/*
  * PMP demuxer.
  * Copyright (c) 2011 Reimar Döffinger
  *
  * 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/intreadwrite.h"
 #include "avformat.h"

 #include "internal.h"

 
 typedef struct {
     int cur_stream;
     int num_streams;
     int audio_packets;
     int current_packet;
     uint32_t *packet_sizes;
     int packet_sizes_alloc;
 } PMPContext;
 
 static int pmp_probe(AVProbeData *p) {
     if (AV_RN32(p->buf) == AV_RN32("pmpm") &&
         AV_RL32(p->buf + 4) == 1)
         return AVPROBE_SCORE_MAX;
     return 0;
 }
 

 static int pmp_header(AVFormatContext *s)

 {

     PMPContext *pmp = s->priv_data;
     AVIOContext *pb = s->pb;
     int tb_num, tb_den;

     uint32_t index_cnt;

     int audio_codec_id = AV_CODEC_ID_NONE;

     int srate, channels;

     unsigned i;

     uint64_t pos;

     int64_t fsize = avio_size(pb);
 

     AVStream *vst = avformat_new_stream(s, NULL);

     if (!vst)
         return AVERROR(ENOMEM);
     vst->codec->codec_type = AVMEDIA_TYPE_VIDEO;
     avio_skip(pb, 8);
     switch (avio_rl32(pb)) {
     case 0:

         vst->codec->codec_id = AV_CODEC_ID_MPEG4;

         break;
     case 1:

         vst->codec->codec_id = AV_CODEC_ID_H264;

         break;
     default:
         av_log(s, AV_LOG_ERROR, "Unsupported video format\n");
         break;
     }
     index_cnt = avio_rl32(pb);
     vst->codec->width  = avio_rl32(pb);
     vst->codec->height = avio_rl32(pb);
 
     tb_num = avio_rl32(pb);
     tb_den = avio_rl32(pb);

     avpriv_set_pts_info(vst, 32, tb_num, tb_den);

     vst->nb_frames = index_cnt;
     vst->duration = index_cnt;
 
     switch (avio_rl32(pb)) {
     case 0:

         audio_codec_id = AV_CODEC_ID_MP3;

         break;
     case 1:
         av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n");

         audio_codec_id = AV_CODEC_ID_AAC;

         break;
     default:
         av_log(s, AV_LOG_ERROR, "Unsupported audio format\n");
         break;
     }
     pmp->num_streams = avio_rl16(pb) + 1;
     avio_skip(pb, 10);
     srate = avio_rl32(pb);
     channels = avio_rl32(pb) + 1;

     pos = avio_tell(pb) + 4LL*index_cnt;

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

         uint32_t size = avio_rl32(pb);

         int flags = size & 1 ? AVINDEX_KEYFRAME : 0;

         if (url_feof(pb)) {
             av_log(s, AV_LOG_FATAL, "Encountered EOF while reading index.\n");
             return AVERROR_INVALIDDATA;
         }

         size >>= 1;

         if (size < 9 + 4*pmp->num_streams) {
             av_log(s, AV_LOG_ERROR, "Packet too small\n");
             return AVERROR_INVALIDDATA;
         }

         av_add_index_entry(vst, pos, i, size, 0, flags);
         pos += size;

         if (fsize > 0 && i == 0 && pos > fsize) {
             av_log(s, AV_LOG_ERROR, "File ends before first packet\n");
             return AVERROR_INVALIDDATA;
         }

     }

     for (i = 1; i < pmp->num_streams; i++) {
         AVStream *ast = avformat_new_stream(s, NULL);
         if (!ast)
             return AVERROR(ENOMEM);
         ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
         ast->codec->codec_id = audio_codec_id;
         ast->codec->channels = channels;
         ast->codec->sample_rate = srate;
         avpriv_set_pts_info(ast, 32, 1, srate);
     }

     return 0;
 }
 

 static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
 {

     PMPContext *pmp = s->priv_data;
     AVIOContext *pb = s->pb;
     int ret = 0;
     int i;
 
     if (url_feof(pb))
         return AVERROR_EOF;
     if (pmp->cur_stream == 0) {
         int num_packets;
         pmp->audio_packets = avio_r8(pb);

         if (!pmp->audio_packets) {

             av_log(s, AV_LOG_ERROR, "No audio packets.\n");
             return AVERROR_INVALIDDATA;

         }

         num_packets = (pmp->num_streams - 1) * pmp->audio_packets + 1;
         avio_skip(pb, 8);
         pmp->current_packet = 0;
         av_fast_malloc(&pmp->packet_sizes,
                        &pmp->packet_sizes_alloc,
                        num_packets * sizeof(*pmp->packet_sizes));

         if (!pmp->packet_sizes_alloc) {
             av_log(s, AV_LOG_ERROR, "Cannot (re)allocate packet buffer\n");
             return AVERROR(ENOMEM);
         }

         for (i = 0; i < num_packets; i++)
             pmp->packet_sizes[i] = avio_rl32(pb);
     }
     ret = av_get_packet(pb, pkt, pmp->packet_sizes[pmp->current_packet]);
     if (ret >= 0) {
         ret = 0;
         pkt->stream_index = pmp->cur_stream;
     }
     if (pmp->current_packet % pmp->audio_packets == 0)
         pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams;
     pmp->current_packet++;
     return ret;
 }
 

 static int pmp_seek(AVFormatContext *s, int stream_index, int64_t ts, int flags)

 {

     PMPContext *pmp = s->priv_data;
     pmp->cur_stream = 0;

     // fall back on default seek now

     return -1;
 }
 
 static int pmp_close(AVFormatContext *s)
 {
     PMPContext *pmp = s->priv_data;
     av_freep(&pmp->packet_sizes);
     return 0;
 }
 
 AVInputFormat ff_pmp_demuxer = {
     .name           = "pmp",

     .long_name      = NULL_IF_CONFIG_SMALL("Playstation Portable PMP"),

     .priv_data_size = sizeof(PMPContext),
     .read_probe     = pmp_probe,
     .read_header    = pmp_header,
     .read_packet    = pmp_packet,
     .read_seek      = pmp_seek,
     .read_close     = pmp_close,
 };