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/*

 * MP3 demuxer

 * Copyright (c) 2003 Fabrice Bellard

 *

 * 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

 */

#include "libavutil/opt.h"

#include "libavutil/avstring.h"

#include "libavutil/intreadwrite.h"

#include "libavutil/dict.h"

#include "libavutil/mathematics.h"

#include "avformat.h"

#include "internal.h"

#include "id3v2.h"

#include "id3v1.h"

#include "libavcodec/mpegaudiodecheader.h"

#define XING_FLAG_FRAMES 0x01

#define XING_FLAG_SIZE   0x02

#define XING_FLAG_TOC    0x04

#define XING_TOC_COUNT 100

typedef struct {

    AVClass *class;

    int64_t filesize;

    int64_t header_filesize;

    int xing_toc;

    int start_pad;

    int end_pad;

    int usetoc;

    int is_cbr;

} MP3DecContext;

/* mp3 read */

static int mp3_read_probe(AVProbeData *p)

{

    int max_frames, first_frames = 0;

    int fsize, frames, sample_rate;

    uint32_t header;

    const uint8_t *buf, *buf0, *buf2, *end;

    AVCodecContext avctx;

    buf0 = p->buf;

    end = p->buf + p->buf_size - sizeof(uint32_t);

    while(buf0 < end && !*buf0)

        buf0++;

    max_frames = 0;

    buf = buf0;

    for(; buf < end; buf= buf2+1) {

        buf2 = buf;

        for(frames = 0; buf2 < end; frames++) {

            header = AV_RB32(buf2);

            fsize = avpriv_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate);

            if(fsize < 0)

                break;

            buf2 += fsize;

        }

        max_frames = FFMAX(max_frames, frames);

        if(buf == buf0)

            first_frames= frames;

    }

    // keep this in sync with ac3 probe, both need to avoid

    // issues with MPEG-files!

    if   (first_frames>=4) return AVPROBE_SCORE_EXTENSION + 1;

    else if(max_frames>200)return AVPROBE_SCORE_EXTENSION;

    else if(max_frames>=4) return AVPROBE_SCORE_EXTENSION / 2;

    else if(ff_id3v2_match(buf0, ID3v2_DEFAULT_MAGIC) && 2*ff_id3v2_tag_len(buf0) >= p->buf_size)

                           return p->buf_size < PROBE_BUF_MAX ? AVPROBE_SCORE_EXTENSION / 4 : AVPROBE_SCORE_EXTENSION - 2;

    else if(max_frames>=1) return 1;

    else                   return 0;

//mpegps_mp3_unrecognized_format.mpg has max_frames=3

}

static void read_xing_toc(AVFormatContext *s, int64_t filesize, int64_t duration)

{

    int i;

    MP3DecContext *mp3 = s->priv_data;

    int fill_index = mp3->usetoc && duration > 0;

    if (!filesize &&

        !(filesize = avio_size(s->pb))) {

        av_log(s, AV_LOG_WARNING, "Cannot determine file size, skipping TOC table.\n");

        fill_index = 0;

    }

    for (i = 0; i < XING_TOC_COUNT; i++) {

        uint8_t b = avio_r8(s->pb);

        if (fill_index)

            av_add_index_entry(s->streams[0],

                           av_rescale(b, filesize, 256),

                           av_rescale(i, duration, XING_TOC_COUNT),

                           0, 0, AVINDEX_KEYFRAME);

    }

    if (fill_index)

        mp3->xing_toc = 1;

}

/**

 * Try to find Xing/Info/VBRI tags and compute duration from info therein

 */

static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)

{

    MP3DecContext *mp3 = s->priv_data;

    uint32_t v, spf;

    unsigned frames = 0; /* Total number of frames in file */

    unsigned size = 0; /* Total number of bytes in the stream */

    static const int64_t xing_offtbl[2][2] = {{32, 17}, {17,9}};

