/* * 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 * 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 "avcodec.h" #include "ac3_parser.h" #include "bitstream.h" #include "crc.h" #include "dsputil.h" #include "random.h" /** * Table of bin locations for rematrixing bands * reference: Section 7.5.2 Rematrixing : Frequency Band Definitions */ static const uint8_t rematrix_band_tab[5] = { 13, 25, 37, 61, 253 }; /** * table for exponent to scale_factor mapping * scale_factors[i] = 2 ^ -i */ static float scale_factors[25]; /** table for grouping exponents */ static uint8_t exp_ungroup_tab[128][3]; /** tables for ungrouping mantissas */ static float b1_mantissas[32][3]; static float b2_mantissas[128][3]; static float b3_mantissas[8]; static float b4_mantissas[128][2]; static float 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_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[6] = { LEVEL_ZERO, LEVEL_ONE, LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB, LEVEL_MINUS_9DB }; /** * Table for center mix levels * reference: Section 5.4.2.4 cmixlev */ static const uint8_t center_levels[4] = { 2, 3, 4, 3 }; /** * Table for surround mix levels * reference: Section 5.4.2.5 surmixlev */ static const uint8_t surround_levels[4] = { 2, 4, 0, 4 }; /** * 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] = { { { 1, 0 }, { 0, 1 }, }, { { 2, 2 }, }, { { 1, 0 }, { 0, 1 }, }, { { 1, 0 }, { 3, 3 }, { 0, 1 }, }, { { 1, 0 }, { 0, 1 }, { 4, 4 }, }, { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 5, 5 }, }, { { 1, 0 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, }, { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, }, }; /* override ac3.h to include coupling channel */ #undef AC3_MAX_CHANNELS #define AC3_MAX_CHANNELS 7 #define CPL_CH 0 #define AC3_OUTPUT_LFEON 8 typedef struct { int channel_mode; ///< channel mode (acmod) int block_switch[AC3_MAX_CHANNELS]; ///< block switch flags int dither_flag[AC3_MAX_CHANNELS]; ///< dither flags int dither_all; ///< true if all channels are dithered int cpl_in_use; ///< coupling in use int channel_in_cpl[AC3_MAX_CHANNELS]; ///< channel in coupling int phase_flags_in_use; ///< phase flags in use int phase_flags[18]; ///< phase flags int cpl_band_struct[18]; ///< coupling band structure int num_rematrixing_bands; ///< number of rematrixing bands int rematrixing_flags[4]; ///< rematrixing flags int exp_strategy[AC3_MAX_CHANNELS]; ///< exponent strategies int snr_offset[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets int fast_gain[AC3_MAX_CHANNELS]; ///< fast gain values (signal-to-mask ratio) int dba_mode[AC3_MAX_CHANNELS]; ///< delta bit allocation mode int dba_nsegs[AC3_MAX_CHANNELS]; ///< number of delta segments uint8_t dba_offsets[AC3_MAX_CHANNELS][8]; ///< delta segment offsets uint8_t dba_lengths[AC3_MAX_CHANNELS][8]; ///< delta segment lengths uint8_t dba_values[AC3_MAX_CHANNELS][8]; ///< delta values for each segment int sample_rate; ///< sample frequency, in Hz int bit_rate; ///< stream bit rate, in bits-per-second int frame_size; ///< current frame size, in bytes int channels; ///< number of total channels int fbw_channels; ///< number of full-bandwidth channels int lfe_on; ///< lfe channel in use int lfe_ch; ///< index of LFE channel int output_mode; ///< output channel configuration int out_channels; ///< number of output channels int center_mix_level; ///< Center mix level index int surround_mix_level; ///< Surround mix level index float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients float dynamic_range[2]; ///< dynamic range float cpl_coords[AC3_MAX_CHANNELS][18]; ///< coupling coordinates int num_cpl_bands; ///< number of coupling bands int num_cpl_subbands; ///< number of coupling sub bands int start_freq[AC3_MAX_CHANNELS]; ///< start frequency bin int end_freq[AC3_MAX_CHANNELS]; ///< end frequency bin AC3BitAllocParameters bit_alloc_params; ///< bit allocation