dca.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

/*
 * DCA compatible decoder
 * Copyright (C) 2004 Gildas Bazin
 * Copyright (C) 2004 Benjamin Zores
 * Copyright (C) 2006 Benjamin Larsson
 * Copyright (C) 2007 Konstantin Shishkov
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 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
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser 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
 */

/**
 * @file dca.c
 */

#include <math.h>
#include <stddef.h>
#include <stdio.h>

#include "avcodec.h"
#include "dsputil.h"
#include "bitstream.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dca.h"

//#define TRACE

#define DCA_PRIM_CHANNELS_MAX (5)
#define DCA_SUBBANDS (32)
#define DCA_ABITS_MAX (32)      /* Should be 28 */
#define DCA_SUBSUBFAMES_MAX (4)
#define DCA_LFE_MAX (3)

enum DCAMode {
    DCA_MONO = 0,
    DCA_CHANNEL,
    DCA_STEREO,
    DCA_STEREO_SUMDIFF,
    DCA_STEREO_TOTAL,
    DCA_3F,
    DCA_2F1R,
    DCA_3F1R,
    DCA_2F2R,
    DCA_3F2R,
    DCA_4F2R
};

#define DCA_DOLBY 101           /* FIXME */

#define DCA_CHANNEL_BITS 6
#define DCA_CHANNEL_MASK 0x3F

#define DCA_LFE 0x80

#define HEADER_SIZE 14
#define CONVERT_BIAS 384

#define DCA_MAX_FRAME_SIZE 16383

/** Bit allocation */
typedef struct {
    int offset;                 ///< code values offset
    int maxbits[8];             ///< max bits in VLC
    int wrap;                   ///< wrap for get_vlc2()
    VLC vlc[8];                 ///< actual codes
} BitAlloc;

static BitAlloc dca_bitalloc_index;    ///< indexes for samples VLC select
static BitAlloc dca_tmode;             ///< transition mode VLCs
static BitAlloc dca_scalefactor;       ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs

/** Pre-calculated cosine modulation coefs for the QMF */
static float cos_mod[544];

static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
{
    return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
}

typedef struct {
    AVCodecContext *avctx;
    /* Frame header */
    int frame_type;             ///< type of the current frame
    int samples_deficit;        ///< deficit sample count
    int crc_present;            ///< crc is present in the bitstream
    int sample_blocks;          ///< number of PCM sample blocks
    int frame_size;             ///< primary frame byte size
    int amode;                  ///< audio channels arrangement
    int sample_rate;            ///< audio sampling rate
    int bit_rate;               ///< transmission bit rate

    int downmix;                ///< embedded downmix enabled
    int dynrange;               ///< embedded dynamic range flag
    int timestamp;              ///< embedded time stamp flag
    int aux_data;               ///< auxiliary data flag
    int hdcd;                   ///< source material is mastered in HDCD
    int ext_descr;              ///< extension audio descriptor flag
    int ext_coding;             ///< extended coding flag
    int aspf;                   ///< audio sync word insertion flag
    int lfe;                    ///< low frequency effects flag
    int predictor_history;      ///< predictor history flag
    int header_crc;             ///< header crc check bytes
    int multirate_inter;        ///< multirate interpolator switch
    int version;                ///< encoder software revision
    int copy_history;           ///< copy history
    int source_pcm_res;         ///< source pcm resolution
    int front_sum;              ///< front sum/difference flag
    int surround_sum;           ///< surround sum/difference flag
    int dialog_norm;            ///< dialog normalisation parameter

    /* Primary audio coding header */
    int subframes;              ///< number of subframes
    int prim_channels;          ///< number of primary audio channels
    int subband_activity[DCA_PRIM_CHANNELS_MAX];    ///< subband activity count
    int vq_start_subband[DCA_PRIM_CHANNELS_MAX];    ///< high frequency vq start subband
    int joint_intensity[DCA_PRIM_CHANNELS_MAX];     ///< joint intensity coding index
    int transient_huffman[DCA_PRIM_CHANNELS_MAX];   ///< transient mode code book
    int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
    int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX];    ///< bit allocation quantizer select
    int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
    float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX];   ///< scale factor adjustment

