/*
 * 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 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_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);
                    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 != 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;
    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, 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 i, j;
    int coeff_prec, qlevel;
    #if __STDC_VERSION__ >= 199901L
    int coeffs[pred_order];
    #else
    int *coeffs=(int*)_alloca(sizeof(int)*pred_order);
    #endif

//    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;

    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] * s->decoded[channel][i-j-1];
            s->decoded[channel][i] += sum >> qlevel;
        }
    } else {
        int sum;
        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;