annotate flac.c @ 2497:69adfbbdcdeb libavcodec

- samples from mplayer ftp in the "adv" profile seem to have profile=2, which isn't the advanced one; and indeed, using adv. profile parser fails. Using normal parser works, and that's what is done - attempt at taking care of stride for NORM2 bitplane decoding - duplication of much code from msmpeg4.c; this code isn't yet used, but goes down as far as the block layer (mainly Transform Type stuff, the remains are wild editing without checking). Unusable yet, and lacks the AC decoding (but a step further in bitstream parsing) patch by anonymous

/*
 * FLAC (Free Lossless Audio Codec) decoder
 * Copyright (c) 2003 Alex Beregszaszi
 *
 * This library 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 of the License, or (at your option) any later version.
 *
 * This library 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 this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

/**
 * @file flac.c
 * FLAC (Free Lossless Audio Codec) decoder
 * @author Alex Beregszaszi
 *
 * For more information on the FLAC format, visit:
 *  http://flac.sourceforge.net/
 *
 * This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
 * through, starting from the initial 'fLaC' signature; or by passing the
 * 34-byte streaminfo structure through avctx->extradata[_size] followed
 * by data starting with the 0xFFF8 marker.
 */
 
#include <limits.h>
 
#include "avcodec.h"
#include "bitstream.h"
#include "golomb.h"

#undef NDEBUG
#include <assert.h>

#define MAX_CHANNELS 8
#define MAX_BLOCKSIZE 65535
#define FLAC_STREAMINFO_SIZE 34

enum decorrelation_type {
    INDEPENDENT,
    LEFT_SIDE,
    RIGHT_SIDE,
    MID_SIDE,
};

typedef struct FLACContext {
    AVCodecContext *avctx;
    GetBitContext gb;

    int min_blocksize, max_blocksize;
    int min_framesize, max_framesize;
    int samplerate, channels;
    int blocksize/*, last_blocksize*/;
    int bps, curr_bps;
    enum decorrelation_type decorrelation;

    int32_t *decoded[MAX_CHANNELS];
    uint8_t *bitstream;
    int bitstream_size;
    int bitstream_index;
    int allocated_bitstream_size;
} FLACContext;

#define METADATA_TYPE_STREAMINFO 0

static int sample_rate_table[] =
{ 0, 0, 0, 0,
  8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
  0, 0, 0, 0 }; 

static int sample_size_table[] = 
{ 0, 8, 12, 0, 16, 20, 24, 0 };

static int blocksize_table[] = {
     0,    192, 576<<0, 576<<1, 576<<2, 576<<3,      0,      0, 
256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7 
};

static const uint8_t table_crc8[256] = {
    0x00, 0x07, 0x0e, 0x09, 0x1c, 0x1b, 0x12, 0x15,
    0x38, 0x3f, 0x36, 0x31, 0x24, 0x23, 0x2a, 0x2d,
    0x70, 0x77, 0x7e, 0x79, 0x6c, 0x6b, 0x62, 0x65,
    0x48, 0x4f, 0x46, 0x41, 0x54, 0x53, 0x5a, 0x5d,
    0xe0, 0xe7, 0xee, 0xe9, 0xfc, 0xfb, 0xf2, 0xf5,
    0xd8, 0xdf, 0xd6, 0xd1, 0xc4, 0xc3, 0xca, 0xcd,
    0x90, 0x97, 0x9e, 0x99, 0x8c, 0x8b, 0x82, 0x85,
    0xa8, 0xaf, 0xa6, 0xa1, 0xb4, 0xb3, 0xba, 0xbd,
    0xc7, 0xc0, 0xc9, 0xce, 0xdb, 0xdc, 0xd5, 0xd2,
    0xff, 0xf8, 0xf1, 0xf6, 0xe3, 0xe4, 0xed, 0xea,
    0xb7, 0xb0, 0xb9, 0xbe, 0xab, 0xac, 0xa5, 0xa2,
    0x8f, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9d, 0x9a,
    0x27, 0x20, 0x29, 0x2e, 0x3b, 0x3c, 0x35, 0x32,
    0x1f, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0d, 0x0a,
    0x57, 0x50, 0x59, 0x5e, 0x4b, 0x4c, 0x45, 0x42,
    0x6f, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7d, 0x7a,
    0x89, 0x8e, 0x87, 0x80, 0x95, 0x92, 0x9b, 0x9c,
    0xb1, 0xb6, 0xbf, 0xb8, 0xad, 0xaa, 0xa3, 0xa4,
    0xf9, 0xfe, 0xf7, 0xf0, 0xe5, 0xe2, 0xeb, 0xec,
    0xc1, 0xc6, 0xcf, 0xc8, 0xdd, 0xda, 0xd3, 0xd4,
    0x69, 0x6e, 0x67, 0x60, 0x75, 0x72, 0x7b, 0x7c,
    0x51, 0x56, 0x5f, 0x58, 0x4d, 0x4a, 0x43, 0x44,
    0x19, 0x1e, 0x17, 0x10, 0x05, 0x02, 0x0b, 0x0c,
    0x21, 0x26, 0x2f, 0x28, 0x3d, 0x3a, 0x33, 0x34,
    0x4e, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5c, 0x5b,
    0x76, 0x71, 0x78, 0x7f, 0x6a, 0x6d, 0x64, 0x63,
    0x3e, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2c, 0x2b,
    0x06, 0x01, 0x08, 0x0f, 0x1a, 0x1d, 0x14, 0x13,
    0xae, 0xa9, 0xa0, 0xa7, 0xb2, 0xb5, 0xbc, 0xbb,
    0x96, 0x91, 0x98, 0x9f, 0x8a, 0x8d, 0x84, 0x83,
    0xde, 0xd9, 0xd0, 0xd7, 0xc2, 0xc5, 0xcc, 0xcb,
    0xe6, 0xe1, 0xe8, 0xef, 0xfa, 0xfd, 0xf4, 0xf3
};

