Mercurial > libavcodec.hg
view ac3dec.c @ 7202:2a4ac127112c libavcodec
Cosmetics: remove braces
| author | vitor |
|---|---|
| date | Sat, 05 Jul 2008 18:43:24 +0000 |
| parents | e943e1409077 |
| children | b5dacf4fc65b |
line wrap: on
line source
/* * AC-3 Audio Decoder * This code is developed as part of Google Summer of Code 2006 Program. * * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com). * Copyright (c) 2007 Justin Ruggles * * Portions of this code are derived from liba52 * http://liba52.sourceforge.net * Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org> * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca> * * 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 <stdio.h> #include <stddef.h> #include <math.h> #include <string.h> #include "libavutil/crc.h" #include "libavutil/random.h" #include "avcodec.h" #include "ac3_parser.h" #include "bitstream.h" #include "dsputil.h" #include "ac3dec.h" #include "ac3dec_data.h" /** Maximum possible frame size when the specification limit is ignored */ #define AC3_MAX_FRAME_SIZE 21695 /** table for grouping exponents */ static uint8_t exp_ungroup_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 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( i / 9 , 3); b1_mantissas[i][1] = symmetric_dequant((i % 9) / 3, 3); b1_mantissas[i][2] = symmetric_dequant((i % 9) % 3, 3); } for(i=0; i<128; i++) { /* bap=2 mantissas */ b2_mantissas[i][0] = symmetric_dequant( i / 25 , 5); b2_mantissas[i][1] = symmetric_dequant((i % 25) / 5, 5); b2_mantissas[i][2] = symmetric_dequant((i % 25) % 5, 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); } /* generate exponent tables reference: Section 7.1.3 Exponent Decoding */ for(i=0; i<128; i++) { exp_ungroup_tab[i][0] = i / 25; exp_ungroup_tab[i][1] = (i % 25) / 5; exp_ungroup_tab[i][2] = (i % 25) % 5; } } /** * 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_init_random(0, &s->dith_state); /* set bias values for float to int16 conversion */ if(s->dsp.float_to_int16 == ff_float_to_int16_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_resilience >= FF_ER_CAREFUL) { s->input_buffer = av_mallocz(AC3_MAX_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); if (!s->input_buffer) return AVERROR_NOMEM; } return 0; } /** * Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream. * GetBitContext within AC3DecodeContext must point to * start of the synchronized ac3 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 AC3 or E-AC3 frame header */ static int parse_frame_header(AC3DecodeContext *s) { AC3HeaderInfo hdr; GetBitContext *gbc = &s->gbc; int err; err = ff_ac3_parse_header(gbc, &hdr); if(err) return err; if(hdr.bitstream_id > 10) return AC3_PARSE_ERROR_BSID; /* 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; } return ac3_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]]; for(i=0; i<s->fbw_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; } /* calculate adjustment needed for each channel to avoid clipping */ s->downmix_coeff_adjust[0] = s->downmix_coeff_adjust[1] = 0.0f; for(i=0; i<s->fbw_channels; i++) { s->downmix_coeff_adjust[0] += s->downmix_coeffs[i][0]; s->downmix_coeff_adjust[1] += s->downmix_coeffs[i][1]; } s->downmix_coeff_adjust[0] = 1.0f / s->downmix_coeff_adjust[0]; s->downmix_coeff_adjust[1] = 1.0f / s->downmix_coeff_adjust[1]; } /** * Decode the grouped exponents according to exponent strategy. * reference: Section 7.1.3 Exponent Decoding */ static void 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<ngrps; grp++) { expacc = get_bits(gbc, 7); dexp[i++] = exp_ungroup_tab[expacc][0]; dexp[i++] = exp_ungroup_tab[expacc][1]; dexp[i++] = exp_ungroup_tab[expacc][2]; } /* convert to absolute exps and expand groups */ prevexp = absexp; for(i=0; i<ngrps*3; i++) { prevexp = av_clip(prevexp + dexp[i]-2, 0, 24); for(j=0; j<group_size; j++) { dexps[(i*group_size)+j] = prevexp; } } } /** * 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 uncouple_channels(AC3DecodeContext *s) { int i, j, ch, bnd, subbnd; subbnd = -1; i = s->start_freq[CPL_CH]; for(bnd=0; bnd<s->num_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; /** * Get the transform coefficients for a particular channel * reference: Section 7.3 Quantization and Decoding of Mantissas */ static