps[ch][bin]; } } } /** * Decode the transform coefficients. */ static void decode_transform_coeffs(AC3DecodeContext *s, int blk) { int ch, end; int got_cplchan = 0; mant_groups m; m.b1ptr = m.b2ptr = m.b4ptr = 3; for (ch = 1; ch <= s->channels; ch++) { /* transform coefficients for full-bandwidth channel */ decode_transform_coeffs_ch(s, blk, ch, &m); /* tranform coefficients for coupling channel come right after the coefficients for the first coupled channel*/ if (s->channel_in_cpl[ch]) { if (!got_cplchan) { decode_transform_coeffs_ch(s, blk, CPL_CH, &m); calc_transform_coeffs_cpl(s); got_cplchan = 1; } end = s->end_freq[CPL_CH]; } else { end = s->end_freq[ch]; } do s->fixed_coeffs[ch][end] = 0; while(++end < 256); } /* zero the dithered coefficients for appropriate channels */ remove_dithering(s); } /** * Stereo rematrixing. * reference: Section 7.5.4 Rematrixing : Decoding Technique */ static void do_rematrixing(AC3DecodeContext *s) { int bnd, i; int end, bndend; int tmp0, tmp1; end = FFMIN(s->end_freq[1], s->end_freq[2]); for(bnd=0; bndnum_rematrixing_bands; bnd++) { if(s->rematrixing_flags[bnd]) { bndend = FFMIN(end, ff_ac3_rematrix_band_tab[bnd+1]); for(i=ff_ac3_rematrix_band_tab[bnd]; ifixed_coeffs[1][i]; tmp1 = s->fixed_coeffs[2][i]; s->fixed_coeffs[1][i] = tmp0 + tmp1; s->fixed_coeffs[2][i] = tmp0 - tmp1; } } } } /** * Inverse MDCT Transform. * Convert frequency domain coefficients to time-domain audio samples. * reference: Section 7.9.4 Transformation Equations */ static inline void do_imdct(AC3DecodeContext *s, int channels) { int ch; float add_bias = s->add_bias; if(s->out_channels==1 && channels>1) add_bias *= LEVEL_MINUS_3DB; // compensate for the gain in downmix for (ch=1; ch<=channels; ch++) { if (s->block_switch[ch]) { int i; float *x = s->tmp_output+128; for(i=0; i<128; i++) x[i] = s->transform_coeffs[ch][2*i]; ff_imdct_half(&s->imdct_256, s->tmp_output, x); s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, add_bias, 128); for(i=0; i<128; i++) x[i] = s->transform_coeffs[ch][2*i+1]; ff_imdct_half(&s->imdct_256, s->delay[ch-1], x); } else { ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]); s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, add_bias, 128); memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float)); } } } /** * Downmix the output to mono or stereo. */ void ff_ac3_downmix_c(float (*samples)[256], float (*matrix)[2], int out_ch, int in_ch, int len) { int i, j; float v0, v1; if(out_ch == 2) { for(i=0; idelay[0]); switch(s->channel_mode) { case AC3_CHMODE_DUALMONO: case AC3_CHMODE_STEREO: /* upmix mono to stereo */ memcpy(s->delay[1], s->delay[0], channel_data_size); break; case AC3_CHMODE_2F2R: memset(s->delay[3], 0, channel_data_size); case AC3_CHMODE_2F1R: memset(s->delay[2], 0, channel_data_size); break; case AC3_CHMODE_3F2R: memset(s->delay[4], 0, channel_data_size); case AC3_CHMODE_3F1R: memset(s->delay[3], 0, channel_data_size); case AC3_CHMODE_3F: memcpy(s->delay[2], s->delay[1], channel_data_size); memset(s->delay[1], 0, channel_data_size); break; } } /** * Decode band structure for coupling, spectral extension, or enhanced coupling. * @param[in] gbc bit reader context * @param[in] blk block number * @param[in] eac3 flag to indicate E-AC-3 * @param[in] ecpl flag to indicate enhanced coupling * @param[in] start_subband subband number for start of range * @param[in] end_subband subband number for end of range * @param[in] default_band_struct default band structure table * @param[out] band_struct decoded band structure * @param[out] num_subbands number of subbands (optionally NULL) * @param[out] num_bands number of bands (optionally NULL) * @param[out] band_sizes array containing the number of bins in