annotate flac.c @ 6571:ad0c77d2b6ee libavcodec

Increase alignment for DCT block arrays from 8 to 16 Patch by Alexander Strange ( astrange ithinksw com )

/*
 * FLAC (Free Lossless Audio Codec) decoder
 * Copyright (c) 2003 Alex Beregszaszi
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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>

#define ALT_BITSTREAM_READER
#include "avcodec.h"
#include "bitstream.h"
#include "golomb.h"
#include "crc.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;
    unsigned 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 int64_t get_utf8(GetBitContext *gb){
    int64_t val;
    GET_UTF8(val, get_bits(gb, 8), return -1;)
    return val;
}

static void metadata_streaminfo(FLACContext *s);
static void allocate_buffers(FLACContext *s);
static int metadata_parse(FLACContext *s);

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

    if (avctx->extradata_size > 4) {
        /* initialize based on the demuxer-supplied streamdata header */
        init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
        if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
            metadata_streaminfo(s);
            allocate_buffers(s);
        } else {
            metadata_parse(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 */

    dump_headers(s);
}

/**
 * Parse a list of metadata blocks. This list of blocks must begin with
 * the fLaC marker.
 * @param s the flac decoding context containing the gb bit reader used to
 *          parse metadata
 * @return 1 if some metadata was read, 0 if no fLaC marker was found
 */
static int metadata_parse(FLACContext *s)
{
    int i, metadata_last, metadata_type, metadata_size, streaminfo_updated=0;

    if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) {
        skip_bits(&s->gb, 32);

        av_log(s->avctx, AV_LOG_DEBUG, "STREAM HEADER\n");
        do {
            metadata_last = get_bits1(&s->gb);
            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);
                    streaminfo_updated = 1;
                    break;

                default:
                    for (i=0; i<metadata_size; i++)
                        skip_bits(&s->gb, 8);
                }
            }
        } while (!metadata_last);

        if (streaminfo_updated)
            allocate_buffers(s);
        return 1;
    }
    return 0;
}

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 > 1){
        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;
    if (pred_order > samples) {
        av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", pred_order, samples);
        return -1;
    }

    sample=
    i= pred_order;
    for (partition = 0; partition < (1 << rice_order); partition++)
    {
        tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5);
        if (tmp == (method_type == 0 ? 15 : 31))
        {
            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)
{
    const int blocksize = s->blocksize;
    int32_t *decoded = s->decoded[channel];
    int a, b, c, d, 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++)
    {
        decoded[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;

    a = decoded[pred_order-1];
    b = a - decoded[pred_order-2];
    c = b - decoded[pred_order-2] + decoded[pred_order-3];
    d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4];

    switch(pred_order)
    {
        case 0:
            break;
        case 1:
            for (i = pred_order; i < blocksize; i++)
                decoded[i] = a += decoded[i];
            break;
        case 2:
            for (i = pred_order; i < blocksize; i++)
                decoded[i] = a += b += decoded[i];
            break;
        case 3:
            for (i = pred_order; i < blocksize; i++)
                decoded[i] = a += b += c += decoded[i];
            break;
        case 4:
            for (i = pred_order; i < blocksize; i++)
                decoded[i] = a += b += c += d += decoded[i];
            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 i, j;
    int coeff_prec, qlevel;
    int coeffs[pred_order];
    int32_t *decoded = s->decoded[channel];

//    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++)
    {
        decoded[i] = get_sbits(&s->gb, s->curr_bps);
//        av_log(s->avctx, AV_LOG_DEBUG, "    %d: %d\n", i, decoded[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;

    if (s->bps > 16) {
        int64_t sum;
        for (i = pred_order; i < s->blocksize; i++)
        {
            sum = 0;
            for (j = 0; j < pred_order; j++)
                sum += (int64_t)coeffs[j] * decoded[i-j-1];
            decoded[i] += sum >> qlevel;
        }
    } else {
        for (i = pred_order; i < s->blocksize-1; i += 2)
        {
            int c;
            int d = decoded[i-pred_order];
            int s0 = 0, s1 = 0;
            for (j = pred_order-1; j > 0; j--)
            {
                c = coeffs[j];
                s0 += c*d;
                d = decoded[i-j];
                s1 += c*d;
            }
            c = coeffs[0];
            s0 += c*d;
            d = decoded[i] += s0 >> qlevel;
            s1 += c*d;
            decoded[i+1] += s1 >> qlevel;
        }
        if (i < s->blocksize)
        {
            int sum = 0;
            for (j = 0; j < pred_order; j++)
                sum += coeffs[j] * decoded[i-j-1];
            decoded[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_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 alloc_data_size)
{
    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(blocksize * s->channels * sizeof(int16_t) > alloc_data_size)
        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_crc_get_table(AV_CRC_8_ATM), 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,
                            const 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;
    int alloc_data_size= *data_size;

    *data_size=0;

    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 & 0xFFFE) != 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) & 0xFFFE) != 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, alloc_data_size) < 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 av_cold 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,
};