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author | Ronald S. Bultje <rsbultje@gmail.com> | 2009-01-09 01:30:14 +0000 |
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committer | Ronald S. Bultje <rsbultje@gmail.com> | 2009-01-09 01:30:14 +0000 |
commit | e49906c3210514edb6220c37a51551d3c1485120 (patch) | |
tree | 09a1561ffa0770c7c30d4695053580f7dfd43389 | |
parent | 5ce0043367ed44dd17ece7aa6429e08900e1c4de (diff) | |
download | ffmpeg-e49906c3210514edb6220c37a51551d3c1485120.tar.gz |
Increase buffer size for RTP packet data because some ASF streams use a
manual, non-standard blocksize which is bigger than RTP_MAX_PACKET_LENGTH.
See "[PATCH] RTSP-MS 4/15: blocksize detection" thread on mailinglist.
Originally committed as revision 16502 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r-- | libavformat/rtsp.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 26bae089dd..464cffe3bc 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -1299,7 +1299,7 @@ static int rtsp_read_packet(AVFormatContext *s, RTSPState *rt = s->priv_data; RTSPStream *rtsp_st; int ret, len; - uint8_t buf[RTP_MAX_PACKET_LENGTH]; + uint8_t buf[10 * RTP_MAX_PACKET_LENGTH]; if (rt->server_type == RTSP_SERVER_REAL) { int i; |