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

static int dca_subsubframe(DCAContext * s)
{
    int k, l;
    int subsubframe = s->current_subsubframe;

    float *quant_step_table;

    /* FIXME */
    float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];

    /*
     * Audio data
     */

    /* Select quantization step size table */
    if (s->bit_rate == 0x1f)
        quant_step_table = (float *) lossless_quant_d;
    else
        quant_step_table = (float *) lossy_quant_d;

    for (k = 0; k < s->prim_channels; k++) {
        for (l = 0; l < s->vq_start_subband[k]; l++) {
            int m;

            /* Select the mid-tread linear quantizer */
            int abits = s->bitalloc[k][l];

            float quant_step_size = quant_step_table[abits];
            float rscale;

            /*
             * Determine quantization index code book and its type
             */

            /* Select quantization index code book */
            int sel = s->quant_index_huffman[k][abits];

            /*
             * Extract bits from the bit stream
             */
            if(!abits){
                memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
            }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
                if(abits <= 7){
                    /* Block code */
                    int block_code1, block_code2, size, levels;
                    int block[8];

                    size = abits_sizes[abits-1];
                    levels = abits_levels[abits-1];

                    block_code1 = get_bits(&s->gb, size);
                    /* FIXME Should test return value */
                    decode_blockcode(block_code1, levels, block);
                    block_code2 = get_bits(&s->gb, size);
                    decode_blockcode(block_code2, levels, &block[4]);
                    for (m = 0; m < 8; m++)
                        subband_samples[k][l][m] = block[m];
                }else{
                    /* no coding */
                    for (m = 0; m < 8; m++)
                        subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
                }
            }else{
                /* Huffman coded */
                for (m = 0; m < 8; m++)
                    subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
            }

            /* Deal with transients */
            if (s->transition_mode[k][l] &&
                subsubframe >= s->transition_mode[k][l])
                rscale = quant_step_size * s->scale_factor[k][l][1];
            else
                rscale = quant_step_size * s->scale_factor[k][l][0];

            rscale *= s->scalefactor_adj[k][sel];

            for (m = 0; m < 8; m++)
                subband_samples[k][l][m] *= rscale;

            /*
             * Inverse ADPCM if in prediction mode
             */
            if (s->prediction_mode[k][l]) {
                int n;
                for (m = 0; m < 8; m++) {
                    for (n = 1; n <= 4; n++)
                        if (m >= n)
                            subband_samples[k][l][m] +=
                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
                                 subband_samples[k][l][m - n] / 8192);
                        else if (s->predictor_history)
                            subband_samples[k][l][m] +=
                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
                                 s->subband_samples_hist[k][l][m - n +
                                                               4] / 8192);
                }
            }
        }

        /*
         * Decode VQ encoded high frequencies
         */
        for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
            /* 1 vector -> 32 samples but we only need the 8 samples
             * for this subsubframe. */
            int m;

            if (!s->debug_flag & 0x01) {
                av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
                s->debug_flag |= 0x01;
            }

            for (m = 0; m < 8; m++) {
                subband_samples[k][l][m] =
                    high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
                                                        m]
                    * (float) s->scale_factor[k][l][0] / 16.0;
            }
        }
    }

    /* Check for DSYNC after subsubframe */
    if (s->aspf || subsubframe == s->subsubframes - 1) {
        if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */
#ifdef TRACE
            av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
#endif
        } else {
            av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
        }
    }

    /* Backup predictor history for adpcm */
    for (k = 0; k < s->prim_channels; k++)
        for (l = 0; l < s->vq_start_subband[k]; l++)
            memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
                        4 * sizeof(subband_samples[0][0][0]));

    /* 32 subbands QMF */
    for (k = 0; k < s->prim_channels; k++) {
/*        static float pcm_to_double[8] =
            {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
         qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k],
                            2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
                            0 /*s->bias */ );
    }

    /* Down mixing */

    if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
        dca_downmix(s->samples, s->amode, s->downmix_coef);
    }

    /* Generate LFE samples for this subsubframe FIXME!!! */
    if (s->output & DCA_LFE) {
        int lfe_samples = 2 * s->lfe * s->subsubframes;
        int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];

        lfe_interpolation_fir(s->lfe, 2 * s->lfe,
                              s->lfe_data + lfe_samples +
                              2 * s->lfe * subsubframe,
                              &s->samples[256 * i_channels],
                              8388608.0, s->bias);
        /* Outputs 20bits pcm samples */
    }

    return 0;
}


static int dca_subframe_footer(DCAContext * s)
{
    int aux_data_count = 0, i;
    int lfe_samples;

    /*
     * Unpack optional information
     */

    if (s->timestamp)
        get_bits(&s->gb, 32);

    if (s->aux_data)
        aux_data_count = get_bits(&s->gb, 6);

    for (i = 0; i < aux_data_count; i++)
        get_bits(&s->gb, 8);

    if (s->crc_present && (s->downmix || s->dynrange))
        get_bits(&s->gb, 16);

    lfe_samples = 2 * s->lfe * s->subsubframes;
    for (i = 0; i < lfe_samples; i++) {
        s->lfe_data[i] = s->lfe_data[i + lfe_samples];
    }

    return 0;
}

/**
 * Decode a dca frame block
 *
 * @param s     pointer to the DCAContext
 */

static int dca_decode_block(DCAContext * s)
{

    /* Sanity check */
    if (s->current_subframe >= s->subframes) {
        av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
               s->current_subframe, s->subframes);
        return -1;
    }

    if (!s->current_subsubframe) {
#ifdef TRACE
        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
        /* Read subframe header */
        if (dca_subframe_header(s))
            return -1;
    }

