dca.c

ffmpeg移植到symbian的全部源代码

     * Audio data
     */

    /* Select quantization step size table */
    if (s->bit_rate == 0x1f)
        quant_step_table = lossless_quant_d;
    else
        quant_step_table = 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],
                              256.0, 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(const uint8_t * src, int src_size, uint8_t * dst,
                          int max_size)
{
    uint32_t mrk;
    int i, tmp;
    const uint16_t *ssrc = (const uint16_t *) src;
    uint16_t *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,
                            const 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;
    if(avctx->request_channels == 2 && s->prim_channels > 2) {
        channels = 2;
        s->output = DCA_STEREO;
    }

    /* There is nothing that prevents a dts frame to change channel configuration
       but FFmpeg doesn't support that so only set the channels if it is previously
       unset. Ideally during the first probe for channels the crc should be checked
       and only set avctx->channels when the crc is ok. Right now the decoder could
       set the channels based on a broken first frame.*/
    if (!avctx->channels)
        avctx->channels = 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 av_cold void pre_calc_cosmod(DCAContext * s)
{
    int i, j, k;
    static int cosmod_initialized = 0;

    if(cosmod_initialized) 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_initialized = 1;
}


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

static av_cold 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);

    /* allow downmixing to stereo */
    if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
            avctx->request_channels == 2) {
        avctx->channels = avctx->request_channels;
    }

    return 0;
}


AVCodec dca_decoder = {
#ifdef __CW32__
    "dca",
    CODEC_TYPE_AUDIO,
    CODEC_ID_DTS,
    sizeof(DCAContext),
    dca_decode_init,
    0,
    0,
    dca_decode_frame,
    0,
    0,
    0,
    0,
    0,
    NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
#else
    .name = "dca",
    .type = CODEC_TYPE_AUDIO,
    .id = CODEC_ID_DTS,
    .priv_data_size = sizeof(DCAContext),
    .init = dca_decode_init,
    .decode = dca_decode_frame,
    .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
#endif
};