flac.c

从FFMPEG转换而来的H264解码程序,VC下编译..

}

static inline int decode_subframe(FLACContext *s, int channel)
{
    int type, wasted = 0;
    int i, tmp;

    s->curr_bps = s->bps;
    if(channel == 0){
        if(s->decorrelation == RIGHT_SIDE)
            s->curr_bps++;
    }else{
        if(s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE)
            s->curr_bps++;
    }

    if (get_bits1(&s->gb))
    {
        av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
        return -1;
    }
    type = get_bits(&s->gb, 6);
//    wasted = get_bits1(&s->gb);

//    if (wasted)
//    {
//        while (!get_bits1(&s->gb))
//            wasted++;
//        if (wasted)
//            wasted++;
//        s->curr_bps -= wasted;
//    }
#if 0
    wasted= 16 - av_log2(show_bits(&s->gb, 17));
    skip_bits(&s->gb, wasted+1);
    s->curr_bps -= wasted;
#else
    if (get_bits1(&s->gb))
    {
        wasted = 1;
        while (!get_bits1(&s->gb))
            wasted++;
        s->curr_bps -= wasted;
        av_log(s->avctx, AV_LOG_DEBUG, "%d wasted bits\n", wasted);
    }
#endif
//FIXME use av_log2 for types
    if (type == 0)
    {
        av_log(s->avctx, AV_LOG_DEBUG, "coding type: constant\n");
        tmp = get_sbits(&s->gb, s->curr_bps);
        for (i = 0; i < s->blocksize; i++)
            s->decoded[channel][i] = tmp;
    }
    else if (type == 1)
    {
        av_log(s->avctx, AV_LOG_DEBUG, "coding type: verbatim\n");
        for (i = 0; i < s->blocksize; i++)
            s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
    }
    else if ((type >= 8) && (type <= 12))
    {
//        av_log(s->avctx, AV_LOG_DEBUG, "coding type: fixed\n");
        if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
            return -1;
    }
    else if (type >= 32)
    {
//        av_log(s->avctx, AV_LOG_DEBUG, "coding type: lpc\n");
        if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
            return -1;
    }
    else
    {
        av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
        return -1;
    }

    if (wasted)
    {
        int i;
        for (i = 0; i < s->blocksize; i++)
            s->decoded[channel][i] <<= wasted;
    }

    return 0;
}

static int decode_frame(FLACContext *s)
{
    int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8;
    int decorrelation, bps, blocksize, samplerate;

    blocksize_code = get_bits(&s->gb, 4);

    sample_rate_code = get_bits(&s->gb, 4);

    assignment = get_bits(&s->gb, 4); /* channel assignment */
    if (assignment < 8 && s->channels == assignment+1)
        decorrelation = INDEPENDENT;
    else if (assignment >=8 && assignment < 11 && s->channels == 2)
        decorrelation = LEFT_SIDE + assignment - 8;
    else
    {
        av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels);
        return -1;
    }

    sample_size_code = get_bits(&s->gb, 3);
    if(sample_size_code == 0)
        bps= s->bps;
    else if((sample_size_code != 3) && (sample_size_code != 7))
        bps = sample_size_table[sample_size_code];
    else
    {
        av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n", sample_size_code);
        return -1;
    }

    if (get_bits1(&s->gb))
    {
        av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
        return -1;
    }

    if(get_utf8(&s->gb) < 0){
        av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
        return -1;
    }
#if 0
    if (/*((blocksize_code == 6) || (blocksize_code == 7)) &&*/
        (s->min_blocksize != s->max_blocksize)){
    }else{
    }
#endif

    if (blocksize_code == 0)
        blocksize = s->min_blocksize;
    else if (blocksize_code == 6)
        blocksize = get_bits(&s->gb, 8)+1;
    else if (blocksize_code == 7)
        blocksize = get_bits(&s->gb, 16)+1;
    else
        blocksize = blocksize_table[blocksize_code];

    if(blocksize > s->max_blocksize){
        av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize, s->max_blocksize);
        return -1;
    }

    if (sample_rate_code == 0){
        samplerate= s->samplerate;
    }else if ((sample_rate_code > 3) && (sample_rate_code < 12))
        samplerate = sample_rate_table[sample_rate_code];
    else if (sample_rate_code == 12)
        samplerate = get_bits(&s->gb, 8) * 1000;
    else if (sample_rate_code == 13)
        samplerate = get_bits(&s->gb, 16);
    else if (sample_rate_code == 14)
        samplerate = get_bits(&s->gb, 16) * 10;
    else{
        av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n", sample_rate_code);
        return -1;
    }

    skip_bits(&s->gb, 8);
    crc8= av_crc(av_crc07, 0, s->gb.buffer, get_bits_count(&s->gb)/8);
    if(crc8){
        av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8);
        return -1;
    }

    s->blocksize    = blocksize;
    s->samplerate   = samplerate;
    s->bps          = bps;
    s->decorrelation= decorrelation;

//    dump_headers(s);

    /* subframes */
    for (i = 0; i < s->channels; i++)
    {
//        av_log(s->avctx, AV_LOG_DEBUG, "decoded: %x residual: %x\n", s->decoded[i], s->residual[i]);
        if (decode_subframe(s, i) < 0)
            return -1;
    }

    align_get_bits(&s->gb);

