PeerStreamer

Replace @returns by @return.

Originally committed as revision 22729 to svn://svn.ffmpeg.org/ffmpeg/trunk

Add a timeout to the select() call. Patch by Sam Gerstein <sgerstein bluefinlab
com>.

Originally committed as revision 22718 to svn://svn.ffmpeg.org/ffmpeg/trunk

Simplify ff_rtsp_send_cmd_with_content_async, remove an unnecessary buffer

Originally committed as revision 22680 to svn://svn.ffmpeg.org/ffmpeg/trunk

Actually parse the auth headers in RTSP

Originally committed as revision 22677 to svn://svn.ffmpeg.org/ffmpeg/trunk

Add separate method/url parameters to the rtsp_send_cmd functions

Originally committed as revision 22675 to svn://svn.ffmpeg.org/ffmpeg/trunk

Make ff_rtsp_send_cmd simply call ff_rtsp_send_cmd_with_content

Originally committed as revision 22663 to svn://svn.ffmpeg.org/ffmpeg/trunk

Handle errors returned from ff_rtsp_read_reply in udp_read_packet properly

Originally committed as revision 22657 to svn://svn.ffmpeg.org/ffmpeg/trunk

Parse options in the RTSP URL only from the last question mark onwards

This helps if the URL (erroneously?) contains question marks within the path.

Originally committed as revision 22643 to svn://svn.ffmpeg.org/ffmpeg/trunk

Reindent

Originally committed as revision 22635 to svn://svn.ffmpeg.org/ffmpeg/trunk

Use the caller's RTSPMessageHeader in rtsp_setup_input_streams

Currently, the caller doesn't get the status_code and location for rediects,
since rtsp_setup_input_streams uses a copy of RTSPMessageHeader of its own.

Originally committed as revision 22630 to svn://svn.ffmpeg.org/ffmpeg/trunk

Reindent

Originally committed as revision 22546 to svn://svn.ffmpeg.org/ffmpeg/trunk

move ff_url_split() and ff_url_join() declarations to internal.h
those functions are not part of the public API

Originally committed as revision 22534 to svn://svn.ffmpeg.org/ffmpeg/trunk

RTSP muxer: Create the SDP with the numerical IP of the peer
instead of using the original host name

Originally committed as revision 22464 to svn://svn.ffmpeg.org/ffmpeg/trunk

Use rt->control_uri consequently instead of s->filename in all RTSP commands

Originally committed as revision 22403 to svn://svn.ffmpeg.org/ffmpeg/trunk

Reindent

Originally committed as revision 22322 to svn://svn.ffmpeg.org/ffmpeg/trunk

Localize the #define _SVID_SOURCE needed for inet_aton() to os_support.c

Originally committed as revision 22284 to svn://svn.ffmpeg.org/ffmpeg/trunk

Rename RTP depacketizer files from rtp_* to rtpdec_*

Originally committed as revision 22109 to svn://svn.ffmpeg.org/ffmpeg/trunk

Prefix non-static RTSP functions with ff_.

Originally committed as revision 21974 to svn://svn.ffmpeg.org/ffmpeg/trunk

Free metadata in chained RTP muxers in the RTSP muxer

This fixes a minor memory leak

Originally committed as revision 21970 to svn://svn.ffmpeg.org/ffmpeg/trunk

Add declarations and doxygen documentation of generic rtsp support functions
to rtsp.h, and make the functions non-static

Originally committed as revision 21968 to svn://svn.ffmpeg.org/ffmpeg/trunk

Don't follow RTSP redirects when used as a muxer

Originally committed as revision 21966 to svn://svn.ffmpeg.org/ffmpeg/trunk

Create AVFormatContext objects as private transport for output RTSP sessions

Originally committed as revision 21964 to svn://svn.ffmpeg.org/ffmpeg/trunk

Split out input-specific parts of rtsp_read_header() into its own, new,
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

Use mode=receive instead of mode=play if in RTSP muxer (instead of demuxer)
mode.

Patch by Martin Storsjö <$firstname $firstname st>.

Originally committed as revision 21912 to svn://svn.ffmpeg.org/ffmpeg/trunk

Rename RTSP_STATE_PLAYING to _STREAMING, since that better covers the
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

When using RTP-over-UDP, send dummy packets during stream setup, similar to
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

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

Use the control URI from the SDP (if present) rather than the input filename,
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>....

Change on rtsp_send_cmd() to the _async() version since we don't use the
response anyway.

Originally committed as revision 21370 to svn://svn.ffmpeg.org/ffmpeg/trunk

Pretty embarassing bug; we shouldn't use av_strlcatf() on an uninitialized
buffer, that is doomed to not work at some point.

Originally committed as revision 21368 to svn://svn.ffmpeg.org/ffmpeg/trunk

Add correct log context to av_log() calls in parse_rtpmap().

Originally committed as revision 21072 to svn://svn.ffmpeg.org/ffmpeg/trunk

Remove forward declarations.

Originally committed as revision 21020 to svn://svn.ffmpeg.org/ffmpeg/trunk

Remove residual use of the doxygen markup which is deprecated,
consistent with r19122.

Originally committed as revision 20989 to svn://svn.ffmpeg.org/ffmpeg/trunk

Add some "#if"s to avoid compiling the RTSP code when the RTSP demuxer
is disabled, and remove a useless "#if CONFIG_SDP_DEMUXER"

Originally committed as revision 20530 to svn://svn.ffmpeg.org/ffmpeg/trunk

Move some some functions around, so that splitting the SDP code out of
rtsp_read_packet() is simpler.

Originally committed as revision 20525 to svn://svn.ffmpeg.org/ffmpeg/trunk

Support 3xx redirection in rtsp

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

Suppress ?params in the rtsp uri

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...

Remove ancient redir demuxer.
HTTP supports redirection just fine without it.

Originally committed as revision 20361 to svn://svn.ffmpeg.org/ffmpeg/trunk

RTSP basic authentication, patch originally by Philip Coombes
(philip coombes zoneminder com), see "[PATCH]RTSP Basic Authentication"
thread on mailinglist.

Originally committed as revision 19905 to svn://svn.ffmpeg.org/ffmpeg/trunk

Support seeking as defined by the rfc

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

fix compilation with DEBUG defined

Originally committed as revision 19016 to svn://svn.ffmpeg.org/ffmpeg/trunk

Send dummy requests over the TCP connection (WMS wants GET_PARAMETER,
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

Increase the SDP buffer size (again!) and also increase the temporary
buffer size of the fmtp parameter buffer. For Vorbis RTP, 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,...

Remove slash-skipping code because the function called right after that
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

Free metadata if already allocated; fixes a memleak if the header occurs twice
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

Remove useless comment about something that is deprecated. See summary in
"[PATCH] rtsp.c small cleanups" thread on mailinglist.

Originally committed as revision 18124 to svn://svn.ffmpeg.org/ffmpeg/trunk

Merge functional code from get_word() and get_word_sep() into a single
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]...

Reindent after r18023.

Originally committed as revision 18024 to svn://svn.ffmpeg.org/ffmpeg/trunk

rtpmap is case-insensitive, see comment from Luca in "[PATCH] rtsp.c:
keep-alive" thread.

Originally committed as revision 17862 to svn://svn.ffmpeg.org/ffmpeg/trunk

Split rtsp_send_cmd() into two functions, one for the actual sending of the
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]...