    MPADecodeHeader c;

    int vbrtag_size = 0;

    int is_cbr;

    v = avio_rb32(s->pb);

    if(ff_mpa_check_header(v) < 0)

      return -1;

    if (avpriv_mpegaudio_decode_header(&c, v) == 0)

        vbrtag_size = c.frame_size;

    if(c.layer != 3)

        return -1;

    spf = c.lsf ? 576 : 1152; /* Samples per frame, layer 3 */

    /* Check for Xing / Info tag */

    avio_skip(s->pb, xing_offtbl[c.lsf == 1][c.nb_channels == 1]);

    v = avio_rb32(s->pb);

    is_cbr = v == MKBETAG('I', 'n', 'f', 'o');

    if (v == MKBETAG('X', 'i', 'n', 'g') || is_cbr) {

        v = avio_rb32(s->pb);

        if(v & XING_FLAG_FRAMES)

            frames = avio_rb32(s->pb);

        if(v & XING_FLAG_SIZE)

            size = avio_rb32(s->pb);

        if (v & XING_FLAG_TOC)

            read_xing_toc(s, size, av_rescale_q(frames, (AVRational){spf, c.sample_rate},

                                    st->time_base));

        if(v & 8)

            avio_skip(s->pb, 4);

        v = avio_rb32(s->pb);

        if(v == MKBETAG('L', 'A', 'M', 'E') || v == MKBETAG('L', 'a', 'v', 'f')) {

            avio_skip(s->pb, 21-4);

            v= avio_rb24(s->pb);

            mp3->start_pad = v>>12;

            mp3->  end_pad = v&4095;

            st->skip_samples = mp3->start_pad + 528 + 1;

            av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3->  end_pad);

        }

    }

    /* Check for VBRI tag (always 32 bytes after end of mpegaudio header) */

    avio_seek(s->pb, base + 4 + 32, SEEK_SET);

    v = avio_rb32(s->pb);

    if(v == MKBETAG('V', 'B', 'R', 'I')) {

        /* Check tag version */

        if(avio_rb16(s->pb) == 1) {

            /* skip delay and quality */

            avio_skip(s->pb, 4);

            size = avio_rb32(s->pb);

            frames = avio_rb32(s->pb);

        }

    }

    if(!frames && !size)

        return -1;

    /* Skip the vbr tag frame */

    avio_seek(s->pb, base + vbrtag_size, SEEK_SET);

    if(frames)

        st->duration = av_rescale_q(frames, (AVRational){spf, c.sample_rate},

                                    st->time_base);

    if (size && frames && !is_cbr)

        st->codec->bit_rate = av_rescale(size, 8 * c.sample_rate, frames * (int64_t)spf);

    mp3->is_cbr          = is_cbr;

    mp3->header_filesize = size;

    return 0;

}

static int mp3_read_header(AVFormatContext *s)

{

    MP3DecContext *mp3 = s->priv_data;

    AVStream *st;

    int64_t off;

    st = avformat_new_stream(s, NULL);

    if (!st)

        return AVERROR(ENOMEM);

    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;

    st->codec->codec_id = AV_CODEC_ID_MP3;

    st->need_parsing = AVSTREAM_PARSE_FULL_RAW;

    st->start_time = 0;

    // lcm of all mp3 sample rates

    avpriv_set_pts_info(st, 64, 1, 14112000);

    s->pb->maxsize = -1;

    off = avio_tell(s->pb);

    if (!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX))

        ff_id3v1_read(s);

    if(s->pb->seekable)

        mp3->filesize = avio_size(s->pb);

    if (mp3_parse_vbr_tags(s, st, off) < 0)

        avio_seek(s->pb, off, SEEK_SET);

    /* the parameters will be extracted from the compressed bitstream */

    return 0;

}

#define MP3_PACKET_SIZE 1024

static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)

{

    MP3DecContext *mp3 = s->priv_data;

    int ret, size;

    int64_t pos;

    size= MP3_PACKET_SIZE;

    pos = avio_tell(s->pb);

    if(mp3->filesize > ID3v1_TAG_SIZE && pos < mp3->filesize)

        size= FFMIN(size, mp3->filesize - pos);

    ret= av_get_packet(s->pb, pkt, size);

    if (ret <= 0) {

        if(ret<0)

            return ret;

        return AVERROR_EOF;

