[FFmpeg-cvslog] mov: Make sure no streams after the header lack a timescale.

[Michael Niedermayer]

ffmpeg | branch: master | Michael Niedermayer <michaelni at gmx.at> | Sat Nov 10 13:28:26 2012 +0100| [eb3dc237051242d28570c2fa13108c05793f5a4e] | committer: Michael Niedermayer

mov: Make sure no streams after the header lack a timescale.

Fixes division by 0

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=eb3dc237051242d28570c2fa13108c05793f5a4e
---

 libavformat/mov.c |   18 ++++++++++++------
 1 file changed, 12 insertions(+), 6 deletions(-)

diff --git a/libavformat/mov.c b/libavformat/mov.c
index d7d7f21..cd260b4 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -2170,6 +2170,16 @@ static int mov_open_dref(AVIOContext **pb, const char *src, MOVDref *ref,
     return AVERROR(ENOENT);
 }
 
+static void fix_timescale(MOVContext *c, MOVStreamContext *sc)
+{
+    if (sc->time_scale <= 0) {
+        av_log(c->fc, AV_LOG_WARNING, "stream %d, timescale not set\n", sc->ffindex);
+        sc->time_scale = c->time_scale;
+        if (sc->time_scale <= 0)
+            sc->time_scale = 1;
+    }
+}
+
 static int mov_read_trak(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 {
     AVStream *st;
@@ -2197,12 +2207,7 @@ static int mov_read_trak(MOVContext *c, AVIOContext *pb, MOVAtom atom)
         return 0;
     }
 
-    if (sc->time_scale <= 0) {
-        av_log(c->fc, AV_LOG_WARNING, "stream %d, timescale not set\n", st->index);
-        sc->time_scale = c->time_scale;
-        if (sc->time_scale <= 0)
-            sc->time_scale = 1;
-    }
+    fix_timescale(c, sc);
 
     avpriv_set_pts_info(st, 64, 1, sc->time_scale);
 
@@ -3180,6 +3185,7 @@ static int mov_read_header(AVFormatContext *s)
     for (i = 0; i < s->nb_streams; i++) {
         AVStream *st = s->streams[i];
         MOVStreamContext *sc = st->priv_data;
+        fix_timescale(mov, sc);
         if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO && st->codec->codec_id == AV_CODEC_ID_AAC) {
             st->skip_samples = sc->start_pad;
         }