Mercurial > libavcodec.hg
view ac3dec.c @ 5677:b031d95d8fae libavcodec
fix decoding of DolbyNet AC3
| author | jbr |
|---|---|
| date | Sat, 15 Sep 2007 00:00:57 +0000 |
| parents | da33495f0621 |
| children | 53c43e7156bc |
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/* * 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 "avcodec.h" #include "ac3_parser.h" #include "bitstream.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_tbl[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_tbl[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 qntztab[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 dynrng_tbl[256]; /** dialogue normalization table */ static float dialnorm_tbl[32]; /** 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 clevs[4] = { 2, 3, 4, 3 }; /** * Table for surround mix levels * reference: Section 5.4.2.5 surmixlev */ static const uint8_t slevs[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 acmod; ///< audio coding mode int dsurmod; ///< dolby surround mode int blksw[AC3_MAX_CHANNELS]; ///< block switch flags int dithflag[AC3_MAX_CHANNELS]; ///< dither flags int dither_all; ///< true if all channels are dithered int cplinu; ///< coupling in use int chincpl[AC3_MAX_CHANNELS]; ///< channel in coupling int phsflginu; ///< phase flags in use int cplbndstrc[18]; ///< coupling band structure int rematstr; ///< rematrixing strategy int nrematbnd; ///< number of rematrixing bands int rematflg[4]; ///< rematrixing flags int expstr[AC3_MAX_CHANNELS]; ///< exponent strategies int snroffst[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets int fgain[AC3_MAX_CHANNELS]; ///< fast gain values (signal-to-mask ratio) int deltbae[AC3_MAX_CHANNELS]; ///< delta bit allocation exists int deltnseg[AC3_MAX_CHANNELS]; ///< number of delta segments uint8_t deltoffst[AC3_MAX_CHANNELS][8]; ///< delta segment offsets uint8_t deltlen[AC3_MAX_CHANNELS][8]; ///< delta segment lengths uint8_t deltba[AC3_MAX_CHANNELS][8]; ///< delta values for each segment int sampling_rate; ///< sample frequency, in Hz int bit_rate; ///< stream bit rate, in bits-per-second int frame_size; ///< current frame size, in bytes int nchans; ///< number of total channels int nfchans; ///< number of full-bandwidth channels int lfeon; ///< lfe channel in use int lfe_ch; ///< index of LFE channel int output_mode; ///< output channel configuration int out_channels; ///< number of output channels float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients float dialnorm[2]; ///< dialogue normalization float dynrng[2]; ///< dynamic range float cplco[AC3_MAX_CHANNELS][18]; ///< coupling coordinates int ncplbnd; ///< number of coupling bands int ncplsubnd; ///< number of coupling sub bands int startmant[AC3_MAX_CHANNELS]; ///< start frequency bin int endmant[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 bndpsd[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 gb; ///< 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; dynrng_tbl[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20); } /* generate dialogue normalization table references: Section 5.4.2.8 dialnorm Section 7.6 Dialogue Normalization */ for(i=1; i<32; i++) { dialnorm_tbl[i] = expf((i-31) * M_LN10 / 20.0f); } dialnorm_tbl[0] = dialnorm_tbl[31]; /* 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_tbl[i][0] = i / 25; exp_ungroup_tbl[i][1] = (i % 25) / 5; exp_ungroup_tbl[i][2] = (i % 25) % 5; } } /** * AVCodec initialization */ static int ac3_decode_init(AVCodecContext *avctx) { AC3DecodeContext *ctx = avctx->priv_data; ctx->avctx = avctx; ac3_common_init(); ac3_tables_init(); ff_mdct_init(&ctx->imdct_256, 8, 1); ff_mdct_init(&ctx->imdct_512, 9, 1); ac3_window_init(ctx->window); dsputil_init(&ctx->dsp, avctx); av_init_random(0, &ctx->dith_state); /* set bias values for float to int16 conversion */ if(ctx->dsp.float_to_int16 == ff_float_to_int16_c) { ctx->add_bias = 385.0f; ctx->mul_bias = 1.0f; } else { ctx->add_bias = 0.0f; ctx->mul_bias = 32767.0f; } 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 *ctx) { AC3HeaderInfo hdr; GetBitContext *gb = &ctx->gb; float cmixlev, surmixlev; int err, i; err = ff_ac3_parse_header(gb->buffer, &hdr); if(err) return err; /* get decoding parameters from header info */ ctx->bit_alloc_params.fscod = hdr.fscod; ctx->acmod = hdr.acmod; cmixlev = gain_levels[clevs[hdr.cmixlev]]; surmixlev = gain_levels[slevs[hdr.surmixlev]]; ctx->dsurmod = hdr.dsurmod; ctx->lfeon = hdr.lfeon; ctx->bit_alloc_params.halfratecod = hdr.halfratecod; ctx->sampling_rate = hdr.sample_rate; ctx->bit_rate = hdr.bit_rate; ctx->nchans = hdr.channels; ctx->nfchans = ctx->nchans - ctx->lfeon; ctx->lfe_ch = ctx->nfchans + 1; ctx->frame_size = hdr.frame_size; /* set default output to all source channels */ ctx->out_channels = ctx->nchans; ctx->output_mode = ctx->acmod; if(ctx->lfeon) ctx->output_mode |= AC3_OUTPUT_LFEON; /* skip over portion of header which has already been read */ skip_bits(gb, 16); // skip the sync_word skip_bits(gb, 16); // skip crc1 skip_bits(gb, 8); // skip fscod and frmsizecod skip_bits(gb, 11); // skip bsid, bsmod, and acmod if(ctx->acmod == AC3_ACMOD_STEREO) { skip_bits(gb, 2); // skip dsurmod } else { if((ctx->acmod & 1) && ctx->acmod != AC3_ACMOD_MONO) skip_bits(gb, 2); // skip cmixlev if(ctx->acmod & 4) skip_bits(gb, 2); // skip surmixlev } skip_bits1(gb); // skip lfeon /* read the rest of the bsi. read twice for dual mono mode. */ i = !(ctx->acmod); do { ctx->dialnorm[i] = dialnorm_tbl[get_bits(gb, 5)]; // dialogue normalization if (get_bits1(gb)) skip_bits(gb, 8); //skip compression if (get_bits1(gb)) skip_bits(gb, 8); //skip language code if (get_bits1(gb)) skip_bits(gb, 7); //skip audio production information } while (i--); skip_bits(gb, 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(gb)) skip_bits(gb, 14); //skip timecode1 / xbsi1 if (get_bits1(gb)) skip_bits(gb, 14); //skip timecode2 / xbsi2 /* skip additional bitstream info */ if (get_bits1(gb)) { i = get_bits(gb, 6); do { skip_bits(gb, 8); } while(i--); } /* set stereo downmixing coefficients reference: Section 7.8.2 Downmixing Into Two Channels */ for(i=0; i<ctx->nfchans; i++) { ctx->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[ctx->acmod][i][0]]; ctx->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[ctx->acmod][i][1]]; } if(ctx->acmod > 1 && ctx->acmod & 