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-rw-r--r--libavformat/pmpdec.c91
1 files changed, 54 insertions, 37 deletions
diff --git a/libavformat/pmpdec.c b/libavformat/pmpdec.c
index 484be43ce9..71f450e9d3 100644
--- a/libavformat/pmpdec.c
+++ b/libavformat/pmpdec.c
@@ -1,21 +1,21 @@
/*
- * PMP demuxer
+ * PMP demuxer.
* Copyright (c) 2011 Reimar Döffinger
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -23,18 +23,18 @@
#include "avformat.h"
#include "internal.h"
-typedef struct PMPContext {
- int cur_stream;
- int num_streams;
- int audio_packets;
- int current_packet;
+typedef struct {
+ int cur_stream;
+ int num_streams;
+ int audio_packets;
+ int current_packet;
uint32_t *packet_sizes;
- int packet_sizes_alloc;
+ int packet_sizes_alloc;
} PMPContext;
-static int pmp_probe(AVProbeData *p)
-{
- if (!memcmp(p->buf, "pmpm\1\0\0\0", 8))
+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;
}
@@ -44,11 +44,13 @@ static int pmp_header(AVFormatContext *s)
PMPContext *pmp = s->priv_data;
AVIOContext *pb = s->pb;
int tb_num, tb_den;
- int index_cnt;
+ uint32_t index_cnt;
int audio_codec_id = AV_CODEC_ID_NONE;
int srate, channels;
- int i;
+ unsigned i;
uint64_t pos;
+ int64_t fsize = avio_size(pb);
+
AVStream *vst = avformat_new_stream(s, NULL);
if (!vst)
return AVERROR(ENOMEM);
@@ -65,7 +67,7 @@ static int pmp_header(AVFormatContext *s)
av_log(s, AV_LOG_ERROR, "Unsupported video format\n");
break;
}
- index_cnt = avio_rl32(pb);
+ index_cnt = avio_rl32(pb);
vst->codec->width = avio_rl32(pb);
vst->codec->height = avio_rl32(pb);
@@ -73,14 +75,14 @@ static int pmp_header(AVFormatContext *s)
tb_den = avio_rl32(pb);
avpriv_set_pts_info(vst, 32, tb_num, tb_den);
vst->nb_frames = index_cnt;
- vst->duration = 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_WARNING, "AAC is not yet correctly supported\n");
+ av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n");
audio_codec_id = AV_CODEC_ID_AAC;
break;
default:
@@ -89,26 +91,38 @@ static int pmp_header(AVFormatContext *s)
}
pmp->num_streams = avio_rl16(pb) + 1;
avio_skip(pb, 10);
- srate = avio_rl32(pb);
+ 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->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);
}
- 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;
}
@@ -119,11 +133,15 @@ static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
int ret = 0;
int i;
- if (pb->eof_reached)
+ 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) {
+ avpriv_request_sample(s, "0 audio packets");
+ return AVERROR_PATCHWELCOME;
+ }
num_packets = (pmp->num_streams - 1) * pmp->audio_packets + 1;
avio_skip(pb, 8);
pmp->current_packet = 0;
@@ -138,7 +156,7 @@ static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
pmp->packet_sizes[i] = avio_rl32(pb);
}
ret = av_get_packet(pb, pkt, pmp->packet_sizes[pmp->current_packet]);
- if (ret > 0) {
+ if (ret >= 0) {
ret = 0;
// FIXME: this is a hack that should be removed once
// compute_pkt_fields() can handle timestamps properly
@@ -146,14 +164,13 @@ static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
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)
+ 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_idx, int64_t ts, int flags)
+static int pmp_seek(AVFormatContext *s, int stream_index, int64_t ts, int flags)
{
PMPContext *pmp = s->priv_data;
pmp->cur_stream = 0;
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