mirror of
https://git.ffmpeg.org/ffmpeg.git
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1b5d065ca7
The code cannot handle there being none, but that should not happen for valid files. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC:libav-stable@libav.org
186 lines
5.5 KiB
C
186 lines
5.5 KiB
C
/*
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* PMP demuxer
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* Copyright (c) 2011 Reimar Döffinger
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "internal.h"
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typedef struct PMPContext {
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int cur_stream;
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int num_streams;
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int audio_packets;
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int current_packet;
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uint32_t *packet_sizes;
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int packet_sizes_alloc;
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} PMPContext;
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static int pmp_probe(AVProbeData *p)
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{
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if (!memcmp(p->buf, "pmpm\1\0\0\0", 8))
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return AVPROBE_SCORE_MAX;
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return 0;
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}
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static int pmp_header(AVFormatContext *s)
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{
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PMPContext *pmp = s->priv_data;
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AVIOContext *pb = s->pb;
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int tb_num, tb_den;
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int index_cnt;
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int audio_codec_id = AV_CODEC_ID_NONE;
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int srate, channels;
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int i;
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uint64_t pos;
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AVStream *vst = avformat_new_stream(s, NULL);
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if (!vst)
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return AVERROR(ENOMEM);
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vst->codec->codec_type = AVMEDIA_TYPE_VIDEO;
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avio_skip(pb, 8);
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switch (avio_rl32(pb)) {
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case 0:
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vst->codec->codec_id = AV_CODEC_ID_MPEG4;
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break;
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case 1:
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vst->codec->codec_id = AV_CODEC_ID_H264;
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break;
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default:
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av_log(s, AV_LOG_ERROR, "Unsupported video format\n");
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break;
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}
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index_cnt = avio_rl32(pb);
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vst->codec->width = avio_rl32(pb);
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vst->codec->height = avio_rl32(pb);
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tb_num = avio_rl32(pb);
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tb_den = avio_rl32(pb);
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avpriv_set_pts_info(vst, 32, tb_num, tb_den);
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vst->nb_frames = index_cnt;
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vst->duration = index_cnt;
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switch (avio_rl32(pb)) {
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case 0:
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audio_codec_id = AV_CODEC_ID_MP3;
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break;
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case 1:
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av_log(s, AV_LOG_WARNING, "AAC is not yet correctly supported\n");
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audio_codec_id = AV_CODEC_ID_AAC;
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break;
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default:
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av_log(s, AV_LOG_ERROR, "Unsupported audio format\n");
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break;
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}
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pmp->num_streams = avio_rl16(pb) + 1;
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avio_skip(pb, 10);
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srate = avio_rl32(pb);
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channels = avio_rl32(pb) + 1;
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for (i = 1; i < pmp->num_streams; i++) {
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AVStream *ast = avformat_new_stream(s, NULL);
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if (!ast)
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return AVERROR(ENOMEM);
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ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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ast->codec->codec_id = audio_codec_id;
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ast->codec->channels = channels;
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ast->codec->sample_rate = srate;
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avpriv_set_pts_info(ast, 32, 1, srate);
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}
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pos = avio_tell(pb) + 4 * index_cnt;
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for (i = 0; i < index_cnt; i++) {
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int size = avio_rl32(pb);
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int flags = size & 1 ? AVINDEX_KEYFRAME : 0;
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size >>= 1;
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av_add_index_entry(vst, pos, i, size, 0, flags);
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pos += size;
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}
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return 0;
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}
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static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
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{
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PMPContext *pmp = s->priv_data;
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AVIOContext *pb = s->pb;
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int ret = 0;
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int i;
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if (pb->eof_reached)
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return AVERROR_EOF;
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if (pmp->cur_stream == 0) {
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int num_packets;
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pmp->audio_packets = avio_r8(pb);
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if (!pmp->audio_packets) {
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av_log(s, AV_LOG_ERROR, "No audio packets.\n");
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return AVERROR_INVALIDDATA;
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}
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num_packets = (pmp->num_streams - 1) * pmp->audio_packets + 1;
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avio_skip(pb, 8);
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pmp->current_packet = 0;
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av_fast_malloc(&pmp->packet_sizes,
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&pmp->packet_sizes_alloc,
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num_packets * sizeof(*pmp->packet_sizes));
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if (!pmp->packet_sizes_alloc) {
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av_log(s, AV_LOG_ERROR, "Cannot (re)allocate packet buffer\n");
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return AVERROR(ENOMEM);
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}
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for (i = 0; i < num_packets; i++)
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pmp->packet_sizes[i] = avio_rl32(pb);
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}
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ret = av_get_packet(pb, pkt, pmp->packet_sizes[pmp->current_packet]);
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if (ret > 0) {
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ret = 0;
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// FIXME: this is a hack that should be removed once
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// compute_pkt_fields() can handle timestamps properly
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if (pmp->cur_stream == 0)
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pkt->dts = s->streams[0]->cur_dts++;
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pkt->stream_index = pmp->cur_stream;
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}
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pmp->current_packet++;
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if (pmp->current_packet == 1 || pmp->current_packet > pmp->audio_packets)
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pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams;
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return ret;
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}
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static int pmp_seek(AVFormatContext *s, int stream_idx, int64_t ts, int flags)
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{
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PMPContext *pmp = s->priv_data;
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pmp->cur_stream = 0;
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// fall back on default seek now
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return -1;
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}
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static int pmp_close(AVFormatContext *s)
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{
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PMPContext *pmp = s->priv_data;
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av_freep(&pmp->packet_sizes);
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return 0;
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}
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AVInputFormat ff_pmp_demuxer = {
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.name = "pmp",
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.long_name = NULL_IF_CONFIG_SMALL("Playstation Portable PMP"),
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.priv_data_size = sizeof(PMPContext),
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.read_probe = pmp_probe,
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.read_header = pmp_header,
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.read_packet = pmp_packet,
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.read_seek = pmp_seek,
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.read_close = pmp_close,
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};
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