1) { ctx->downmix_coeffs[1][0] = ctx->downmix_coeffs[1][1] = cmixlev; } if(ctx->acmod == AC3_ACMOD_2F1R || ctx->acmod == AC3_ACMOD_3F1R) { int nf = ctx->acmod - 2; ctx->downmix_coeffs[nf][0] = ctx->downmix_coeffs[nf][1] = surmixlev * LEVEL_MINUS_3DB; } if(ctx->acmod == AC3_ACMOD_2F2R || ctx->acmod == AC3_ACMOD_3F2R) { int nf = ctx->acmod - 4; ctx->downmix_coeffs[nf][0] = ctx->downmix_coeffs[nf+1][1] = surmixlev; } return 0; } /** * Decode the grouped exponents according to exponent strategy. * reference: Section 7.1.3 Exponent Decoding */ static void decode_exponents(GetBitContext *gb, int expstr, int ngrps, uint8_t absexp, int8_t *dexps) { int i, j, grp, grpsize; int dexp[256]; int expacc, prevexp; /* unpack groups */ grpsize = expstr + (expstr == EXP_D45); for(grp=0,i=0; grp<ngrps; grp++) { expacc = get_bits(gb, 7); dexp[i++] = exp_ungroup_tbl[expacc][0]; dexp[i++] = exp_ungroup_tbl[expacc][1]; dexp[i++] = exp_ungroup_tbl[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<grpsize; j++) { dexps[(i*grpsize)+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 *ctx) { int i, j, ch, bnd, subbnd; subbnd = -1; i = ctx->startmant[CPL_CH]; for(bnd=0; bnd<ctx->ncplbnd; bnd++) { do { subbnd++; for(j=0; j<12; j++) { for(ch=1; ch<=ctx->nfchans; ch++) { if(ctx->chincpl[ch]) ctx->transform_coeffs[ch][i] = ctx->transform_coeffs[CPL_CH][i] * ctx->cplco[ch][bnd] * 8.0f; } i++; } } while(ctx->cplbndstrc[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 *ctx, int ch_index, mant_groups *m) { GetBitContext *gb = &ctx->gb; int i, gcode, tbap, start, end; uint8_t *exps; uint8_t *bap; float *coeffs; exps = ctx->dexps[ch_index]; bap = ctx->bap[ch_index]; coeffs = ctx->transform_coeffs[ch_index]; start = ctx->startmant[ch_index]; end = ctx->endmant[ch_index]; for (i = start; i < end; i++) { tbap = bap[i]; switch (tbap) { case 0: coeffs[i] = ((av_random(&ctx->dith_state) & 0xFFFF) * LEVEL_MINUS_3DB) / 32768.0f; break; case 1: if(m->b1ptr > 2) { gcode = get_bits(gb, 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(gb, 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(gb, 3)]; break; case 4: if(m->b4ptr > 1) { gcode = get_bits(gb, 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(gb, 4)]; break; default: /* asymmetric dequantization */ coeffs[i] = get_sbits(gb, qntztab[tbap]) * scale_factors[qntztab[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 *ctx) { int ch, i; int end=0; float *coeffs; uint8_t *bap; for(ch=1; ch<=ctx->nfchans; ch++) { if(!ctx->dithflag[ch]) { coeffs = ctx->transform_coeffs[ch]; bap = ctx->bap[ch]; if(ctx->chincpl[ch]) end = ctx->startmant[CPL_CH]; else end = ctx->endmant[ch]; for(i=0; i<end; i++) { if(bap[i] == 0) coeffs[i] = 0.0f; } if(ctx->chincpl[ch]) { bap = ctx->bap[CPL_CH]; for(; i<ctx->endmant[CPL_CH]; i++) { if(bap[i] == 0) coeffs[i] = 0.0f; } } } } } /** * Get the transform coefficients. */ static int get_transform_coeffs(AC3DecodeContext * ctx) { int ch, end; int got_cplchan = 0; mant_groups m; m.b1ptr = m.b2ptr = m.b4ptr = 3; for (ch = 1; ch <= ctx->nchans; ch++) { /* transform coefficients for full-bandwidth channel */ if (get_transform_coeffs_ch(ctx, ch, &m)) return -1; /* tranform coefficients