atrac3.c

ffmpeg移植到symbian的全部源代码

    }

    /* number of coded QMF bands */
    pSnd->bandsCoded = get_bits(gb,2);

    result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
    if (result) return result;

    pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
    if (pSnd->numComponents == -1) return -1;

    numSubbands = decodeSpectrum (gb, pSnd->spectrum);

    /* Merge the decoded spectrum and tonal components. */
    lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);


    /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
    numBands = (subbandTab[numSubbands] - 1) >> 8;
    if (lastTonal >= 0)
        numBands = FFMAX((lastTonal + 256) >> 8, numBands);


    /* Reconstruct time domain samples. */
    for (band=0; band<4; band++) {
        /* Perform the IMDCT step without overlapping. */
        if (band <= numBands) {
            IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1,q->mdct_tmp);
        } else
            memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));

        /* gain compensation and overlapping */
        gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
                                    &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
                                    &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
    }

    /* Swap the gain control buffers for the next frame. */
    pSnd->gcBlkSwitch ^= 1;

    return 0;
}

/**
 * Frame handling
 *
 * @param q             Atrac3 private context
 * @param databuf       the input data
 */

static int decodeFrame(ATRAC3Context *q, uint8_t* databuf)
{
    int   result, i;
    float   *p1, *p2, *p3, *p4;
    uint8_t    *ptr1, *ptr2;

    if (q->codingMode == JOINT_STEREO) {

        /* channel coupling mode */
        /* decode Sound Unit 1 */
        init_get_bits(&q->gb,databuf,q->bits_per_frame);

        result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
        if (result != 0)
            return (result);

        /* Framedata of the su2 in the joint-stereo mode is encoded in
         * reverse byte order so we need to swap it first. */
        ptr1 = databuf;
        ptr2 = databuf+q->bytes_per_frame-1;
        for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
            FFSWAP(uint8_t,*ptr1,*ptr2);
        }

        /* Skip the sync codes (0xF8). */
        ptr1 = databuf;
        for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
            if (i >= q->bytes_per_frame)
                return -1;
        }


        /* set the bitstream reader at the start of the second Sound Unit*/
        init_get_bits(&q->gb,ptr1,q->bits_per_frame);

        /* Fill the Weighting coeffs delay buffer */
        memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
        q->weighting_delay[4] = get_bits1(&q->gb);
        q->weighting_delay[5] = get_bits(&q->gb,3);

        for (i = 0; i < 4; i++) {
            q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
            q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
            q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
        }

        /* Decode Sound Unit 2. */
        result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
        if (result != 0)
            return (result);

        /* Reconstruct the channel coefficients. */
        reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);

        channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);

    } else {
        /* normal stereo mode or mono */
        /* Decode the channel sound units. */
        for (i=0 ; i<q->channels ; i++) {

            /* Set the bitstream reader at the start of a channel sound unit. */
            init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);

            result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
            if (result != 0)
                return (result);
        }
    }

    /* Apply the iQMF synthesis filter. */
    p1= q->outSamples;
    for (i=0 ; i<q->channels ; i++) {
        p2= p1+256;
        p3= p2+256;
        p4= p3+256;
        iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
        iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
        iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
        p1 +=1024;
    }

    return 0;
}


/**
 * Atrac frame decoding
 *
 * @param avctx     pointer to the AVCodecContext
 */

static int atrac3_decode_frame(AVCodecContext *avctx,
            void *data, int *data_size,
            const uint8_t *buf, int buf_size) {
    ATRAC3Context *q = avctx->priv_data;
    int result = 0, i;
    uint8_t* databuf;
    int16_t* samples = data;

    if (buf_size < avctx->block_align)
        return buf_size;

    /* Check if we need to descramble and what buffer to pass on. */
    if (q->scrambled_stream) {
        decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
        databuf = q->decoded_bytes_buffer;
    } else {
#ifndef __CW32__
        databuf = buf;
#else
        databuf = (uint8_t*)buf;
#endif
    }

    result = decodeFrame(q, databuf);

    if (result != 0) {
        av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
        return -1;
    }

    if (q->channels == 1) {
        /* mono */
        for (i = 0; i<1024; i++)
            samples[i] = av_clip_int16(round(q->outSamples[i]));
        *data_size = 1024 * sizeof(int16_t);
    } else {
        /* stereo */
        for (i = 0; i < 1024; i++) {
            samples[i*2] = av_clip_int16(round(q->outSamples[i]));
            samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
        }
        *data_size = 2048 * sizeof(int16_t);
    }

    return avctx->block_align;
}


/**
 * Atrac3 initialization
 *
 * @param avctx     pointer to the AVCodecContext
 */

static int atrac3_decode_init(AVCodecContext *avctx)
{
    int i;
    const uint8_t *edata_ptr = avctx->extradata;
    ATRAC3Context *q = avctx->priv_data;