static int64_t get_utf8(GetBitContext *gb)
{
    uint64_t val;
    int ones=0, bytes;
    
    while(get_bits1(gb))
        ones++;

    if     (ones==0) bytes=0;
    else if(ones==1) return -1;
    else             bytes= ones - 1;
    
    val= get_bits(gb, 7-ones);
    while(bytes--){
        const int tmp = get_bits(gb, 8);
        
        if((tmp>>6) != 2)
            return -1;
        val<<=6;
        val|= tmp&0x3F;
    }
    return val;
}

static int skip_utf8(GetBitContext *gb)
{
    int ones=0, bytes;
    
    while(get_bits1(gb))
        ones++;

    if     (ones==0) bytes=0;
    else if(ones==1) return -1;
    else             bytes= ones - 1;
    
    skip_bits(gb, 7-ones);
    while(bytes--){
        const int tmp = get_bits(gb, 8);
        
        if((tmp>>6) != 2)
            return -1;
    }
    return 0;
}

static int get_crc8(const uint8_t *buf, int count){
    int crc=0;
    int i;
    
    for(i=0; i<count; i++){
        crc = table_crc8[crc ^ buf[i]];
    }

    return crc;
}

static void metadata_streaminfo(FLACContext *s);
static void dump_headers(FLACContext *s);

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

    /* initialize based on the demuxer-supplied streamdata header */
    if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
        init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
        metadata_streaminfo(s);
        dump_headers(s);
    }

    return 0;
}

static void dump_headers(FLACContext *s)
{
    av_log(s->avctx, AV_LOG_DEBUG, "  Blocksize: %d .. %d (%d)\n", s->min_blocksize, s->max_blocksize, s->blocksize);
    av_log(s->avctx, AV_LOG_DEBUG, "  Framesize: %d .. %d\n", s->min_framesize, s->max_framesize);
    av_log(s->avctx, AV_LOG_DEBUG, "  Samplerate: %d\n", s->samplerate);
    av_log(s->avctx, AV_LOG_DEBUG, "  Channels: %d\n", s->channels);
    av_log(s->avctx, AV_LOG_DEBUG, "  Bits: %d\n", s->bps);
}

static void allocate_buffers(FLACContext *s){
    int i;

    assert(s->max_blocksize);

    if(s->max_framesize == 0 && s->max_blocksize){
        s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; //FIXME header overhead
    }

    for (i = 0; i < s->channels; i++)
    {
        s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t)*s->max_blocksize);
    }

    s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
}

static void metadata_streaminfo(FLACContext *s)
{
    /* mandatory streaminfo */
    s->min_blocksize = get_bits(&s->gb, 16);
    s->max_blocksize = get_bits(&s->gb, 16);

