/* * 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 "demux.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(const 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->codecpar->codec_type = AVMEDIA_TYPE_VIDEO; avio_skip(pb, 8); switch (avio_rl32(pb)) { case 0: vst->codecpar->codec_id = AV_CODEC_ID_MPEG4; break; case 1: vst->codecpar->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->codecpar->width = avio_rl32(pb); vst->codecpar->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 (avio_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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; ast->codecpar->codec_id = audio_codec_id; ast->codecpar->ch_layout.nb_channels = channels; ast->codecpar->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 (avio_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; } const FFInputFormat ff_pmp_demuxer = { .p.name = "pmp", .p.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, };