    }

    pkt->flags &= ~AV_PKT_FLAG_CORRUPT;

    pkt->stream_index = 0;

    if (ret >= ID3v1_TAG_SIZE &&

        memcmp(&pkt->data[ret - ID3v1_TAG_SIZE], "TAG", 3) == 0)

        ret -= ID3v1_TAG_SIZE;

    /* note: we need to modify the packet size here to handle the last

       packet */

    pkt->size = ret;

    return ret;

}

static int check(AVFormatContext *s, int64_t pos)

{

    int64_t ret = avio_seek(s->pb, pos, SEEK_SET);

    unsigned header;

    MPADecodeHeader sd;

    if (ret < 0)

        return ret;

    header = avio_rb32(s->pb);

    if (ff_mpa_check_header(header) < 0)

        return -1;

    if (avpriv_mpegaudio_decode_header(&sd, header) == 1)

        return -1;

    return sd.frame_size;

}

static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp,

                    int flags)

{

    MP3DecContext *mp3 = s->priv_data;

    AVIndexEntry *ie, ie1;

    AVStream *st = s->streams[0];

    int64_t ret  = av_index_search_timestamp(st, timestamp, flags);

    int i, j;

    int dir = (flags&AVSEEK_FLAG_BACKWARD) ? -1 : 1;

    if (mp3->is_cbr && st->duration > 0 && mp3->header_filesize > s->data_offset) {

        int64_t filesize = avio_size(s->pb);

        int64_t duration;

        if (filesize <= s->data_offset)

            filesize = mp3->header_filesize;

        filesize -= s->data_offset;

        duration = av_rescale(st->duration, filesize, mp3->header_filesize - s->data_offset);

        ie = &ie1;

        timestamp = av_clip64(timestamp, 0, duration);

        ie->timestamp = timestamp;

        ie->pos       = av_rescale(timestamp, filesize, duration) + s->data_offset;

    } else if (mp3->xing_toc) {

        if (ret < 0)

            return ret;

        ie = &st->index_entries[ret];

    } else {

        st->skip_samples = timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;

        return -1;

    }

    ret = avio_seek(s->pb, ie->pos, SEEK_SET);

    if (ret < 0)

        return ret;

#define MIN_VALID 3

    for(i=0; i<4096; i++) {

        int64_t pos = ie->pos + i*dir;

        for(j=0; j

            ret = check(s, pos);

            if(ret < 0)

                break;

            pos += ret;

        }

        if(j==MIN_VALID)

            break;

    }

    if(j!=MIN_VALID)

        i=0;

    ret = avio_seek(s->pb, ie->pos + i*dir, SEEK_SET);

    if (ret < 0)

        return ret;

    ff_update_cur_dts(s, st, ie->timestamp);

    st->skip_samples = ie->timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;

    return 0;

}

static const AVOption options[] = {

    { "usetoc", "use table of contents", offsetof(MP3DecContext, usetoc), AV_OPT_TYPE_INT, {.i64 = -1}, -1, 1, AV_OPT_FLAG_DECODING_PARAM},

    { NULL },

};

static const AVClass demuxer_class = {

    .class_name = "mp3",

    .item_name  = av_default_item_name,

    .option     = options,

    .version    = LIBAVUTIL_VERSION_INT,

    .category   = AV_CLASS_CATEGORY_DEMUXER,

};

AVInputFormat ff_mp3_demuxer = {

    .name           = "mp3",

    .long_name      = NULL_IF_CONFIG_SMALL("MP2/3 (MPEG audio layer 2/3)"),

    .read_probe     = mp3_read_probe,

    .read_header    = mp3_read_header,

    .read_packet    = mp3_read_packet,

    .read_seek      = mp3_seek,

    .priv_data_size = sizeof(MP3DecContext),

    .flags          = AVFMT_GENERIC_INDEX,

    .extensions     = "mp2,mp3,m2a,mpa", /* XXX: use probe */

    .priv_class     = &demuxer_class,

};