    /* frame footer */
    skip_bits(&s->gb, 16); /* data crc */

    return 0;
}

static int flac_decode_frame(AVCodecContext *avctx,
                            void *data, int *data_size,
                            uint8_t *buf, int buf_size)
{
    FLACContext *s = avctx->priv_data;
    int tmp = 0, i, j = 0, input_buf_size = 0;
    int16_t *samples = data;

    if(s->max_framesize == 0){
        s->max_framesize= 65536; // should hopefully be enough for the first header
        s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
    }

    if(1 && s->max_framesize){//FIXME truncated
            buf_size= FFMAX(FFMIN(buf_size, s->max_framesize - s->bitstream_size), 0);
            input_buf_size= buf_size;

            if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
//                printf("memmove\n");
                memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
                s->bitstream_index=0;
            }
            memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size);
            buf= &s->bitstream[s->bitstream_index];
            buf_size += s->bitstream_size;
            s->bitstream_size= buf_size;

            if(buf_size < s->max_framesize){
//                printf("wanna more data ...\n");
                return input_buf_size;
            }
    }

    init_get_bits(&s->gb, buf, buf_size*8);

    if (!metadata_parse(s))
    {
        tmp = show_bits(&s->gb, 16);
        if(tmp != 0xFFF8){
            av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
            while(get_bits_count(&s->gb)/8+2 < buf_size && show_bits(&s->gb, 16) != 0xFFF8)
                skip_bits(&s->gb, 8);
            goto end; // we may not have enough bits left to decode a frame, so try next time
        }
        skip_bits(&s->gb, 16);
        if (decode_frame(s) < 0){
            av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
            s->bitstream_size=0;
            s->bitstream_index=0;
            return -1;
        }
    }


#if 0
    /* fix the channel order here */
    if (s->order == MID_SIDE)
    {
        short *left = samples;
        short *right = samples + s->blocksize;
        for (i = 0; i < s->blocksize; i += 2)
        {
            uint32_t x = s->decoded[0][i];
            uint32_t y = s->decoded[0][i+1];

            right[i] = x - (y / 2);
            left[i] = right[i] + y;
        }
        *data_size = 2 * s->blocksize;
    }
    else
    {
    for (i = 0; i < s->channels; i++)
    {
        switch(s->order)
        {
            case INDEPENDENT:
                for (j = 0; j < s->blocksize; j++)
                    samples[(s->blocksize*i)+j] = s->decoded[i][j];
                break;
            case LEFT_SIDE:
            case RIGHT_SIDE:
                if (i == 0)
                    for (j = 0; j < s->blocksize; j++)
                        samples[(s->blocksize*i)+j] = s->decoded[0][j];
                else
                    for (j = 0; j < s->blocksize; j++)
                        samples[(s->blocksize*i)+j] = s->decoded[0][j] - s->decoded[i][j];
                break;
//            case MID_SIDE:
//                av_log(s->avctx, AV_LOG_DEBUG, "mid-side unsupported\n");
        }
        *data_size += s->blocksize;
    }
    }
#else
#define DECORRELATE(left, right)\
            assert(s->channels == 2);\
            for (i = 0; i < s->blocksize; i++)\
            {\
                int a= s->decoded[0][i];\
                int b= s->decoded[1][i];\
                *samples++ = ((left)  << (24 - s->bps)) >> 8;\
                *samples++ = ((right) << (24 - s->bps)) >> 8;\
            }\
            break;

    switch(s->decorrelation)
    {
        case INDEPENDENT:
            for (j = 0; j < s->blocksize; j++)
            {
                for (i = 0; i < s->channels; i++)
                    *samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8;
            }
            break;
        case LEFT_SIDE:
            DECORRELATE(a,a-b)
        case RIGHT_SIDE:
            DECORRELATE(a+b,b)
        case MID_SIDE:
            DECORRELATE( (a-=b>>1) + b, a)
    }
#endif

    *data_size = (int8_t *)samples - (int8_t *)data;
//    av_log(s->avctx, AV_LOG_DEBUG, "data size: %d\n", *data_size);

//    s->last_blocksize = s->blocksize;
end:
    i= (get_bits_count(&s->gb)+7)/8;;
    if(i > buf_size){
        av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
        s->bitstream_size=0;
        s->bitstream_index=0;
        return -1;
    }

    if(s->bitstream_size){
        s->bitstream_index += i;
        s->bitstream_size  -= i;
        return input_buf_size;
    }else
        return i;
}

static int flac_decode_close(AVCodecContext *avctx)
{
    FLACContext *s = avctx->priv_data;
    int i;

    for (i = 0; i < s->channels; i++)
    {
        av_freep(&s->decoded[i]);
    }
    av_freep(&s->bitstream);

    return 0;
}

static void flac_flush(AVCodecContext *avctx){
    FLACContext *s = avctx->priv_data;

    s->bitstream_size=
    s->bitstream_index= 0;
}

AVCodec flac_decoder = {
    "flac",
    CODEC_TYPE_AUDIO,
    CODEC_ID_FLAC,
    sizeof(FLACContext),
    /*.init=*/flac_decode_init,
    /*.encode=*/NULL,
    /*.close=*/flac_decode_close,
    /*.decode=*/flac_decode_frame,
    /*.capabilities=*/0,
    /*.next=*/NULL,
    /*.flush=*/flac_flush,
};