rename flac.c to flacdec.c

Originally committed as revision 16735 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 795 insertions, 0 deletions

diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
new file mode 100644
index 0000000000..16183f3b00
--- /dev/null
+++ b/libavcodec/flacdec.c
@@ -0,0 +1,795 @@
+/*
+ * 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 flacdec.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 "libavutil/crc.h"
+#include "avcodec.h"
+#include "bitstream.h"
+#include "golomb.h"
+#include "flac.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 {
+    FLACSTREAMINFO
+
+    AVCodecContext *avctx;
+    GetBitContext gb;
+
+    int blocksize/*, last_blocksize*/;
+    int curr_bps;
+    enum decorrelation_type decorrelation;
+
+    int32_t *decoded[MAX_CHANNELS];
+    uint8_t *bitstream;
+    unsigned int bitstream_size;
+    unsigned int bitstream_index;
+    unsigned int allocated_bitstream_size;
+} FLACContext;
+
+#define METADATA_TYPE_STREAMINFO 0
+
+static const int sample_rate_table[] =
+{ 0,
+  88200, 176400, 192000,
+  8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
+  0, 0, 0, 0 };
+
+static const int sample_size_table[] =
+{ 0, 8, 12, 0, 16, 20, 24, 0 };
+
+static const 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 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 */
+        if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
+            ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, avctx->extradata);
+            allocate_buffers(s);
+        } else {
+            init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
+            metadata_parse(s);
+        }
+    }
+
+    avctx->sample_fmt = SAMPLE_FMT_S16;
+    return 0;
+}
+
+static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
+{
+    av_log(avctx, AV_LOG_DEBUG, "  Blocksize: %d .. %d\n", s->min_blocksize, s->max_blocksize);
+    av_log(avctx, AV_LOG_DEBUG, "  Max Framesize: %d\n", s->max_framesize);
+    av_log(avctx, AV_LOG_DEBUG, "  Samplerate: %d\n", s->samplerate);
+    av_log(avctx, AV_LOG_DEBUG, "  Channels: %d\n", s->channels);
+    av_log(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);
+    }
+
+    if(s->allocated_bitstream_size < s->max_framesize)
+        s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
+}
+
+void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
+                              const uint8_t *buffer)
+{
+    GetBitContext gb;
+    init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
+
+    /* mandatory streaminfo */
+    s->min_blocksize = get_bits(&gb, 16);
+    s->max_blocksize = get_bits(&gb, 16);
+
+    skip_bits(&gb, 24); /* skip min frame size */
+    s->max_framesize = get_bits_long(&gb, 24);
+
+    s->samplerate = get_bits_long(&gb, 20);
+    s->channels = get_bits(&gb, 3) + 1;
+    s->bps = get_bits(&gb, 5) + 1;
+
+    avctx->channels = s->channels;
+    avctx->sample_rate = s->samplerate;
+
+    skip_bits(&gb, 36); /* total num of samples */
+
+    skip_bits(&gb, 64); /* md5 sum */
+    skip_bits(&gb, 64); /* md5 sum */
+
+    dump_headers(avctx, 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;
+    int initial_pos= get_bits_count(&s->gb);
+
+    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);
+
+            if(get_bits_count(&s->gb) + 8*metadata_size > s->gb.size_in_bits){
+                skip_bits_long(&s->gb, initial_pos - get_bits_count(&s->gb));
+                break;
+            }
+
+            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:
+                    ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, s->gb.buffer+get_bits_count(&s->gb)/8);
+                    streaminfo_updated = 1;
+
+                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;
+
+    if(pred_order > 0)
+        a = decoded[pred_order-1];
+    if(pred_order > 1)
+        b = a - decoded[pred_order-2];
+    if(pred_order > 2)
+        c = b - decoded[pred_order-2] + decoded[pred_order-3];
+    if(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 < 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->avctx, (FLACStreaminfo *)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= FFMAX(4, buf_size); // 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
+            if(s->bitstream_size < 4 || AV_RL32(s->bitstream) != MKTAG('f','L','a','C'))
+                buf_size= FFMIN(buf_size, s->max_framesize - FFMIN(s->bitstream_size, s->max_framesize));
+            input_buf_size= buf_size;
+
+            if(s->bitstream_size + buf_size < buf_size || s->bitstream_index + s->bitstream_size + buf_size < s->bitstream_index)
+                return -1;
+
+            if(s->allocated_bitstream_size < s->bitstream_size + buf_size)
+                s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->bitstream_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 && input_buf_size){
+//                printf("wanna more data ...\n");
+                return input_buf_size;
+            }
+    }
+
+    init_get_bits(&s->gb, buf, buf_size*8);
+
+    if(metadata_parse(s))
+        goto end;
+
+        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,
+    CODEC_CAP_DELAY,
+    .flush= flac_flush,
+    .long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
+};