ac3dec.c

ffmpeg的完整源代码和作者自己写的文档。不但有在Linux的工程哦

/*
 * 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 <[e-mail]>
 * Copyright (C) 1999-2000 Aaron Holtzman <[e-mail]>
 *
 * 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_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 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_tab[256];

/** dialogue normalization table */
static float dialnorm_tab[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_tab[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_tab[i] = expf((i-31) * M_LN10 / 20.0f);
    }
    dialnorm_tab[0] = dialnorm_tab[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_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 *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_tab[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