    /* Take data from the AVCodecContext (RM container). */
    q->sample_rate = avctx->sample_rate;
    q->channels = avctx->channels;
    q->bit_rate = avctx->bit_rate;
    q->bits_per_frame = avctx->block_align * 8;
    q->bytes_per_frame = avctx->block_align;

    /* Take care of the codec-specific extradata. */
    if (avctx->extradata_size == 14) {
        /* Parse the extradata, WAV format */
        av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown value always 1
        q->samples_per_channel = bytestream_get_le32(&edata_ptr);
        q->codingMode = bytestream_get_le16(&edata_ptr);
        av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
        q->frame_factor = bytestream_get_le16(&edata_ptr);  //Unknown always 1
        av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown always 0

        /* setup */
        q->samples_per_frame = 1024 * q->channels;
        q->atrac3version = 4;
        q->delay = 0x88E;
        if (q->codingMode)
            q->codingMode = JOINT_STEREO;
        else
            q->codingMode = STEREO;

        q->scrambled_stream = 0;

        if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) {
        } else {
            av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor);
            return -1;
        }

    } else if (avctx->extradata_size == 10) {
        /* Parse the extradata, RM format. */
        q->atrac3version = bytestream_get_be32(&edata_ptr);
        q->samples_per_frame = bytestream_get_be16(&edata_ptr);
        q->delay = bytestream_get_be16(&edata_ptr);
        q->codingMode = bytestream_get_be16(&edata_ptr);

        q->samples_per_channel = q->samples_per_frame / q->channels;
        q->scrambled_stream = 1;

    } else {
        av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
    }
    /* Check the extradata. */

    if (q->atrac3version != 4) {
        av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
        return -1;
    }

    if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
        return -1;
    }

    if (q->delay != 0x88E) {
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
        return -1;
    }

    if (q->codingMode == STEREO) {
        av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
    } else if (q->codingMode == JOINT_STEREO) {
        av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
    } else {
        av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
        return -1;
    }

    if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
        av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
        return -1;
    }


    if(avctx->block_align >= UINT_MAX/2)
        return -1;

    /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
     * this is for the bitstream reader. */
    if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)))  == NULL)
        return AVERROR(ENOMEM);


    /* Initialize the VLC tables. */
    for (i=0 ; i<7 ; i++) {
        init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
            huff_bits[i], 1, 1,
            huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);
    }

    init_atrac3_transforms(q);

    /* Generate the scale factors. */
    for (i=0 ; i<64 ; i++)
        SFTable[i] = pow(2.0, (i - 15) / 3.0);

    /* Generate gain tables. */
    for (i=0 ; i<16 ; i++)
        gain_tab1[i] = powf (2.0, (4 - i));

    for (i=-15 ; i<16 ; i++)
        gain_tab2[i+15] = powf (2.0, i * -0.125);

    /* init the joint-stereo decoding data */
    q->weighting_delay[0] = 0;
    q->weighting_delay[1] = 7;
    q->weighting_delay[2] = 0;
    q->weighting_delay[3] = 7;
    q->weighting_delay[4] = 0;
    q->weighting_delay[5] = 7;

    for (i=0; i<4; i++) {
        q->matrix_coeff_index_prev[i] = 3;
        q->matrix_coeff_index_now[i] = 3;
        q->matrix_coeff_index_next[i] = 3;
    }

    dsputil_init(&dsp, avctx);

    q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
    if (!q->pUnits) {
        av_free(q->decoded_bytes_buffer);
        return AVERROR(ENOMEM);
    }

    return 0;
}


AVCodec atrac3_decoder =
{
#ifdef __CW32__
	    "atrac3",
	    CODEC_TYPE_AUDIO,
	    CODEC_ID_ATRAC3,
	    sizeof(ATRAC3Context),
	    atrac3_decode_init,
	    0,
	    atrac3_decode_close,
	    atrac3_decode_frame,
	    0,
	    0,
	    0,
	    0,
	    0,
	    NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
#else
	    .name = "atrac3",
	    .type = CODEC_TYPE_AUDIO,
	    .id = CODEC_ID_ATRAC3,
	    .priv_data_size = sizeof(ATRAC3Context),
	    .init = atrac3_decode_init,
	    .close = atrac3_decode_close,
	    .decode = atrac3_decode_frame,
	    .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
#endif
};