[Ffmpeg-devel] FFServer and RTSP
Luca Abeni
lucabe72
Wed Apr 11 15:37:17 CEST 2007
Hi all,
So, I committed my two patches, and now ffserver with RTSP / RTP is
"kind of" working. I say "kind of" because it frequently crashes. To get
a nice crash, just start playing an RTSP url (with vlc, for example),
then stop it, and play it again...
While searching for the cause of the crash, I found out that there are a
lot bugs in ffserver that can cause this crash.
- First of all, rtsp_cmd_teardown() is calling close_connection(rtp_c),
which frees rtp_c and all its buffers, and then is accessing
rtp_c->session_id and other freed fields... :(
- Then, there are some other strange things like the
c->rtp_ctx[stream_index]->streams[0]->priv_data field which is not
dynamically allocated (every time that an RTSP connection is createdm
this filed is copied from an AVStream structure that is allocated during
the initialization of the program), but is av_free()d when the
connection ends (this is why ffserver crashes when the second connection
ends, I think).
Even not considering the crashes, ffserver has some other "funny
behaviour" when streaming RTP: for example, the first two frames read
from the file have the same PTSs and DTSs, sometimes resulting in a
"error, non monotone timestamps 45000 >= 45000 st:1" error when trying
to stream them (I say sometimes, because if the first PTS is 0 then the
error does not occourr...). I think the error is due to the fact that
the file is opened (with av_open_input_file()), but
av_find_stream_info() is not invoked...
Is it legal to open a file for reading (av_open_input_file()), and start
reading frames from it without performing an av_find_stream_info()?
Anyway, it seems to me that ffserver is in a very bad shape regarding
RTSP/RTS streaming...
So I'd like to have an opinion from other people (an opinion from the
official maintainer would be appreciated :): is there any hope in fixing
ffserver for RTP streaming? (did it ever work, at least one time?).
Or would it be simpler to start developing a new RTP streaming
application from scratch? I am beginning to think the second option
would require less work... :(
Thanks,
Luca
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