view ac3dec.c @ 7354:456957d86106 libavcodec

My commit at r14340 was not the right solution. For a monochromatic frame, there will be only one centroid and it will be impossible to find three distinct ones. It is better to just avoid shifting if there are not three different centroids.

/*
 * 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 "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) - 0x400000;
                break;

            case 1:
                if(m->b1ptr > 2) {
                    gcode = get_bits(gbc, 5);
                    m->b1_mant[0] = b1_mantissas[gcode][0];
                    m->b1_mant[1] = b1_mantissas[gcode][1];
                    m->b1_mant[2] = b1_mantissas[gcode][2];
                    m->b1ptr = 0;
                }
                coeffs[i] = m->b1_mant[m->b1ptr++];
                break;

            case 2:
                if(m->b2ptr > 2) {
                    gcode = get_bits(gbc, 7);
                    m->b2_mant[0] = b2_mantissas[gcode][0];
                    m->b2_mant[1] = b2_mantissas[gcode][1];
                    m->b2_mant[2] = b2_mantissas[gcode][2];
                    m->b2ptr = 0;
                }
                coeffs[i] = m->b2_mant[m->b2ptr++];
                break;

            case 3:
                coeffs[i] = b3_mantissas[get_bits(gbc, 3)];
                break;

            case 4:
                if(m->b4ptr > 1) {
                    gcode = get_bits(gbc, 7);
                    m->b4_mant[0] = b4_mantissas[gcode][0];
                    m->b4_mant[1] = b4_mantissas[gcode][1];
                    m->b4ptr = 0;
                }
                coeffs[i] = m->b4_mant[m->b4ptr++];
                break;

            case 5:
                coeffs[i] = b5_mantissas[get_bits(gbc, 4)];
                break;

            default: {
                /* asymmetric dequantization */
                int qlevel = quantization_tab[tbap];
                coeffs[i] = get_sbits(gbc, qlevel) << (24 - qlevel);
                break;
            }
        }
        coeffs[i] >>= exps[i];
    }
}

/**
 * Remove random dithering from coefficients with zero-bit mantissas
 * reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0)
 */
static void remove_dithering(AC3DecodeContext *s) {
    int ch, i;
    int end=0;
    int *coeffs;
    uint8_t *bap;

    for(ch=1; ch<=s->fbw_channels; ch++) {
        if(!s->dither_flag[ch]) {
            coeffs = s->fixed_coeffs[ch];
            bap = s->bap[ch];
            if(s->channel_in_cpl[ch])
                end = s->start_freq[CPL_CH];
            else
                end = s->end_freq[ch];
            for(i=0; 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);
        s->cpl_in_use[blk] = get_bits1(gbc);
        if (s->cpl_in_use[blk]) {
            /* 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 {
        s->cpl_in_use[blk] = s->cpl_in_use[blk-1];
    }
    cpl_in_use = s->cpl_in_use[blk];

    /* coupling coordinates */
    if (cpl_in_use) {
        int cpl_coords_exist = 0;

        for (ch = 1; ch <= fbw_channels; ch++) {
            if (s->channel_in_cpl[ch]) {
                if (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"),
};