    s->min_framesize = get_bits_long(&s->gb, 24);
    s->max_framesize = get_bits_long(&s->gb, 24);
    
    s->samplerate = get_bits_long(&s->gb, 20);
    s->channels = get_bits(&s->gb, 3) + 1;
    s->bps = get_bits(&s->gb, 5) + 1;
    
    s->avctx->channels = s->channels;
    s->avctx->sample_rate = s->samplerate;

    skip_bits(&s->gb, 36); /* total num of samples */
    
    skip_bits(&s->gb, 64); /* md5 sum */
    skip_bits(&s->gb, 64); /* md5 sum */
    
    allocate_buffers(s);
}

static int decode_residuals(FLACContext *s, int channel, int pred_order)
{
    int i, tmp, partition, method_type, rice_order;
    int sample = 0, samples;

    method_type = get_bits(&s->gb, 2);
    if (method_type != 0){
        av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", method_type);
        return -1;
    }
    
    rice_order = get_bits(&s->gb, 4);

    samples= s->blocksize >> rice_order;

    sample= 
    i= pred_order;
    for (partition = 0; partition < (1 << rice_order); partition++)
    {
        tmp = get_bits(&s->gb, 4);
        if (tmp == 15)
        {
            av_log(s->avctx, AV_LOG_DEBUG, "fixed len partition\n");
            tmp = get_bits(&s->gb, 5);
            for (; i < samples; i++, sample++)
                s->decoded[channel][sample] = get_sbits(&s->gb, tmp);
        }
        else
        {
//            av_log(s->avctx, AV_LOG_DEBUG, "rice coded partition k=%d\n", tmp);
            for (; i < samples; i++, sample++){
                s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
            }
        }
        i= 0;
    }

//    av_log(s->avctx, AV_LOG_DEBUG, "partitions: %d, samples: %d\n", 1 << rice_order, sample);

    return 0;
}    

static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
{
    int i;
        
//    av_log(s->avctx, AV_LOG_DEBUG, "  SUBFRAME FIXED\n");
        
    /* warm up samples */
//    av_log(s->avctx, AV_LOG_DEBUG, "   warm up samples: %d\n", pred_order);
        
    for (i = 0; i < pred_order; i++)
    {
        s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
//        av_log(s->avctx, AV_LOG_DEBUG, "    %d: %d\n", i, s->decoded[channel][i]);
    }
    
    if (decode_residuals(s, channel, pred_order) < 0)
        return -1;

    switch(pred_order)
    {
        case 0:
            break;
        case 1:
            for (i = pred_order; i < s->blocksize; i++)
                s->decoded[channel][i] +=   s->decoded[channel][i-1];
            break;
        case 2:
            for (i = pred_order; i < s->blocksize; i++)
                s->decoded[channel][i] += 2*s->decoded[channel][i-1]
                                          - s->decoded[channel][i-2];
            break;
        case 3:
            for (i = pred_order; i < s->blocksize; i++)
                s->decoded[channel][i] += 3*s->decoded[channel][i-1] 
                                        - 3*s->decoded[channel][i-2]
                                        +   s->decoded[channel][i-3];
            break;
        case 4:
            for (i = pred_order; i < s->blocksize; i++)
                s->decoded[channel][i] += 4*s->decoded[channel][i-1] 
                                        - 6*s->decoded[channel][i-2]
                                        + 4*s->decoded[channel][i-3]
                                        -   s->decoded[channel][i-4];
            break;
        default:
            av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
            return -1;
    }

    return 0;
}

static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
{
    int sum, i, j;
    int coeff_prec, qlevel;
    int coeffs[pred_order];
        
//    av_log(s->avctx, AV_LOG_DEBUG, "  SUBFRAME LPC\n");
        