    /* Read subsubframe */
#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
    if (dca_subsubframe(s))
        return -1;

    /* Update state */
    s->current_subsubframe++;
    if (s->current_subsubframe >= s->subsubframes) {
        s->current_subsubframe = 0;
        s->current_subframe++;
    }
    if (s->current_subframe >= s->subframes) {
#ifdef TRACE
        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
        /* Read subframe footer */
        if (dca_subframe_footer(s))
            return -1;
    }

    return 0;
}

/**
 * Convert bitstream to one representation based on sync marker
 */
static int dca_convert_bitstream(uint8_t * src, int src_size, uint8_t * dst,
                          int max_size)
{
    uint32_t mrk;
    int i, tmp;
    uint16_t *ssrc = (uint16_t *) src, *sdst = (uint16_t *) dst;
    PutBitContext pb;

    if((unsigned)src_size > (unsigned)max_size) {
        av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
        return -1;
    }

    mrk = AV_RB32(src);
    switch (mrk) {
    case DCA_MARKER_RAW_BE:
        memcpy(dst, src, FFMIN(src_size, max_size));
        return FFMIN(src_size, max_size);
    case DCA_MARKER_RAW_LE:
        for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++)
            *sdst++ = bswap_16(*ssrc++);
        return FFMIN(src_size, max_size);
    case DCA_MARKER_14B_BE:
    case DCA_MARKER_14B_LE:
        init_put_bits(&pb, dst, max_size);
        for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
            tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
            put_bits(&pb, 14, tmp);
        }
        flush_put_bits(&pb);
        return (put_bits_count(&pb) + 7) >> 3;
    default:
        return -1;
    }
}

/**
 * Main frame decoding function
 * FIXME add arguments
 */
static int dca_decode_frame(AVCodecContext * avctx,
                            void *data, int *data_size,
                            uint8_t * buf, int buf_size)
{

    int i, j, k;
    int16_t *samples = data;
    DCAContext *s = avctx->priv_data;
    int channels;


    s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
    if (s->dca_buffer_size == -1) {
        av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
        return -1;
    }

    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
    if (dca_parse_frame_header(s) < 0) {
        //seems like the frame is corrupt, try with the next one
        *data_size=0;
        return buf_size;
    }
    //set AVCodec values with parsed data
    avctx->sample_rate = s->sample_rate;
    avctx->bit_rate = s->bit_rate;

    channels = s->prim_channels + !!s->lfe;
    avctx->channels = avctx->request_channels;
    if(avctx->channels == 0) {
        avctx->channels = channels;
    } else if(channels < avctx->channels) {
        av_log(avctx, AV_LOG_WARNING, "DTS source channels are less than "
               "specified: output to %d channels.\n", channels);
        avctx->channels = channels;
    }
    if(avctx->channels == 2) {
        s->output = DCA_STEREO;
    } else if(avctx->channels != channels) {
        av_log(avctx, AV_LOG_ERROR, "Cannot downmix DTS to %d channels.\n",
               avctx->channels);
        return -1;
    }

    channels = avctx->channels;
    if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
        return -1;
    *data_size = 0;
    for (i = 0; i < (s->sample_blocks / 8); i++) {
        dca_decode_block(s);
        s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
        /* interleave samples */
        for (j = 0; j < 256; j++) {
            for (k = 0; k < channels; k++)
                samples[k] = s->tsamples[j + k * 256];
            samples += channels;
        }
        *data_size += 256 * sizeof(int16_t) * channels;
    }

    return buf_size;
}



/**
 * Build the cosine modulation tables for the QMF
 *
 * @param s     pointer to the DCAContext
 */

static void pre_calc_cosmod(DCAContext * s)
{
    int i, j, k;
    static int cosmod_inited = 0;

    if(cosmod_inited) return;
    for (j = 0, k = 0; k < 16; k++)
        for (i = 0; i < 16; i++)
            cos_mod[j++] = cos((2 * i + 1) * (2 * k + 1) * M_PI / 64);

    for (k = 0; k < 16; k++)
        for (i = 0; i < 16; i++)
            cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);

    for (k = 0; k < 16; k++)
        cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));

    for (k = 0; k < 16; k++)
        cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));

    cosmod_inited = 1;
}


/**
 * DCA initialization
 *
 * @param avctx     pointer to the AVCodecContext
 */

static int dca_decode_init(AVCodecContext * avctx)
{
    DCAContext *s = avctx->priv_data;

    s->avctx = avctx;
    dca_init_vlcs();
    pre_calc_cosmod(s);

    dsputil_init(&s->dsp, avctx);
    return 0;
}


AVCodec dca_decoder = {
    .name = "dca",
    .type = CODEC_TYPE_AUDIO,
    .id = CODEC_ID_DTS,
    .priv_data_size = sizeof(DCAContext),
    .init = dca_decode_init,
    .decode = dca_decode_frame,
};