--- /dev/null
+/*
+ * PMP demuxer
+ * Copyright (c) 2011 Reimar Döffinger
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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 PMPContext {
+ 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 (!memcmp(p->buf, "pmpm\1\0\0\0", 8))
+ 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 = 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 = 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_WARNING, "AAC is 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 = 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;
+ 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 (pb->eof_reached)
+ 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));
+ 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;
+ // FIXME: this is a hack that should be removed once
+ // compute_pkt_fields() can handle timestamps properly
+ if (pmp->cur_stream == 0)
+ pkt->dts = s->streams[0]->cur_dts++;
+ pkt->stream_index = pmp->cur_stream;
+ }
+ pmp->current_packet++;
+ if (pmp->current_packet == 1 || pmp->current_packet > pmp->audio_packets)
+ pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams;
+
+ return ret;
+}
+
+static int pmp_seek(AVFormatContext *s, int stream_idx, 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,
+};
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