    /* warm up samples */
//    av_log(s->avctx, AV_LOG_DEBUG, "   warm up samples: %d\n", pred_order);
        
    for (i = 0; i < pred_order; i++)
    {
        s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
//        av_log(s->avctx, AV_LOG_DEBUG, "    %d: %d\n", i, s->decoded[channel][i]);
    }
    
    coeff_prec = get_bits(&s->gb, 4) + 1;
    if (coeff_prec == 16)
    {
        av_log(s->avctx, AV_LOG_DEBUG, "invalid coeff precision\n");
        return -1;
    }
//    av_log(s->avctx, AV_LOG_DEBUG, "   qlp coeff prec: %d\n", coeff_prec);
    qlevel = get_sbits(&s->gb, 5);
//    av_log(s->avctx, AV_LOG_DEBUG, "   quant level: %d\n", qlevel);
    if(qlevel < 0){
        av_log(s->avctx, AV_LOG_DEBUG, "qlevel %d not supported, maybe buggy stream\n", qlevel);
        return -1;
    }

    for (i = 0; i < pred_order; i++)
    {
        coeffs[i] = get_sbits(&s->gb, coeff_prec);
//        av_log(s->avctx, AV_LOG_DEBUG, "    %d: %d\n", i, coeffs[i]);
    }
    
    if (decode_residuals(s, channel, pred_order) < 0)
        return -1;

    for (i = pred_order; i < s->blocksize; i++)
    {
        sum = 0;
        for (j = 0; j < pred_order; j++)
            sum += coeffs[j] * s->decoded[channel][i-j-1];
        s->decoded[channel][i] += sum >> qlevel;
    }
    
    return 0;
}

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_DEBUG, "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_DEBUG, "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_DEBUG, "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_DEBUG, "invalid sample size code (%d)\n", sample_size_code);
        return -1;
    }

    if (get_bits1(&s->gb))
    {
        av_log(s->avctx, AV_LOG_DEBUG, "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= get_crc8(s->gb.buffer, get_bits_count(&s->gb)/8);
    if(crc8){
        av_log(s->avctx, AV_LOG_ERROR, "header crc missmatch 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 metadata_last, metadata_type, metadata_size;
    int tmp = 0, i, j = 0, input_buf_size = 0;
    int16_t *samples = data;

    if(s->max_framesize == 0){
        s->max_framesize= 8192; // 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= FFMIN(buf_size, s->max_framesize - s->bitstream_size);
            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);
    
    /* fLaC signature (be) */
    if (show_bits_long(&s->gb, 32) == bswap_32(ff_get_fourcc("fLaC")))
    {
        skip_bits(&s->gb, 32);

        av_log(s->avctx, AV_LOG_DEBUG, "STREAM HEADER\n");
        do {
            metadata_last = get_bits(&s->gb, 1);
            metadata_type = get_bits(&s->gb, 7);
            metadata_size = get_bits_long(&s->gb, 24);
            
            av_log(s->avctx, AV_LOG_DEBUG, " metadata block: flag = %d, type = %d, size = %d\n",
                metadata_last, metadata_type,
                metadata_size);
            if(metadata_size){
                switch(metadata_type)
                {
                case METADATA_TYPE_STREAMINFO:
                    metadata_streaminfo(s);
                    dump_headers(s);
                    break;
                default:
                    for(i=0; i<metadata_size; i++)
                        skip_bits(&s->gb, 8);
                }
            }
        } while(!metadata_last);
    }
    else
    {
        
        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
    switch(s->decorrelation)
    {
        case INDEPENDENT:
            for (j = 0; j < s->blocksize; j++)
            {
                for (i = 0; i < s->channels; i++)
                    *(samples++) = s->decoded[i][j];
            }
            break;
        case LEFT_SIDE:
            assert(s->channels == 2);
            for (i = 0; i < s->blocksize; i++)
            {
                *(samples++) = s->decoded[0][i];
                *(samples++) = s->decoded[0][i] - s->decoded[1][i];
            }
            break;
        case RIGHT_SIDE:
            assert(s->channels == 2);
            for (i = 0; i < s->blocksize; i++)
            {
                *(samples++) = s->decoded[0][i] + s->decoded[1][i];
                *(samples++) = s->decoded[1][i];
            }
            break;
        case MID_SIDE:
            assert(s->channels == 2);
            for (i = 0; i < s->blocksize; i++)
            {
                int mid, side;
                mid = s->decoded[0][i];
                side = s->decoded[1][i];

#if 1 //needs to be checked but IMHO it should be binary identical
                mid -= side>>1;
                *(samples++) = mid + side;
                *(samples++) = mid;
#else
                
                mid <<= 1;
                if (side & 1)
                    mid++;
                *(samples++) = (mid + side) >> 1;
                *(samples++) = (mid - side) >> 1;
#endif
            }
            break;
    }
#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),
    flac_decode_init,
    NULL,
    flac_decode_close,
    flac_decode_frame,
    .flush= flac_flush,    
};