for coupling channel come right after the coefficients for the first coupled channel*/ if (ctx->chincpl[ch]) { if (!got_cplchan) { if (get_transform_coeffs_ch(ctx, CPL_CH, &m)) { av_log(ctx->avctx, AV_LOG_ERROR, "error in decoupling channels\n"); return -1; } uncouple_channels(ctx); got_cplchan = 1; } end = ctx->endmant[CPL_CH]; } else { end = ctx->endmant[ch]; } do ctx->transform_coeffs[ch][end] = 0; while(++end < 256); } /* if any channel doesn't use dithering, zero appropriate coefficients */ if(!ctx->dither_all) remove_dithering(ctx); return 0; } /** * Stereo rematrixing. * reference: Section 7.5.4 Rematrixing : Decoding Technique */ static void do_rematrixing(AC3DecodeContext *ctx) { int bnd, i; int end, bndend; float tmp0, tmp1; end = FFMIN(ctx->endmant[1], ctx->endmant[2]); for(bnd=0; bnd<ctx->nrematbnd; bnd++) { if(ctx->rematflg[bnd]) { bndend = FFMIN(end, rematrix_band_tbl[bnd+1]); for(i=rematrix_band_tbl[bnd]; i<bndend; i++) { tmp0 = ctx->transform_coeffs[1][i]; tmp1 = ctx->transform_coeffs[2][i]; ctx->transform_coeffs[1][i] = tmp0 + tmp1; ctx->transform_coeffs[2][i] = tmp0 - tmp1; } } } } /** * Perform the 256-point IMDCT */ static void do_imdct_256(AC3DecodeContext *ctx, int chindex) { int i, k; DECLARE_ALIGNED_16(float, x[128]); FFTComplex z[2][64]; float *o_ptr = ctx->tmp_output; for(i=0; i<2; i++) { /* de-interleave coefficients */ for(k=0; k<128; k++) { x[k] = ctx->transform_coeffs[chindex][2*k+i]; } /* run standard IMDCT */ ctx->imdct_256.fft.imdct_calc(&ctx->imdct_256, o_ptr, x, ctx->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 *ctx) { int ch; int nchans; /* Don't perform the IMDCT on the LFE channel unless it's used in the output */ nchans = ctx->nfchans; if(ctx->output_mode & AC3_OUTPUT_LFEON) nchans++; for (ch=1; ch<=nchans; ch++) { if (ctx->blksw[ch]) { do_imdct_256(ctx, ch); } else { ctx->imdct_512.fft.imdct_calc(&ctx->imdct_512, ctx->tmp_output, ctx->transform_coeffs[ch], ctx->tmp_imdct); } /* For the first half of the block, apply the window, add the delay from the previous block, and send to output */ ctx->dsp.vector_fmul_add_add(ctx->output[ch-1], ctx->tmp_output, ctx->window, ctx->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 */ ctx->dsp.vector_fmul_reverse(ctx->delay[ch-1], ctx->tmp_output+256, ctx->window, 256); } } /** * Downmix the output to mono or stereo. */ static void ac3_downmix(float samples[AC3_MAX_CHANNELS][256], int nfchans, int output_mode, float coef[AC3_MAX_CHANNELS][2]) { int i, j; float v0, v1, s0, s1; for(i=0; i<256; i++) { v0 = v1 = s0 = s1 = 0.0f; for(j=0; j<nfchans; j++) { v0 += samples[j][i] * coef[j][0]; v1 += samples[j][i] * coef[j][1]; s0 += coef[j][0]; s1 += coef[j][1]; } v0 /= s0; v1 /= s1; if(output_mode == AC3_ACMOD_MONO) { samples[0][i] = (v0 + v1) * LEVEL_MINUS_3DB; } else if(output_mode == AC3_ACMOD_STEREO) { samples[0][i] = v0; samples[1][i] = v1; } } } /** * Parse an audio block from AC-3 bitstream. */ static int ac3_parse_audio_block(AC3DecodeContext *ctx, int blk) { int nfchans = ctx->nfchans; int acmod = ctx->acmod; int i, bnd, seg, ch; GetBitContext *gb = &ctx->gb; uint8_t bit_alloc_stages[AC3_MAX_CHANNELS]; memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS); /* block switch flags */ for (ch = 1; ch <= nfchans; ch++) ctx->blksw[ch] = get_bits1(gb); /* dithering flags */ ctx->dither_all = 1; for (ch = 1; ch <= nfchans; ch++) { ctx->dithflag[ch] = get_bits1(gb); if(!ctx->dithflag[ch]) ctx->dither_all = 0; } /* dynamic range */ i = !