parameters int8_t dexps[AC3_MAX_CHANNELS][256]; ///< decoded exponents uint8_t bap[AC3_MAX_CHANNELS][256]; ///< bit allocation pointers int16_t psd[AC3_MAX_CHANNELS][256]; ///< scaled exponents int16_t band_psd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents int16_t mask[AC3_MAX_CHANNELS][50]; ///< masking curve values DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients /* For IMDCT. */ MDCTContext imdct_512; ///< for 512 sample IMDCT MDCTContext imdct_256; ///< for 256 sample IMDCT DSPContext dsp; ///< for optimization float add_bias; ///< offset for float_to_int16 conversion float mul_bias; ///< scaling for float_to_int16 conversion DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]); ///< output after imdct transform and windowing DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]); ///< delay - added to the next block DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients /* Miscellaneous. */ GetBitContext gbc; ///< bitstream reader AVRandomState dith_state; ///< for dither generation AVCodecContext *avctx; ///< parent context } AC3DecodeContext; /** * Generate a Kaiser-Bessel Derived Window. */ static void ac3_window_init(float *window) { int i, j; double sum = 0.0, bessel, tmp; double local_window[256]; double alpha2 = (5.0 * M_PI / 256.0) * (5.0 * M_PI / 256.0); for (i = 0; i < 256; i++) { tmp = i * (256 - i) * alpha2; bessel = 1.0; for (j = 100; j > 0; j--) /* default to 100 iterations */ bessel = bessel * tmp / (j * j) + 1; sum += bessel; local_window[i] = sum; } sum++; for (i = 0; i < 256; i++) window[i] = sqrt(local_window[i] / sum); } /** * Symmetrical Dequantization * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization * Tables 7.19 to 7.23 */ static inline float symmetric_dequant(int code, int levels) { return (code - (levels >> 1)) * (2.0f / levels); } /* * Initialize tables at runtime. */ static 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 scale factors for exponents and asymmetrical dequantization reference: Section 7.3.2 Expansion of Mantissas for Asymmetric Quantization */ for (i = 0; i < 25; i++) scale_factors[i] = pow(2.0, -i); /* 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 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); ac3_window_init(s->window); 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; } 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) { AC3HeaderInfo hdr; GetBitContext *gbc = &s->gbc; int err, i; err = ff_ac3_parse_header(gbc->buffer, &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; /* set default output to all source channels */ s->out_channels = s->channels; s->output_mode = s->channel_mode; if(s->lfe_on) s->output_mode |= AC3_OUTPUT_LFEON; /* set default mix levels */ s->center_mix_level = 3; // -4.5dB s->surround_mix_level = 4; // -6.0dB /* skip over portion of header which has already been read */ skip_bits(gbc, 16); // skip the sync_word skip_bits(gbc, 16); // skip crc1 skip_bits(gbc, 8); // skip fscod and frmsizecod skip_bits(gbc, 11); // skip bsid, bsmod, and acmod if(s->channel_mode == AC3_CHMODE_STEREO) { skip_bits(gbc, 2); // skip dsurmod } else { if((s->channel_mode & 1) && s->channel_mode != AC3_CHMODE_MONO) s->center_mix_level = center_levels[get_bits(gbc, 2)]; if(s->channel_mode & 4) s->surround_mix_level = surround_levels[get_bits(gbc, 2)]; } skip_bits1(gbc); // skip lfeon /* 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; } /** * 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[s->center_mix_level]; float smix = gain_levels[s->surround_mix_level]; 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; } } /** * 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; grpstart_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->transform_coeffs[ch][i] = s->transform_coeffs[CPL_CH][i] * s->cpl_coords[ch][bnd] * 8.0f; if (ch == 2 && s->phase_flags[bnd]) s->transform_coeffs[ch][i] = -s->transform_coeffs[ch][i]; } } i++; } } while(s->cpl_band_struct[subbnd]); } } /** * Grouped