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
command and a second, new function to read the reply to this command. This
will make it possible to read server notices that are not in response to a
command in future versions, such as EOS or interrupt notices. See "[PATCH]
rtsp.c: split rtsp_send_cmd() in a send- and a receive-function" thread.
Originally committed as revision 17797 to svn://svn.ffmpeg.org/ffmpeg/trunk
fd2) and one was just removed, so naming the other "fd1" is counter-intuitive.
See "[RFC] rtsp.c EOF support" thread.
Originally committed as revision 17780 to svn://svn.ffmpeg.org/ffmpeg/trunk
associated with the I/O handle (e.g. the fd returned by open()). See
"[RFC] rtsp.c EOF support" thread.
There were previously some URI-specific implementations of the same idea,
e.g. rtp_get_file_handles() and udp_get_file_handle(). All of these are
deprecated by this patch and will be removed at the next major API bump.
Originally committed as revision 17779 to svn://svn.ffmpeg.org/ffmpeg/trunk
RTSP-MS UDP" thread on mailinglist.
Basically, UDP setup needs to be done in a particular order (first rtx
on two UDP ports (one for RTP, one for RTCP), then the other streams over
one, single port for all of them together). Not doing this correctly results
in a "461" error (invalid transport) during setup.
Originally committed as revision 17777 to svn://svn.ffmpeg.org/ffmpeg/trunk
sessions.
This type is used in RTP/ASF (served by WMS servers), and is required to
make UDP sessions work, but breaks TCP sessions. Therefore, we disable setup
for application streams in TCP/WMS streams.
See discussion in "[PATCH] RTSP-MS 8/15: fix RTSP-MS UDP" thread.
Originally committed as revision 17776 to svn://svn.ffmpeg.org/ffmpeg/trunk
structure is meant to represent. See "[PATCH] rtsp.[ch]: RTSPHeader ->
RTSPServerResponse" and "[PATCH] document rtsp.h" threads on ML.
Originally committed as revision 17504 to svn://svn.ffmpeg.org/ffmpeg/trunk
"cur_transport_priv", as discussed in the "[PATCH] rtsp.h: rename tx
variables" thread.
Originally committed as revision 17012 to svn://svn.ffmpeg.org/ffmpeg/trunk
servers when trying to set up a session over TCP:
- add the interleave property
- add unicast, only for WMS (since it is normally only UDP, but WMS expects it
for UDP and TCP)
- add mode=play
See discussion in "[PATCH] RTSP-MS 9/15: add interleave property to the TCP
transport line of the SETUP request" thread on mailinglist.
Originally committed as revision 16913 to svn://svn.ffmpeg.org/ffmpeg/trunk
subsequent a= lines from the m= block to be applied to the previous
m= line, thus breaking otherwise functional RTP streams. See discussion in
[PATCH] RTSP-MS 7/15: parse and allow unknown m= line codes" thread on
mailinglist.
Originally committed as revision 16737 to svn://svn.ffmpeg.org/ffmpeg/trunk
come directly after each m= line if required. See "[PATCH] RTSP-MS 5-6/15:
parse only the last m= line stream per rtpmap line" thread on ML.
Originally committed as revision 16509 to svn://svn.ffmpeg.org/ffmpeg/trunk
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
to is a Microsoft Windows Media Server (the field will be "WMServer/version").
See "[PATCH] RTSP-MS 3/15: Add Windows Media Server type" thread on
mailinglist.
Originally committed as revision 16472 to svn://svn.ffmpeg.org/ffmpeg/trunk
access to these structures in functions that will be located in rtp_asf.c.
See "[PATCH] RTSP-MS 2/15: export RTSPState and RTSPStream" mailinglist
thread.
Originally committed as revision 16471 to svn://svn.ffmpeg.org/ffmpeg/trunk
this RTSP/RDT session. This basically implies full RDT support, including
stream selection in ffmpeg and multi-stream backupping in ffmpeg (by mapping
each stream to an output). See "[PATCH] RTSP/RDT: subscriptions" thread on
mailinglist.
Originally committed as revision 16469 to svn://svn.ffmpeg.org/ffmpeg/trunk
streams" (a single RTSPStream / RDTDemuxContext can represent several
AVStreams, that's just how Real/RDT was designed...). This will fill in
most of the AVStream/AVCodecContext header fields, similar to reading a
RM file header would. See "[PATCH] multi-stream MDPR parsing" thread on
mailinglist.
Originally committed as revision 16468 to svn://svn.ffmpeg.org/ffmpeg/trunk
each "rule" described in the ASMRuleBook. Each rule represents a stream
of identical content compared to other streams in the same rulebook, but
with a possibly different codec/bitrate/etc. See "[PATCH] rdt.c: ASM
rulebook parsing and AVStream creation" thread on mailinglist.
Originally committed as revision 16466 to svn://svn.ffmpeg.org/ffmpeg/trunk
This function is called in rtsp.c for each a= line in the SDP of the Describe
response after m= RTSP stream descriptors. The function prototype used to
take an AVStream argument. For RDT, however, every RTSPStream represents
a set of streams of identical content, and can thus represent multiple
AVStreams. Therefore, it should not take an AVStream as argument. This
patch modifies it to accept a AVFormatContext (of the RTSP/SDP demuxer)
instead. See discussion in "[PATCH/RFC] change function prototype of
parse_sdp_a_line" thread on ML.
Originally committed as revision 16024 to svn://svn.ffmpeg.org/ffmpeg/trunk