(ctx->acmod); do { if(get_bits1(gb)) { ctx->dynrng[i] = dynrng_tbl[get_bits(gb, 8)]; } else if(blk == 0) { ctx->dynrng[i] = 1.0f; } } while(i--); /* coupling strategy */ if (get_bits1(gb)) { memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); ctx->cplinu = get_bits1(gb); if (ctx->cplinu) { /* coupling in use */ int cplbegf, cplendf; /* determine which channels are coupled */ for (ch = 1; ch <= nfchans; ch++) ctx->chincpl[ch] = get_bits1(gb); /* phase flags in use */ if (acmod == AC3_ACMOD_STEREO) ctx->phsflginu = get_bits1(gb); /* coupling frequency range and band structure */ cplbegf = get_bits(gb, 4); cplendf = get_bits(gb, 4); if (3 + cplendf - cplbegf < 0) { av_log(ctx->avctx, AV_LOG_ERROR, "cplendf = %d < cplbegf = %d\n", cplendf, cplbegf); return -1; } ctx->ncplbnd = ctx->ncplsubnd = 3 + cplendf - cplbegf; ctx->startmant[CPL_CH] = cplbegf * 12 + 37; ctx->endmant[CPL_CH] = cplendf * 12 + 73; for (bnd = 0; bnd < ctx->ncplsubnd - 1; bnd++) { if (get_bits1(gb)) { ctx->cplbndstrc[bnd] = 1; ctx->ncplbnd--; } } } else { /* coupling not in use */ for (ch = 1; ch <= nfchans; ch++) ctx->chincpl[ch] = 0; } } /* coupling coordinates */ if (ctx->cplinu) { int cplcoe = 0; for (ch = 1; ch <= nfchans; ch++) { if (ctx->chincpl[ch]) { if (get_bits1(gb)) { int mstrcplco, cplcoexp, cplcomant; cplcoe = 1; mstrcplco = 3 * get_bits(gb, 2); for (bnd = 0; bnd < ctx->ncplbnd; bnd++) { cplcoexp = get_bits(gb, 4); cplcomant = get_bits(gb, 4); if (cplcoexp == 15) ctx->cplco[ch][bnd] = cplcomant / 16.0f; else ctx->cplco[ch][bnd] = (cplcomant + 16.0f) / 32.0f; ctx->cplco[ch][bnd] *= scale_factors[cplcoexp + mstrcplco]; } } } } /* phase flags */ if (acmod == AC3_ACMOD_STEREO && ctx->phsflginu && cplcoe) { for (bnd = 0; bnd < ctx->ncplbnd; bnd++) { if (get_bits1(gb)) ctx->cplco[2][bnd] = -ctx->cplco[2][bnd]; } } } /* stereo rematrixing strategy and band structure */ if (acmod == AC3_ACMOD_STEREO) { ctx->rematstr = get_bits1(gb); if (ctx->rematstr) { ctx->nrematbnd = 4; if(ctx->cplinu && ctx->startmant[CPL_CH] <= 61) ctx->nrematbnd -= 1 + (ctx->startmant[CPL_CH] == 37); for(bnd=0; bnd<ctx->nrematbnd; bnd++) ctx->rematflg[bnd] = get_bits1(gb); } } /* exponent strategies for each channel */ ctx->expstr[CPL_CH] = EXP_REUSE; ctx->expstr[ctx->lfe_ch] = EXP_REUSE; for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) { if(ch == ctx->lfe_ch) ctx->expstr[ch] = get_bits(gb, 1); else ctx->expstr[ch] = get_bits(gb, 2); if(ctx->expstr[ch] != EXP_REUSE) bit_alloc_stages[ch] = 3; } /* channel bandwidth */ for (ch = 1; ch <= nfchans; ch++) { ctx->startmant[ch] = 0; if (ctx->expstr[ch] != EXP_REUSE) { int prev = ctx->endmant[ch]; if (ctx->chincpl[ch]) ctx->endmant[ch] = ctx->startmant[CPL_CH]; else { int chbwcod = get_bits(gb, 6); if (chbwcod > 60) { av_log(ctx->avctx, AV_LOG_ERROR, "chbwcod = %d > 60", chbwcod); return -1; } ctx->endmant[ch] = chbwcod * 3 + 73; } if(blk > 0 && ctx->endmant[ch] != prev) memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); } } ctx->startmant[ctx->lfe_ch] = 0; ctx->endmant[ctx->lfe_ch] = 7; /* decode exponents for each channel */ for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) { if (ctx->expstr[ch] != EXP_REUSE) { int grpsize, ngrps; grpsize = 3 << (ctx->expstr[ch] - 1); if(ch == CPL_CH) ngrps = (ctx->endmant[ch] - ctx->startmant[ch]) / grpsize; else if(ch == ctx->lfe_ch) ngrps = 2; else ngrps = (ctx->endmant[ch] + grpsize - 4) / grpsize; ctx->dexps[ch][0] = get_bits(gb, 4) << !ch; decode_exponents(gb, ctx->expstr[ch], ngrps, ctx->dexps[ch][0], &ctx->dexps[ch][ctx->startmant[ch]+!!ch]); if(ch != CPL_CH && ch != ctx->lfe_ch) skip_bits(gb, 2); /* skip gainrng */ } } /* bit allocation information */ if (get_bits1(gb)) { ctx->bit_alloc_params.sdecay = ff_sdecaytab[get_bits(gb, 2)] >> ctx->bit_alloc_params.halfratecod; ctx->bit_alloc_params.fdecay = ff_fdecaytab[get_bits(gb, 2)] >> ctx->bit_alloc_params.halfratecod; ctx->bit_alloc_params.sgain = ff_sgaintab[get_bits(gb, 2)]; ctx->bit_alloc_params.dbknee = ff_dbkneetab[get_bits(gb, 2)]; ctx->bit_alloc_params.floor = ff_floortab[get_bits(gb, 3)]; for(ch=!ctx->cplinu; ch<=ctx->nchans; 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(gb)) { int csnr; csnr = (get_bits(gb, 6) - 15) << 4; for (ch = !ctx->cplinu; ch <= ctx->nchans; ch++) { /* snr offset and fast gain */ ctx->snroffst[ch] = (csnr + get_bits(gb, 4)) << 2; ctx->fgain[ch] = ff_fgaintab[get_bits(gb, 3)]; } memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS); } /* coupling leak information */ if (ctx->cplinu && get_bits1(gb)) { ctx->bit_alloc_params.cplfleak = get_bits(gb, 3); ctx->bit_alloc_params.cplsleak = get_bits(gb, 3); bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2); } /* delta bit allocation information */ if (get_bits1(gb)) { /* delta bit allocation exists (strategy) */ for (ch = !ctx->cplinu; ch <= nfchans; ch++) { ctx->deltbae[ch] = get_bits(gb, 2); if (ctx->deltbae[ch] == DBA_RESERVED) { av_log(ctx->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 = !ctx->cplinu; ch <= nfchans; ch++) { if (ctx->deltbae[ch] == DBA_NEW) { ctx->deltnseg[ch] = get_bits(gb, 3); for (seg = 0; seg <= ctx->deltnseg[ch]; seg++) { ctx->deltoffst[ch][seg] = get_bits(gb, 5); ctx->deltlen[ch][seg] = get_bits(gb, 4); ctx->deltba[ch][seg] = get_bits(gb, 3); } } } } else if(blk == 0) { for(ch=0; ch<=ctx->nchans; ch++) { ctx->deltbae[ch] = DBA_NONE; } } /* Bit allocation */ for(ch=!ctx->cplinu; ch<=ctx->nchans; ch++) { if(bit_alloc_stages[ch] > 2) { /* Exponent mapping into PSD and PSD integration */ ff_ac3_bit_alloc_calc_psd(ctx->dexps[ch], ctx->startmant[ch], ctx->endmant[ch], ctx->psd[ch], ctx->bndpsd[ch]); } if(bit_alloc_stages[ch] > 1) { /* Compute excitation function, Compute masking curve, and Apply delta bit allocation */ ff_ac3_bit_alloc_calc_mask(&ctx->bit_alloc_params, ctx->bndpsd[ch], ctx->startmant[ch], ctx->endmant[ch], ctx->fgain[ch], (ch == ctx->lfe_ch), ctx->deltbae[ch], ctx->deltnseg[ch], ctx->deltoffst[ch], ctx->deltlen[ch], ctx->deltba[ch], ctx->mask[ch]); } if(bit_alloc_stages[ch] > 0) { /* Compute