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| author | Ronald S. Bultje <rsbultje@gmail.com> | 2010-02-10 18:30:55 +0000 |
|---|---|---|
| committer | Ronald S. Bultje <rsbultje@gmail.com> | 2010-02-10 18:30:55 +0000 |
| commit | 170870b77c8c71304f9eae946d49aa5e30a464bc (patch) | |
| tree | a7e26f4f191afc72acc7252e83688ffc8957a3de | |
| parent | 556aa7a1022c31566f64c9cb8fe8de0d4d5b44b2 (diff) | |
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
| -rw-r--r-- | libavformat/rtsp.c | 6 |
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 */ |