/*
  * 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"
 
 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, AVFormatParameters *ap) {
     PMPContext *pmp = s->priv_data;
     AVIOContext *pb = s->pb;
     int tb_num, tb_den;
     int index_cnt;
     int audio_codec_id = CODEC_ID_NONE;
     int srate, channels;
     int i;
     uint64_t pos;
     AVStream *vst = av_new_stream(s, 0);
     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 = CODEC_ID_MPEG4;
         break;
     case 1:
         vst->codec->codec_id = 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);
     av_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 = CODEC_ID_MP3;
         break;
     case 1:
         av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n");
         audio_codec_id = 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;
     for (i = 1; i < pmp->num_streams; i++) {
         AVStream *ast = av_new_stream(s, i);
         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;
         av_set_pts_info(ast, 32, 1, srate);
     }
     pos = avio_tell(pb) + 4*index_cnt;
     for (i = 0; i < index_cnt; i++) {
         int size = avio_rl32(pb);
         int flags = size & 1 ? AVINDEX_KEYFRAME : 0;
         size >>= 1;
         av_add_index_entry(vst, pos, i, size, 0, flags);
         pos += size;
     }
     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);
         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));
         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;
         // FIXME: this is a hack that should be remove once
         // compute_pkt_fields can handle
         if (pmp->cur_stream == 0)
             pkt->dts = s->streams[0]->cur_dts++;
         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;
     // fallback to 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 format"),
     .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,
 };