void get_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_random(&s->dith_state) & 0x7FFFFF) - 4194304; 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; i<end; i++) { if(!bap[i]) coeffs[i] = 0; } if(s->channel_in_cpl[ch]) { bap = s->bap[CPL_CH]; for(; i<s->end_freq[CPL_CH]; i++) { if(!bap[i]) coeffs[i] = 0; } } } } } /** * Get the transform coefficients. */ static void get_transform_coeffs(AC3DecodeContext *s) { 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 */ get_transform_coeffs_ch(s, 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) { get_transform_coeffs_ch(s, CPL_CH, &m); uncouple_channels(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); } /* if any channel doesn't use dithering, zero appropriate coefficients */ if(!s->dither_all) 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; bnd<s->num_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]; i<bndend; i++) { tmp0 = s->fixed_coeffs[1][i]; tmp1 = s->fixed_coeffs[2][i]; s->fixed_coeffs[1][i] = tmp0 + tmp1; s->fixed_coeffs[2][i] = tmp0 - tmp1; } } } } /** * Perform the 256-point IMDCT */ static void do_imdct_256(AC3DecodeContext *s, int chindex) { int i, k; DECLARE_ALIGNED_16(float, x[128]); FFTComplex z[2][64]; float *o_ptr = s->tmp_output; for(i=0; i<2; i++) { /* de-interleave coefficients */ for(k=0; k<128; k++) { x[k] = s->transform_coeffs[chindex][2*k+i]; } /* run standard IMDCT */ s->imdct_256.fft.imdct_calc(&s->imdct_256, o_ptr, x, s->tmp_imdct); /* reverse the post-rotation & reordering from standard IMDCT */ for(k=0; k<32; k++) { z[i][32+k].re = -o_ptr[128+2*k]; z[i][32+k].im = -o_ptr[2*k]; z[i][31-k].re = o_ptr[2*k+1]; z[i][31-k].im = o_ptr[128+2*k+1]; } } /* apply AC-3 post-rotation & reordering */ for(k=0; k<64; k++) { o_ptr[ 2*k ] = -z[0][ k].im; o_ptr[ 2*k+1] = z[0][63-k].re; o_ptr[128+2*k ] = -z[0][ k].re; o_ptr[128+2*k+1] = z[0][63-k].im; o_ptr[256+2*k ] = -z[1][ k].re; o_ptr[256+2*k+1] = z[1][63-k].im; o_ptr[384+2*k ] = z[1][ k].im; o_ptr[384+2*k+1] = -z[1][63-k].re; } } /** * 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; for (ch=1; ch<=channels; ch++) { if (s->block_switch[ch]) { do_imdct_256(s, ch); } else { s->imdct_512.fft.imdct_calc(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch], s->tmp_imdct); } /* For the first half of the block, apply the window, add the delay from the previous block, and send to output */ s->dsp.vector_fmul_add_add(s->output[ch-1], s->tmp_output, s->window, s->delay[ch-1], 0, 256, 1); /* For the second half of the block, apply the window and store the samples to delay, to be combined with the next block */ s->dsp.vector_fmul_reverse(s->delay[ch-1], s->tmp_output+256, s->window, 256); } } /** * Downmix the output to mono or stereo. */ static void ac3_downmix(AC3DecodeContext *s, float samples[AC3_MAX_CHANNELS][256], int ch_offset) { int i, j; float v0, v1; for(i=0; i<256; i++) { v0 = v1 = 0.0f; for(j=0; j<s->fbw_channels; j++) { v0 += samples[j+ch_offset][i] * s->downmix_coeffs[j][0]; v1 += samples[j+ch_offset][i] * s->downmix_coeffs[j][1]; } v0 *= s->downmix_coeff_adjust[0]; v1 *= s->downmix_coeff_adjust[1]; if(s->output_mode == AC3_CHMODE_MONO) { samples[ch_offset][i] = (v0 + v1) * LEVEL_MINUS_3DB; } else if(s->output_mode == AC3_CHMODE_STEREO) { samples[ ch_offset][i] = v0; samples[1+ch_offset][i] = v1; } } } /** * Upmix delay samples from stereo to original channel layout. */ static void ac3_upmix_delay(AC3DecodeContext *s) { int channel_data_size = sizeof(s->delay[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; } } /** * Parse an audio block from AC-3 bitstream. */ static int ac3_parse_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; 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 */ s->dither_all = 1; for (ch = 1; ch <= fbw_channels; ch++) { s->dither_flag[ch] = get_bits1(gbc); if(!s->dither_flag[ch]) s->dither_all = 0; } /* 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--); /* coupling strategy */ if (get_bits1(gbc)) { memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); cpl_in_use = get_bits1(gbc); if (cpl_in_use) { /* coupling in use */ int cpl_begin_freq, cpl_end_freq; if (channel_mode < AC3_CHMODE_STEREO) { av_log(s->avctx, AV_LOG_ERROR, "coupling not allowed in mono or dual-mono\n"); return -1; } /* determine