function (rtsp_setup_input_streams()), as preparation for the upcoming
RTSP muxer.
Patch by Martin Storsjö <$firstname $firstname st>.
Originally committed as revision 21914 to svn://svn.ffmpeg.org/ffmpeg/trunk
don't send them when acting as a RTSP muxer.
Patch by Martin Storsjö <$firstname $firstname st>.
Originally committed as revision 21913 to svn://svn.ffmpeg.org/ffmpeg/trunk
argument, so we can use AVFormatContext->* here in the future.
Patch by Martin Storsjö <$firstname $firstname st>.
Originally committed as revision 21911 to svn://svn.ffmpeg.org/ffmpeg/trunk
future use of the rtsp* codebase for RTSP muxing.
Patch by Martin Storsjö <$firstname $firstname st>.
Originally committed as revision 21896 to svn://svn.ffmpeg.org/ffmpeg/trunk
be used eventually in the RTSP muxer (see thread "[PATCH] RTSP muxer, round
3" on mailinglist).
Patch by Martin Storsjö <$firstname $firstname st>.
Originally committed as revision 21862 to svn://svn.ffmpeg.org/ffmpeg/trunk
what e.g. RealPlayer does. This allows proper port forwarding setup in NAT-
based environments.
Patch by Martin Storsjö <$firstname at $firstname dot st>.
Originally committed as revision 21856 to svn://svn.ffmpeg.org/ffmpeg/trunk
not supported in FFmpeg. This will cause crashes later because the samplerate
is used to initialize the timebase.
Originally committed as revision 21741 to svn://svn.ffmpeg.org/ffmpeg/trunk
reading direct SDP files to set up UDP-based RTP-streams). Fixes
issue 1713. Patch by Jeremy Morton <ffmpeg game-point net>.
Originally committed as revision 21461 to svn://svn.ffmpeg.org/ffmpeg/trunk
if present. This fixes playback of a number of MS-RTSP streams, mostly these
for which playback contains a session key in the URI. Fixes issue 1697.
Patch by Alan Steremberg <$firstname dot $lastname () gmail com>.
Originally committed as revision 21381 to svn://svn.ffmpeg.org/ffmpeg/trunk
parts of FFmpeg. Also change a starting condition; while (condition) {
... bla = bla->next; } loop into a proper for() loop.
Originally committed as revision 21071 to svn://svn.ffmpeg.org/ffmpeg/trunk
sdp_read_packet -> rtsp_fetch_packet
This way describes slightly better what it does.
Originally committed as revision 20982 to svn://svn.ffmpeg.org/ffmpeg/trunk
All the error codes 3xx got managed the same way.
After setup/early play redirection will not be managed
REDIRECT method is yet to be supported (if somebody knows a server implementing
it please contact me)
Originally committed as revision 20369 to svn://svn.ffmpeg.org/ffmpeg/trunk
Right now rtsp demuxer receives it's ffmpeg specific params encoded in the url
That made the server receiving requests with the url ending with "?udp",
"?multicast" and "?tcp". That may or may not cause problems to servers with
overly strict or overly simple uri parsers
Patch from Armand Bendanan (name.surnameATfreeDOTfr)
Originally committed as revision 20363 to svn://svn.ffmpeg.org/ffmpeg/trunk
Transport:destination in rtsp is optional, c= line in sdp is compulsory
Patch from Armand Bendanan (name.surnameATfreeDOTfr)
Originally committed as revision 20362 to svn://svn.ffmpeg.org/ffmpeg/trunk
(philip coombes zoneminder com), see "[PATCH]RTSP Basic Authentication"
thread on mailinglist.
Originally committed as revision 19905 to svn://svn.ffmpeg.org/ffmpeg/trunk
implement RTCP/bye. See "[PATCH] rtsp.c: EOS support" thread from a few
months back.
Originally committed as revision 19517 to svn://svn.ffmpeg.org/ffmpeg/trunk
a PLAY with Range alone while in PLAY status should be interpreted
as an enqueue
a PAUSE followed by a PLAY with Range is the proper way to ask to
seek to a point.
See rfc2326
Originally committed as revision 19143 to svn://svn.ffmpeg.org/ffmpeg/trunk
Real wants OPTIONS) while the connection is idle, otherwise it will
be aborted after a short period (usually a minute). See the thread
"[PATCH] rtsp.c: keep-alive" on the mailinglist.
Originally committed as revision 18525 to svn://svn.ffmpeg.org/ffmpeg/trunk
qualification task, see "RTP/Vorbis payload implementation (GSoC qual
task)" thread on mailinglist.
Originally committed as revision 18509 to svn://svn.ffmpeg.org/ffmpeg/trunk
buffer size of the fmtp parameter buffer. For Vorbis RT(S)P, these
contain full Vorbis headers, which can be up to 12kb each, formatted
in base64, so 16kb total. Patch required for proper Vorbis/RTP playback,
submitted as GSoC qualification task in the thread "RTP/Vorbis payload
implementation (GSoC qual task)" by Colin McQuillan m.niloc googlemail
com.
Originally committed as revision 18508 to svn://svn.ffmpeg.org/ffmpeg/trunk
redir_isspace(char) to check if '\0' is a space or not. Therefore, we now
use memchr(), since then we can give the length of the string (i.e. the
length excluding the terminating '\0'). Fixes issue 919, see also the
follow-ups in the "[PATCH] rtsp.c small cleanups" mailinglist thread.
Originally committed as revision 18177 to svn://svn.ffmpeg.org/ffmpeg/trunk
statement (get_word_sep()) already does that all by itself. See summary in
"[PATCH] rtsp.c small cleanups" thread on mailinglist.
Originally committed as revision 18128 to svn://svn.ffmpeg.org/ffmpeg/trunk
same line as the if. See summary in "[PATCH] rtsp.c small cleanups" thread on
mailinglist.
Originally committed as revision 18127 to svn://svn.ffmpeg.org/ffmpeg/trunk
in a stream (e.g. malicious input, broken file, etc.). See summary in "[PATCH]
rtsp.c small cleanups" thread on mailinglist.
Originally committed as revision 18126 to svn://svn.ffmpeg.org/ffmpeg/trunk
it is '\0' rather than its content (char *p; if (p == '\0') instead of if
(*p == '\0')). See summary in "[PATCH] rtsp.c small cleanups" thread on
mailinglist.
Originally committed as revision 18125 to svn://svn.ffmpeg.org/ffmpeg/trunk
function, since they both do approximately the same thing. At the same time,
remove redir_isspace() altogether since code elsewhere (including
get_word_sep()) uses strchr() for the same purpose. See summary in "[PATCH]
rtsp.c small cleanups" thread.
Originally committed as revision 18122 to svn://svn.ffmpeg.org/ffmpeg/trunk
data packets or in addition to UDP data packets, over the RTSP/TCP connection.
See discussion in [PATCH] rtsp.c: read TCP server notifications/messages"
thread on mailinglist.
Originally committed as revision 18121 to svn://svn.ffmpeg.org/ffmpeg/trunk
Martin Storsjö
Ronald S. Bultje
Jeremy Morton
Alan Steremberg
Stefano Sabatini
Luca Barbato
Luca Abeni
Diego Biurrun
Kostya Shishkov
Baptiste Coudurier