    /* Primary audio coding side information */
    int subsubframes;           ///< number of subsubframes
    int partial_samples;        ///< partial subsubframe samples count
    int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< prediction mode (ADPCM used or not)
    int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];      ///< prediction VQ coefs
    int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];           ///< bit allocation index
    int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< transition mode (transients)
    int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];    ///< scale factors (2 if transient)
    int joint_huff[DCA_PRIM_CHANNELS_MAX];                       ///< joint subband scale factors codebook
    int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
    int downmix_coef[DCA_PRIM_CHANNELS_MAX][2];                  ///< stereo downmix coefficients
    int dynrange_coef;                                           ///< dynamic range coefficient

    int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];       ///< VQ encoded high frequency subbands

    float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
                   2 /*history */ ];    ///< Low frequency effect data
    int lfe_scale_factor;

    /* Subband samples history (for ADPCM) */
    float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
    float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
    float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];

    int output;                 ///< type of output
    int bias;                   ///< output bias

    DECLARE_ALIGNED_16(float, samples[1536]);  /* 6 * 256 = 1536, might only need 5 */
    DECLARE_ALIGNED_16(int16_t, tsamples[1536]);

    uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
    int dca_buffer_size;        ///< how much data is in the dca_buffer

    GetBitContext gb;
    /* Current position in DCA frame */
    int current_subframe;
    int current_subsubframe;

    int debug_flag;             ///< used for suppressing repeated error messages output
    DSPContext dsp;
} DCAContext;

static void dca_init_vlcs(void)
{
    static int vlcs_inited = 0;
    int i, j;

    if (vlcs_inited)
        return;

    dca_bitalloc_index.offset = 1;
    dca_bitalloc_index.wrap = 2;
    for (i = 0; i < 5; i++)
        init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
                 bitalloc_12_bits[i], 1, 1,
                 bitalloc_12_codes[i], 2, 2, 1);
    dca_scalefactor.offset = -64;
    dca_scalefactor.wrap = 2;
    for (i = 0; i < 5; i++)
        init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
                 scales_bits[i], 1, 1,
                 scales_codes[i], 2, 2, 1);
    dca_tmode.offset = 0;
    dca_tmode.wrap = 1;
    for (i = 0; i < 4; i++)
        init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
                 tmode_bits[i], 1, 1,
                 tmode_codes[i], 2, 2, 1);

    for(i = 0; i < 10; i++)
        for(j = 0; j < 7; j++){
            if(!bitalloc_codes[i][j]) break;
            dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
            dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
            init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
                     bitalloc_sizes[i],
                     bitalloc_bits[i][j], 1, 1,
                     bitalloc_codes[i][j], 2, 2, 1);
        }
    vlcs_inited = 1;
}

static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
{
    while(len--)
        *dst++ = get_bits(gb, bits);
}

static int dca_parse_frame_header(DCAContext * s)
{
    int i, j;
    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
    static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };

    s->bias = CONVERT_BIAS;

    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

    /* Sync code */
    get_bits(&s->gb, 32);

    /* Frame header */
    s->frame_type        = get_bits(&s->gb, 1);
    s->samples_deficit   = get_bits(&s->gb, 5) + 1;
    s->crc_present       = get_bits(&s->gb, 1);
    s->sample_blocks     = get_bits(&s->gb, 7) + 1;
    s->frame_size        = get_bits(&s->gb, 14) + 1;
    if (s->frame_size < 95)
        return -1;
    s->amode             = get_bits(&s->gb, 6);
    s->sample_rate       = dca_sample_rates[get_bits(&s->gb, 4)];
    if (!s->sample_rate)
        return -1;
    s->bit_rate          = dca_bit_rates[get_bits(&s->gb, 5)];
    if (!s->bit_rate)
        return -1;

    s->downmix           = get_bits(&s->gb, 1);
    s->dynrange          = get_bits(&s->gb, 1);
    s->timestamp         = get_bits(&s->gb, 1);
    s->aux_data          = get_bits(&s->gb, 1);
    s->hdcd              = get_bits(&s->gb, 1);
    s->ext_descr         = get_bits(&s->gb, 3);
    s->ext_coding        = get_bits(&s->gb, 1);
    s->aspf              = get_bits(&s->gb, 1);
    s->lfe               = get_bits(&s->gb, 2);
    s->predictor_history = get_bits(&s->gb, 1);

