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authorRonald S. Bultje <rsbultje@gmail.com>2010-02-10 18:30:55 +0000
committerRonald S. Bultje <rsbultje@gmail.com>2010-02-10 18:30:55 +0000
commit170870b77c8c71304f9eae946d49aa5e30a464bc (patch)
treea7e26f4f191afc72acc7252e83688ffc8957a3de /libavformat/rtsp.c
parent556aa7a1022c31566f64c9cb8fe8de0d4d5b44b2 (diff)
downloadffmpeg-170870b77c8c71304f9eae946d49aa5e30a464bc.tar.gz
Don't forget to set known audio parameters (samplerate, etc.) if the codec is
not supported in FFmpeg. This will cause crashes later because the samplerate is used to initialize the timebase. Originally committed as revision 21741 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtsp.c')
-rw-r--r--libavformat/rtsp.c6
1 files changed, 1 insertions, 5 deletions
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index d1738dd046..171d707cae 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -131,9 +131,8 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
if (c && c->name)
c_name = c->name;
else
- c_name = (char *) NULL;
+ c_name = "(null)";
- if (c_name) {
get_word_sep(buf, sizeof(buf), "/", &p);
i = atoi(buf);
switch (codec->codec_type) {
@@ -164,9 +163,6 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
break;
}
return 0;
- }
-
- return -1;
}
/* return the length and optionally the data */