bit allocation */ ff_ac3_bit_alloc_calc_bap(ctx->mask[ch], ctx->psd[ch], ctx->startmant[ch], ctx->endmant[ch], ctx->snroffst[ch], ctx->bit_alloc_params.floor, ctx->bap[ch]); } } /* unused dummy data */ if (get_bits1(gb)) { int skipl = get_bits(gb, 9); while(skipl--) skip_bits(gb, 8); } /* unpack the transform coefficients this also uncouples channels if coupling is in use. */ if (get_transform_coeffs(ctx)) { av_log(ctx->avctx, AV_LOG_ERROR, "Error in routine get_transform_coeffs\n"); return -1; } /* recover coefficients if rematrixing is in use */ if(ctx->acmod == AC3_ACMOD_STEREO) do_rematrixing(ctx); /* apply scaling to coefficients (headroom, dialnorm, dynrng) */ for(ch=1; ch<=ctx->nchans; ch++) { float gain = 2.0f * ctx->mul_bias; if(ctx->acmod == AC3_ACMOD_DUALMONO) { gain *= ctx->dialnorm[ch-1] * ctx->dynrng[ch-1]; } else { gain *= ctx->dialnorm[0] * ctx->dynrng[0]; } for(i=0; i<ctx->endmant[ch]; i++) { ctx->transform_coeffs[ch][i] *= gain; } } do_imdct(ctx); /* downmix output if needed */ if(ctx->nchans != ctx->out_channels && !((ctx->output_mode & AC3_OUTPUT_LFEON) && ctx->nfchans == ctx->out_channels)) { ac3_downmix(ctx->output, ctx->nfchans, ctx->output_mode, ctx->downmix_coeffs); } /* convert float to 16-bit integer */ for(ch=0; ch<ctx->out_channels; ch++) { for(i=0; i<256; i++) { ctx->output[ch][i] += ctx->add_bias; } ctx->dsp.float_to_int16(ctx->int_output[ch], ctx->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 *ctx = (AC3DecodeContext *)avctx->priv_data; int16_t *out_samples = (int16_t *)data; int i, blk, ch; /* initialize the GetBitContext with the start of valid AC-3 Frame */ init_get_bits(&ctx->gb, buf, buf_size * 8); /* parse the syncinfo */ if (ac3_parse_header(ctx)) { av_log(avctx, AV_LOG_ERROR, "\n"); *data_size = 0; return buf_size; } avctx->sample_rate = ctx->sampling_rate; avctx->bit_rate = ctx->bit_rate; /* channel config */ ctx->out_channels = ctx->nchans; if (avctx->channels == 0) { avctx->channels = ctx->out_channels; } else if(ctx->out_channels < avctx->channels) { av_log(avctx, AV_LOG_ERROR, "Cannot upmix AC3 from %d to %d channels.\n", ctx->out_channels, avctx->channels); return -1; } if(avctx->channels == 2) { ctx->output_mode = AC3_ACMOD_STEREO; } else if(avctx->channels == 1) { ctx->output_mode = AC3_ACMOD_MONO; } else if(avctx->channels != ctx->out_channels) { av_log(avctx, AV_LOG_ERROR, "Cannot downmix AC3 from %d to %d channels.\n", ctx->out_channels, avctx->channels); return -1; } ctx->out_channels = avctx->channels; /* parse the audio blocks */ for (blk = 0; blk < NB_BLOCKS; blk++) { if (ac3_parse_audio_block(ctx, blk)) { av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n"); *data_size = 0; return ctx->frame_size; } for (i = 0; i < 256; i++) for (ch = 0; ch < ctx->out_channels; ch++) *(out_samples++) = ctx->int_output[ch][i]; } *data_size = NB_BLOCKS * 256 * avctx->channels * sizeof (int16_t); return ctx->frame_size; } /** * Uninitialize the AC-3 decoder. */ static int ac3_decode_end(AVCodecContext *avctx) { AC3DecodeContext *ctx = (AC3DecodeContext *)avctx->priv_data; ff_mdct_end(&ctx->imdct_512); ff_mdct_end(&ctx->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, };