which channels are coupled */ 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 and band structure */ cpl_begin_freq = get_bits(gbc, 4); cpl_end_freq = get_bits(gbc, 4); if (3 + cpl_end_freq - cpl_begin_freq < 0) { av_log(s->avctx, AV_LOG_ERROR, "3+cplendf = %d < cplbegf = %d\n", 3+cpl_end_freq, cpl_begin_freq); return -1; } s->num_cpl_bands = s->num_cpl_subbands = 3 + cpl_end_freq - cpl_begin_freq; s->start_freq[CPL_CH] = cpl_begin_freq * 12 + 37; s->end_freq[CPL_CH] = cpl_end_freq * 12 + 73; for (bnd = 0; bnd < s->num_cpl_subbands - 1; bnd++) { if (get_bits1(gbc)) { s->cpl_band_struct[bnd] = 1; s->num_cpl_bands--; } } s->cpl_band_struct[s->num_cpl_subbands-1] = 0; } else { /* coupling not in use */ for (ch = 1; ch <= fbw_channels; ch++) s->channel_in_cpl[ch] = 0; } } else if (!blk) { av_log(s->avctx, AV_LOG_ERROR, "new coupling strategy must be present in block 0\n"); return -1; } else { cpl_in_use = s->cpl_in_use[blk-1]; } s->cpl_in_use[blk] = cpl_in_use; /* 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 (get_bits1(gbc)) { int master_cpl_coord, cpl_coord_exp, cpl_coord_mant; 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; } } } /* 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 (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; bnd<s->num_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 */ s->exp_strategy[blk][CPL_CH] = EXP_REUSE; s->exp_strategy[blk][s->lfe_ch] = EXP_REUSE; for (ch = !cpl_in_use; ch <= s->channels; ch++) { 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", 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; 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]); if(ch != CPL_CH && ch != s->lfe_ch) skip_bits(gbc, 2); /* skip gainrng */ } } /* bit allocation information */ 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 (get_bits1(gbc)) { int csnr; csnr = (get_bits(gbc, 6) - 15) << 4; for (ch = !cpl_in_use; ch <= s->channels; ch++) { /* snr offset and fast gain */ s->snr_offset[ch] = (csnr + get_bits(gbc, 4)) << 2; s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)]; } memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); } else if (!blk) { av_log(s->avctx, AV_LOG_ERROR, "new snr offsets must be present in block 0\n"); return -1; } /* coupling leak information */ if (cpl_in_use) { if (get_bits1(gbc)) { s->bit_alloc_params.cpl_fast_leak = get_bits(gbc, 3); s->bit_alloc_params.cpl_slow_leak = get_bits(gbc, 3); bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2); } else if (!blk) { av_log(s->avctx, AV_LOG_ERROR, "new coupling leak info must be present in block 0\n"); return -1; } } /* delta bit allocation information */ if (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 */ 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]); } if(bit_alloc_stages[ch] > 0) { /* Compute bit allocation */ 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, ff_ac3_bap_tab, s->bap[ch]); } } /* unused dummy data */ if (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. */ get_transform_coeffs(s); /* 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]; } for(i=0; i<256; i++) { s->transform_coeffs[ch][i] = s->fixed_coeffs[ch][i] * gain; } } /* 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) { ac3_downmix(s, s->output, 0); } } else { if(downmix_output) { ac3_downmix(s, s->transform_coeffs, 1); } if(!s->downmixed) { s->downmixed = 1; ac3_downmix(s, s->delay, 0); } do_imdct(s, s->out_channels); } /* convert float to 16-bit integer */ for(ch=0; ch<s->out_channels; ch++) { for(i=0; i<256; i++) { s->output[ch][i] += s->add_bias; } s->dsp.float_to_int16(s->int_output[ch], s->output[ch], 256); } 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 i, 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_MAX_FRAME_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_resilience >= 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; } /* parse the audio blocks */ for (blk = 0; blk < s->num_blocks; blk++) { if (!err && ac3_parse_audio_block(s, blk)) { av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n"); } /* interleave output samples */ for (i = 0; i < 256; i++) for (ch = 0; ch < s->out_channels; ch++) *(out_samples++) = s->int_output[ch][i]; } *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/52 / AC-3"), };