mantissas for 3-level 5-level and 11-level quantization */ typedef struct { float b1_mant[3]; float b2_mant[3]; float 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 int 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; float *coeffs; exps = s->dexps[ch_index]; bap = s->bap[ch_index]; coeffs = s->transform_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) & 0xFFFF) / 65535.0f) - 0.5f; 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 */ coeffs[i] = get_sbits(gbc, quantization_tab[tbap]) * scale_factors[quantization_tab[tbap]-1]; break; } coeffs[i] *= scale_factors[exps[i]]; } return 0; } /** * 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; float *coeffs; uint8_t *bap; for(ch=1; ch<=s->fbw_channels; ch++) { if(!s->dither_flag[ch]) { coeffs = s->transform_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] = 0.0f; } } } } } /** * Get the transform coefficients. */ static int 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 */ if (get_transform_coeffs_ch(s, ch, &m)) return -1; /* tranform coefficients for coupling channel come right after the coefficients for the first coupled channel*/ if (s->channel_in_cpl[ch]) { if (!got_cplchan) { if (get_transform_coeffs_ch(s, CPL_CH, &m)) { av_log(s->avctx, AV_LOG_ERROR, "error in decoupling channels\n"); return -1; } uncouple_channels(s); got_cplchan = 1; } end = s->end_freq[CPL_CH]; } else { end = s->end_freq[ch]; } do s->transform_coeffs[ch][end] = 0; while(++end < 256); } /* if any channel doesn't use dithering, zero appropriate coefficients */ if(!s->dither_all) remove_dithering(s); return 0; } /** * Stereo rematrixing. * reference: Section 7.5.4 Rematrixing : Decoding Technique */ static void do_rematrixing(AC3DecodeContext *s) { int bnd, i; int end, bndend; float 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, rematrix_band_tab[bnd+1]); for(i=rematrix_band_tab[bnd]; itransform_coeffs[1][i]; tmp1 = s->transform_coeffs[2][i]; s->transform_coeffs[1][i] = tmp0 + tmp1; s->transform_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 ch; int channels; /* Don't perform the IMDCT on the LFE channel unless it's used in the output */ channels = s->fbw_channels; if(s->output_mode & AC3_OUTPUT_LFEON) channels++; 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) { int i, j; float v0, v1, s0, s1; for(i=0; i<256; i++) { v0 = v1 = s0 = s1 = 0.0f; for(j=0; jfbw_channels; j++) { v0 += s->output[j][i] * s->downmix_coeffs[j][0]; v1 += s->output[j][i] * s->downmix_coeffs[j][1]; s0 += s->downmix_coeffs[j][0]; s1 += s->downmix_coeffs[j][1]; } v0 /= s0; v1 /= s1; if(s->output_mode == AC3_CHMODE_MONO) { s->output[0][i] = (v0 + v1) * LEVEL_MINUS_3DB; } else if(s->output_mode == AC3_CHMODE_STEREO) { s->output[0][i] = v0; s->output[1][i] = v1; } } } /** * 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; GetBitContext *gbc = &s->gbc; uint8_t bit_alloc_stages[AC3_MAX_CHANNELS]; memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS); /* block switch flags */ for (ch = 1; ch <= fbw_channels; ch++) s->block_switch[ch] = get_bits1(gbc); /* 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); s->cpl_in_use = get_bits1(gbc); if (s->cpl_in_use) { /* coupling in use */ int cpl_begin_freq, cpl_end_freq; /* 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; } } /* coupling coordinates */ if (s->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 / 16.0f; else s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16.0f) / 32.0f; s->cpl_coords[ch][bnd] *= scale_factors[cpl_coord_exp + master_cpl_coord]; } } } } /* 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(s->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); } } /* exponent strategies for each channel */ s->exp_strategy[CPL_CH] = EXP_REUSE; s->exp_strategy[s->lfe_ch] = EXP_REUSE; for (ch = !s->cpl_in_use; ch <= s->channels; ch++) { if(ch == s->lfe_ch) s->exp_strategy[ch] = get_bits(gbc, 1); else s->exp_strategy[ch] = get_bits(gbc, 2); if(s->exp_strategy[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[ch] != EXP_REUSE) { 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; } if(blk > 0 && s->end_freq[ch] != prev) memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); } } s->start_freq[s->lfe_ch] = 0; s->end_freq[s->lfe_ch] = 7; /* decode exponents for each channel */ for (ch = !s->cpl_in_use; ch <= s->channels; ch++) { if (s->exp_strategy[ch] != EXP_REUSE) { int group_size, num_groups; group_size = 3 << (s->exp_strategy[ch] - 1); if(ch == CPL_CH) num_groups = (s->end_freq[ch] - s->start_freq[ch]) / group_size; else if(ch == s->lfe_ch) num_groups = 2; else num_groups = (s->end_freq[ch] + group_size - 4) / group_size; s->dexps[ch][0] = get_bits(gbc, 4) << !ch; decode_exponents(gbc, s->exp_strategy[ch], num_groups, 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=!s->cpl_in_use; ch<=s->channels; ch++) { bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2); } } /* 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 = !s->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); } /* coupling leak information */ if (s->cpl_in_use && 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); } /* delta bit allocation information */ if (get_bits1(gbc)) { /* delta bit allocation exists (strategy) */ for (ch = !s->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 = !s->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); } } } } else if(blk == 0) { for(ch=0; ch<=s->channels; ch++) { s->dba_mode[ch] = DBA_NONE; } } /* Bit allocation */ for(ch=!s->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, 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. */ if (get_transform_coeffs(s)) { av_log(s->avctx, AV_LOG_ERROR, "Error in routine get_transform_coeffs\n"); return -1; } /* 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 = 2.0f * s->mul_bias; if(s->channel_mode == AC3_CHMODE_DUALMONO) { gain *= s->dynamic_range[ch-1]; } else { gain *= s->dynamic_range[0]; } for(i=0; iend_freq[ch]; i++) { s->transform_coeffs[ch][i] *= gain; } } do_imdct(s); /* downmix output if needed */ if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) && s->fbw_channels == s->out_channels)) { ac3_downmix(s); } /* convert float to 16-bit integer */ for(ch=0; chout_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, 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 */ init_get_bits(&s->gbc, buf, buf_size * 8); /* parse the syncinfo */ err = ac3_parse_header(s); if(err) { switch(err) { case AC3_PARSE_ERROR_SYNC: av_log(avctx, AV_LOG_ERROR, "frame sync error\n"); break; 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; default: av_log(avctx, AV_LOG_ERROR, "invalid header\n"); break; } return -1; } /* check that reported frame size fits in input buffer */ if(s->frame_size > buf_size) { av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); return -1; } /* check for crc mismatch */ if(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"); return -1; } /* TODO: error concealment */ } avctx->sample_rate = s->sample_rate; avctx->bit_rate = s->bit_rate; /* channel config */ s->out_channels = s->channels; 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); } /* parse the audio blocks */ for (blk = 0; blk < NB_BLOCKS; blk++) { if (ac3_parse_audio_block(s, blk)) { av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n"); *data_size = 0; return s->frame_size; } for (i = 0; i < 256; i++) for (ch = 0; ch < s->out_channels; ch++) *(out_samples++) = s->int_output[ch][i]; } *data_size = NB_BLOCKS * 256 * avctx->channels * sizeof (int16_t); return s->frame_size; } /** * Uninitialize the AC-3 decoder. */ static int ac3_decode_end(AVCodecContext *avctx) { AC3DecodeContext *s = avctx->priv_data; ff_mdct_end(&s->imdct_512); ff_mdct_end(&s->imdct_256); 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, };