each band (optionally NULL) */ static void decode_band_structure(GetBitContext *gbc, int blk, int eac3, int ecpl, int start_subband, int end_subband, const uint8_t *default_band_struct, uint8_t *band_struct, int *num_subbands, int *num_bands, uint8_t *band_sizes) { int subbnd, bnd, n_subbands, n_bands=0; uint8_t bnd_sz[22]; n_subbands = end_subband - start_subband; /* decode band structure from bitstream or use default */ if (!eac3 || get_bits1(gbc)) { for (subbnd = 0; subbnd < n_subbands - 1; subbnd++) { band_struct[subbnd] = get_bits1(gbc); } } else if (!blk) { memcpy(band_struct, &default_band_struct[start_subband+1], n_subbands-1); } band_struct[n_subbands-1] = 0; /* calculate number of bands and band sizes based on band structure. note that the first 4 subbands in enhanced coupling span only 6 bins instead of 12. */ if (num_bands || band_sizes ) { n_bands = n_subbands; bnd_sz[0] = ecpl ? 6 : 12; for (bnd = 0, subbnd = 1; subbnd < n_subbands; subbnd++) { int subbnd_size = (ecpl && subbnd < 4) ? 6 : 12; if (band_struct[subbnd-1]) { n_bands--; bnd_sz[bnd] += subbnd_size; } else { bnd_sz[++bnd] = subbnd_size; } } } /* set optional output params */ if (num_subbands) *num_subbands = n_subbands; if (num_bands) *num_bands = n_bands; if (band_sizes) memcpy(band_sizes, bnd_sz, n_bands); } /** * Decode a single audio block from the AC-3 bitstream. */ static int decode_audio_block(AC3DecodeContext *s, int blk) { int fbw_channels = s->fbw_channels; int channel_mode = s->channel_mode; int i, bnd, seg, ch; int different_transforms; int downmix_output; int cpl_in_use; GetBitContext *gbc = &s->gbc; uint8_t bit_alloc_stages[AC3_MAX_CHANNELS]; memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS); /* block switch flags */ different_transforms = 0; if (s->block_switch_syntax) { for (ch = 1; ch <= fbw_channels; ch++) { s->block_switch[ch] = get_bits1(gbc); if(ch > 1 && s->block_switch[ch] != s->block_switch[1]) different_transforms = 1; } } /* dithering flags */ if (s->dither_flag_syntax) { for (ch = 1; ch <= fbw_channels; ch++) { s->dither_flag[ch] = get_bits1(gbc); } } /* dynamic range */ i = !(s->channel_mode); do { if(get_bits1(gbc)) { s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)]-1.0) * s->avctx->drc_scale)+1.0; } else if(blk == 0) { s->dynamic_range[i] = 1.0f; } } while(i--); /* spectral extension strategy */ if (s->eac3 && (!blk || get_bits1(gbc))) { if (get_bits1(gbc)) { ff_log_missing_feature(s->avctx, "Spectral extension", 1); return -1; } /* TODO: parse spectral extension strategy info */ } /* TODO: spectral extension coordinates */ /* coupling strategy */ if (s->eac3 ? s->cpl_strategy_exists[blk] : get_bits1(gbc)) { memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); if (!s->eac3) s->cpl_in_use[blk] = get_bits1(gbc); if (s->cpl_in_use[blk]) { /* coupling in use */ int cpl_start_subband, cpl_end_subband; if (channel_mode < AC3_CHMODE_STEREO) { av_log(s->avctx, AV_LOG_ERROR, "coupling not allowed in mono or dual-mono\n"); return -1; } /* check for enhanced coupling */ if (s->eac3 && get_bits1(gbc)) { /* TODO: parse enhanced coupling strategy info */ ff_log_missing_feature(s->avctx, "Enhanced coupling", 1); return -1; } /* determine which channels are coupled */ if (s->eac3 && s->channel_mode == AC3_CHMODE_STEREO) { s->channel_in_cpl[1] = 1; s->channel_in_cpl[2] = 1; } else { for (ch = 1; ch <= fbw_channels; ch++) s->channel_in_cpl[ch] = get_bits1(gbc); } /* phase flags in use */ if (channel_mode == AC3_CHMODE_STEREO) s->phase_flags_in_use = get_bits1(gbc); /* coupling frequency range */ /* TODO: modify coupling end freq if spectral extension is used */ cpl_start_subband = get_bits(gbc, 4); cpl_end_subband = get_bits(gbc, 4) + 3; s->num_cpl_subbands = cpl_end_subband - cpl_start_subband; if (s->num_cpl_subbands < 0) { av_log(s->avctx, AV_LOG_ERROR, "invalid coupling range (%d > %d)\n", cpl_start_subband, cpl_end_subband); return -1; } s->start_freq[CPL_CH] = cpl_start_subband * 12 + 37; s->end_freq[CPL_CH] = cpl_end_subband * 12 + 37; decode_band_structure(gbc, blk, s->eac3, 0, cpl_start_subband, cpl_end_subband, ff_eac3_default_cpl_band_struct, s->cpl_band_struct, &s->num_cpl_subbands, &s->num_cpl_bands, NULL); } else { /* coupling not in use */ for (ch = 1; ch <= fbw_channels; ch++) { s->channel_in_cpl[ch] = 0; s->first_cpl_coords[ch] = 1; } s->first_cpl_leak = s->eac3; s->phase_flags_in_use = 0; } } else if (!s->eac3) { if(!blk) { av_log(s->avctx, AV_LOG_ERROR, "new coupling strategy must be present in block 0\n"); return -1; } else { s->cpl_in_use[blk] = s->cpl_in_use[blk-1]; } } cpl_in_use = s->cpl_in_use[blk]; /* coupling coordinates */ if (cpl_in_use) { int cpl_coords_exist = 0; for (ch = 1; ch <= fbw_channels; ch++) { if (s->channel_in_cpl[ch]) { if ((s->eac3 && s->first_cpl_coords[ch]) || get_bits1(gbc)) { int master_cpl_coord, cpl_coord_exp, cpl_coord_mant; s->first_cpl_coords[ch] = 0; cpl_coords_exist = 1; master_cpl_coord = 3 * get_bits(gbc, 2); for (bnd = 0; bnd < s->num_cpl_bands; bnd++) { cpl_coord_exp = get_bits(gbc, 4); cpl_coord_mant = get_bits(gbc, 4); if (cpl_coord_exp == 15) s->cpl_coords[ch][bnd] = cpl_coord_mant << 22; else s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16) << 21; s->cpl_coords[ch][bnd] >>= (cpl_coord_exp + master_cpl_coord); } } else if (!blk) { av_log(s->avctx, AV_LOG_ERROR, "new coupling coordinates must be present in block 0\n"); return -1; } } else { /* channel not in coupling */ s->first_cpl_coords[ch] = 1; } } /* phase flags */ if (channel_mode == AC3_CHMODE_STEREO && cpl_coords_exist) { for (bnd = 0; bnd < s->num_cpl_bands; bnd++) { s->phase_flags[bnd] = s->phase_flags_in_use? get_bits1(gbc) : 0; } } } /* stereo rematrixing strategy and band structure */ if (channel_mode == AC3_CHMODE_STEREO) { if ((s->eac3 && !blk) || get_bits1(gbc)) { s->num_rematrixing_bands = 4; if(cpl_in_use && s->start_freq[CPL_CH] <= 61) s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37); for(bnd=0; bndnum_rematrixing_bands; bnd++) s->rematrixing_flags[bnd] = get_bits1(gbc); } else if (!blk) { av_log(s->avctx, AV_LOG_ERROR, "new rematrixing strategy must be present in block 0\n"); return -1; } } /* exponent strategies for each channel */ for (ch = !cpl_in_use; ch <= s->channels; ch++) { if (!s->eac3) s->exp_strategy[blk][ch] = get_bits(gbc, 2 - (ch == s->lfe_ch)); if(s->exp_strategy[blk][ch] != EXP_REUSE) bit_alloc_stages[ch] = 3; } /* channel bandwidth */ for (ch = 1; ch <= fbw_channels; ch++) { s->start_freq[ch] = 0; if (s->exp_strategy[blk][ch] != EXP_REUSE) { int group_size; int prev = s->end_freq[ch]; if (s->channel_in_cpl[ch]) s->end_freq[ch] = s->start_freq[CPL_CH]; else { int bandwidth_code = get_bits(gbc, 6); if (bandwidth_code > 60) { av_log(s->avctx, AV_LOG_ERROR, "bandwidth code = %d > 60\n", bandwidth_code); return -1; } s->end_freq[ch] = bandwidth_code * 3 + 73; } group_size = 3 << (s->exp_strategy[blk][ch] - 1); s->num_exp_groups[ch] = (s->end_freq[ch]+group_size-4) / group_size; if(blk > 0 && s->end_freq[ch] != prev) memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); } } if (cpl_in_use && s->exp_strategy[blk][CPL_CH] != EXP_REUSE) { s->num_exp_groups[CPL_CH] = (s->end_freq[CPL_CH] - s->start_freq[CPL_CH]) / (3 << (s->exp_strategy[blk][CPL_CH] - 1)); } /* decode exponents for each channel */ for (ch = !cpl_in_use; ch <= s->channels; ch++) { if (s->exp_strategy[blk][ch] != EXP_REUSE) { s->dexps[ch][0] = get_bits(gbc, 4) << !ch; if (decode_exponents(gbc, s->exp_strategy[blk][ch], s->num_exp_groups[ch], s->dexps[ch][0], &s->dexps[ch][s->start_freq[ch]+!!ch])) { av_log(s->avctx, AV_LOG_ERROR, "exponent out-of-range\n"); return -1; } if(ch != CPL_CH && ch != s->lfe_ch) skip_bits(gbc, 2); /* skip gainrng */ } } /* bit allocation information */ if (s->bit_allocation_syntax) { if (get_bits1(gbc)) { s->bit_alloc_params.slow_decay = ff_ac3_slow_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift; s->bit_alloc_params.fast_decay = ff_ac3_fast_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift; s->bit_alloc_params.slow_gain = ff_ac3_slow_gain_tab[get_bits(gbc, 2)]; s->bit_alloc_params.db_per_bit = ff_ac3_db_per_bit_tab[get_bits(gbc, 2)]; s->bit_alloc_params.floor = ff_ac3_floor_tab[get_bits(gbc, 3)]; for(ch=!cpl_in_use; ch<=s->channels; ch++) bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2); } else if (!blk) { av_log(s->avctx, AV_LOG_ERROR, "new bit allocation info must be present in block 0\n"); return -1; } } /* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */ if(!s->eac3 || !blk){ if(s->snr_offset_strategy && get_bits1(gbc)) { int snr = 0; int csnr; csnr = (get_bits(gbc, 6) - 15) << 4; for (i = ch = !cpl_in_use; ch <= s->channels; ch++) { /* snr offset */ if (ch == i || s->snr_offset_strategy == 2) snr = (csnr + get_bits(gbc, 4)) << 2; /* run at least last bit allocation stage if snr offset changes */ if(blk && s->snr_offset[ch] != snr) { bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 1); } s->snr_offset[ch] = snr; /* fast gain (normal AC-3 only) */ if (!s->eac3) { int prev = s->fast_gain[ch]; s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)]; /* run last 2 bit allocation stages if fast gain changes */ if(blk && prev != s->fast_gain[ch]) bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2); } } } else if (!s->eac3 && !blk) { av_log(s->avctx, AV_LOG_ERROR, "new snr offsets must be present in block 0\n"); return -1; } } /* fast gain (E-AC-3 only) */ if (s->fast_gain_syntax && get_bits1(gbc)) { for (ch = !cpl_in_use; ch <= s->channels; ch++) { int prev = s->fast_gain[ch]; s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)]; /* run last 2 bit allocation stages if fast gain changes */ if(blk && prev != s->fast_gain[ch]) bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2); } } else if (s->eac3 && !blk) { for (ch = !cpl_in_use; ch <= s->channels; ch++) s->fast_gain[ch] = ff_ac3_fast_gain_tab[4]; } /* E-AC-3 to AC-3 converter SNR offset */ if (s->frame_type == EAC3_FRAME_TYPE_INDEPENDENT && get_bits1(gbc)) { skip_bits(gbc, 10); // skip converter snr offset } /* coupling leak information */ if (cpl_in_use) { if (s->first_cpl_leak || get_bits1(gbc)) { int fl = get_bits(gbc, 3); int sl = get_bits(gbc, 3); /* run last 2 bit allocation stages for coupling channel if coupling leak changes */ if(blk && (fl != s->bit_alloc_params.cpl_fast_leak || sl != s->bit_alloc_params.cpl_slow_leak)) { bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2); } s->bit_alloc_params.cpl_fast_leak = fl; s->bit_alloc_params.cpl_slow_leak = sl; } else if (!s->eac3 && !blk) { av_log(s->avctx, AV_LOG_ERROR, "new coupling leak info must be present in block 0\n"); return -1; } s->first_cpl_leak = 0; } /* delta bit allocation information */ if (s->dba_syntax && get_bits1(gbc)) { /* delta bit allocation exists (strategy) */ for (ch = !cpl_in_use; ch <= fbw_channels; ch++) { s->dba_mode[ch] = get_bits(gbc, 2); if (s->dba_mode[ch] == DBA_RESERVED) { av_log(s->avctx, AV_LOG_ERROR, "delta bit allocation strategy reserved\n"); return -1; } bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2); } /* channel delta offset, len and bit allocation */ for (ch = !cpl_in_use; ch <= fbw_channels; ch++) { if (s->dba_mode[ch] == DBA_NEW) { s->dba_nsegs[ch] = get_bits(gbc, 3); for (seg = 0; seg <= s->dba_nsegs[ch]; seg++) { s->dba_offsets[ch][seg] = get_bits(gbc, 5); s->dba_lengths[ch][seg] = get_bits(gbc, 4); s->dba_values[ch][seg] = get_bits(gbc, 3); } /* run last 2 bit allocation stages if new dba values */ bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2); } } } else if(blk == 0) { for(ch=0; ch<=s->channels; ch++) { s->dba_mode[ch] = DBA_NONE; } } /* Bit allocation */ for(ch=!cpl_in_use; ch<=s->channels; ch++) { if(bit_alloc_stages[ch] > 2) { /* Exponent mapping into PSD and PSD integration */ ff_ac3_bit_alloc_calc_psd(s->dexps[ch], s->start_freq[ch], s->end_freq[ch], s->psd[ch], s->band_psd[ch]); } if(bit_alloc_stages[ch] > 1) { /* Compute excitation function, Compute masking curve, and Apply delta bit allocation */ if (ff_ac3_bit_alloc_calc_mask(&s->bit_alloc_params, s->band_psd[ch], s->start_freq[ch], s->end_freq[ch], s->fast_gain[ch], (ch == s->lfe_ch), s->dba_mode[ch], s->dba_nsegs[ch], s->dba_offsets[ch], s->dba_lengths[ch], s->dba_values[ch], s->mask[ch])) { av_log(s->avctx, AV_LOG_ERROR, "error in bit allocation\n"); return -1; } } if(bit_alloc_stages[ch] > 0) { /* Compute bit allocation */ const uint8_t *bap_tab = s->channel_uses_aht[ch] ? ff_eac3_hebap_tab : ff_ac3_bap_tab; ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch], s->start_freq[ch], s->end_freq[ch], s->snr_offset[ch], s->bit_alloc_params.floor, bap_tab, s->bap[ch]); } } /* unused dummy data */ if (s->skip_syntax && get_bits1(gbc)) { int skipl = get_bits(gbc, 9); while(skipl--) skip_bits(gbc, 8); } /* unpack the transform coefficients this also uncouples channels if coupling is in use. */ decode_transform_coeffs(s, blk); /* TODO: generate enhanced coupling coordinates and uncouple */ /* TODO: apply spectral extension */ /* recover coefficients if rematrixing is in use */ if(s->channel_mode == AC3_CHMODE_STEREO) do_rematrixing(s); /* apply scaling to coefficients (headroom, dynrng) */ for(ch=1; ch<=s->channels; ch++) { float gain = s->mul_bias / 4194304.0f; if(s->channel_mode == AC3_CHMODE_DUALMONO) { gain *= s->dynamic_range[ch-1]; } else { gain *= s->dynamic_range[0]; } s->dsp.int32_to_float_fmul_scalar(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256); } /* downmix and MDCT. order depends on whether block switching is used for any channel in this block. this is because coefficients for the long and short transforms cannot be mixed. */ downmix_output = s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) && s->fbw_channels == s->out_channels); if(different_transforms) { /* the delay samples have already been downmixed, so we upmix the delay samples in order to reconstruct all channels before downmixing. */ if(s->downmixed) { s->downmixed = 0; ac3_upmix_delay(s); } do_imdct(s, s->channels); if(downmix_output) { s->dsp.ac3_downmix(s->output, s->downmix_coeffs, s->out_channels, s->fbw_channels, 256); } } else { if(downmix_output) { s->dsp.ac3_downmix(s->transform_coeffs+1, s->downmix_coeffs, s->out_channels, s->fbw_channels, 256); } if(downmix_output && !s->downmixed) { s->downmixed = 1; s->dsp.ac3_downmix(s->delay, s->downmix_coeffs, s->out_channels, s->fbw_channels, 128); } do_imdct(s, s->out_channels); } return 0; } /** * Decode a single AC-3 frame. */ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, const uint8_t *buf, int buf_size) { AC3DecodeContext *s = avctx->priv_data; int16_t *out_samples = (int16_t *)data; int blk, ch, err; /* initialize the GetBitContext with the start of valid AC-3 Frame */ if (s->input_buffer) { /* copy input buffer to decoder context to avoid reading past the end of the buffer, which can be caused by a damaged