    /* TODO: check CRC */
    if (s->crc_present)
        s->header_crc    = get_bits(&s->gb, 16);

    s->multirate_inter   = get_bits(&s->gb, 1);
    s->version           = get_bits(&s->gb, 4);
    s->copy_history      = get_bits(&s->gb, 2);
    s->source_pcm_res    = get_bits(&s->gb, 3);
    s->front_sum         = get_bits(&s->gb, 1);
    s->surround_sum      = get_bits(&s->gb, 1);
    s->dialog_norm       = get_bits(&s->gb, 4);

    /* FIXME: channels mixing levels */
    s->output = s->amode;
    if(s->lfe) s->output |= DCA_LFE;

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
    av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
    av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
    av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
           s->sample_blocks, s->sample_blocks * 32);
    av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
    av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
           s->amode, dca_channels[s->amode]);
    av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n",
           s->sample_rate, dca_sample_rates[s->sample_rate]);
    av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n",
           s->bit_rate, dca_bit_rates[s->bit_rate]);
    av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
    av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
    av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
    av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
    av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
    av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
    av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
    av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
    av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
    av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
           s->predictor_history);
    av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
    av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
           s->multirate_inter);
    av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
    av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
    av_log(s->avctx, AV_LOG_DEBUG,
           "source pcm resolution: %i (%i bits/sample)\n",
           s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
    av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
    av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
    av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
    av_log(s->avctx, AV_LOG_DEBUG, "\n");
#endif

    /* Primary audio coding header */
    s->subframes         = get_bits(&s->gb, 4) + 1;
    s->prim_channels     = get_bits(&s->gb, 3) + 1;


    for (i = 0; i < s->prim_channels; i++) {
        s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
        if (s->subband_activity[i] > DCA_SUBBANDS)
            s->subband_activity[i] = DCA_SUBBANDS;
    }
    for (i = 0; i < s->prim_channels; i++) {
        s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
        if (s->vq_start_subband[i] > DCA_SUBBANDS)
            s->vq_start_subband[i] = DCA_SUBBANDS;
    }
    get_array(&s->gb, s->joint_intensity,     s->prim_channels, 3);
    get_array(&s->gb, s->transient_huffman,   s->prim_channels, 2);
    get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
    get_array(&s->gb, s->bitalloc_huffman,    s->prim_channels, 3);

    /* Get codebooks quantization indexes */
    memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
    for (j = 1; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);

    /* Get scale factor adjustment */
    for (j = 0; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            s->scalefactor_adj[i][j] = 1;

    for (j = 1; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            if (s->quant_index_huffman[i][j] < thr[j])
                s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];

    if (s->crc_present) {
        /* Audio header CRC check */
        get_bits(&s->gb, 16);
    }

    s->current_subframe = 0;
    s->current_subsubframe = 0;

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
    av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
    for(i = 0; i < s->prim_channels; i++){
        av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
        for (j = 0; j < 11; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %i",
                   s->quant_index_huffman[i][j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
        av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
        for (j = 0; j < 11; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
#endif

    return 0;
}


static inline int get_scale(GetBitContext *gb, int level, int value)
{
   if (level < 5) {
       /* huffman encoded */
       value += get_bitalloc(gb, &dca_scalefactor, level);
   } else if(level < 8)
       value = get_bits(gb, level + 1);
   return value;
}

static int dca_subframe_header(DCAContext * s)
{
    /* Primary audio coding side information */
    int j, k;

    s->subsubframes = get_bits(&s->gb, 2) + 1;
    s->partial_samples = get_bits(&s->gb, 3);
    for (j = 0; j < s->prim_channels; j++) {
        for (k = 0; k < s->subband_activity[j]; k++)
            s->prediction_mode[j][k] = get_bits(&s->gb, 1);
    }

    /* Get prediction codebook */
    for (j = 0; j < s->prim_channels; j++) {
        for (k = 0; k < s->subband_activity[j]; k++) {
            if (s->prediction_mode[j][k] > 0) {
                /* (Prediction coefficient VQ address) */
                s->prediction_vq[j][k] = get_bits(&s->gb, 12);
            }