input stream. */ memcpy(s->input_buffer, buf, FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE)); init_get_bits(&s->gbc, s->input_buffer, buf_size * 8); } else { init_get_bits(&s->gbc, buf, buf_size * 8); } /* parse the syncinfo */ *data_size = 0; err = parse_frame_header(s); /* check that reported frame size fits in input buffer */ if(s->frame_size > buf_size) { av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); err = AC3_PARSE_ERROR_FRAME_SIZE; } /* check for crc mismatch */ if(err != AC3_PARSE_ERROR_FRAME_SIZE && avctx->error_recognition >= FF_ER_CAREFUL) { if(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) { av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n"); err = AC3_PARSE_ERROR_CRC; } } if(err && err != AC3_PARSE_ERROR_CRC) { switch(err) { case AC3_PARSE_ERROR_SYNC: av_log(avctx, AV_LOG_ERROR, "frame sync error\n"); return -1; case AC3_PARSE_ERROR_BSID: av_log(avctx, AV_LOG_ERROR, "invalid bitstream id\n"); break; case AC3_PARSE_ERROR_SAMPLE_RATE: av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n"); break; case AC3_PARSE_ERROR_FRAME_SIZE: av_log(avctx, AV_LOG_ERROR, "invalid frame size\n"); break; case AC3_PARSE_ERROR_FRAME_TYPE: /* skip frame if CRC is ok. otherwise use error concealment. */ /* TODO: add support for substreams and dependent frames */ if(s->frame_type == EAC3_FRAME_TYPE_DEPENDENT || s->substreamid) { av_log(avctx, AV_LOG_ERROR, "unsupported frame type : skipping frame\n"); return s->frame_size; } else { av_log(avctx, AV_LOG_ERROR, "invalid frame type\n"); } break; default: av_log(avctx, AV_LOG_ERROR, "invalid header\n"); break; } } /* if frame is ok, set audio parameters */ if (!err) { avctx->sample_rate = s->sample_rate; avctx->bit_rate = s->bit_rate; /* channel config */ s->out_channels = s->channels; s->output_mode = s->channel_mode; if(s->lfe_on) s->output_mode |= AC3_OUTPUT_LFEON; if (avctx->request_channels > 0 && avctx->request_channels <= 2 && avctx->request_channels < s->channels) { s->out_channels = avctx->request_channels; s->output_mode = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO; } avctx->channels = s->out_channels; /* set downmixing coefficients if needed */ if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) && s->fbw_channels == s->out_channels)) { set_downmix_coeffs(s); } } else if (!s->out_channels) { s->out_channels = avctx->channels; if(s->out_channels < s->channels) s->output_mode = s->out_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO; } /* decode the audio blocks */ for (blk = 0; blk < s->num_blocks; blk++) { const float *output[s->out_channels]; if (!err && decode_audio_block(s, blk)) { av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n"); err = 1; } for (ch = 0; ch < s->out_channels; ch++) output[ch] = s->output[ch]; s->dsp.float_to_int16_interleave(out_samples, output, 256, s->out_channels); out_samples += 256 * s->out_channels; } *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t); return s->frame_size; } /** * Uninitialize the AC-3 decoder. */ static av_cold int ac3_decode_end(AVCodecContext *avctx) { AC3DecodeContext *s = avctx->priv_data; ff_mdct_end(&s->imdct_512); ff_mdct_end(&s->imdct_256); av_freep(&s->input_buffer); return 0; } AVCodec ac3_decoder = { .name = "ac3", .type = CODEC_TYPE_AUDIO, .id = CODEC_ID_AC3, .priv_data_size = sizeof (AC3DecodeContext), .init = ac3_decode_init, .close = ac3_decode_end, .decode = ac3_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), }; AVCodec eac3_decoder = { .name = "eac3", .type = CODEC_TYPE_AUDIO, .id = CODEC_ID_EAC3, .priv_data_size = sizeof (AC3DecodeContext), .init = ac3_decode_init, .close = ac3_decode_end, .decode = ac3_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"), };