Commit graph

294 commits

Author SHA1 Message Date
Mark Nauwelaerts
81fc784163 rtspsrc: do not set elements to PLAYING when doing seek in PAUSED 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
8599801cae rtspsrc: switch to rtp time based syncing when guessed appropriate 2011-09-19 11:52:08 +02:00
Mark Nauwelaerts
3e33a7a09f rtspsrc: configure rtcp interval if provided
... in PLAY response.
2011-09-19 11:51:47 +02:00
Mark Nauwelaerts
95b5ece2c9 rtspsrc: ensure some initial state variable setup
... which might otherwise be skipped if the PLAY command is issued before
the OPEN command had a chance to actually be acted upon.

Fixes #657376.
2011-09-09 10:53:08 +02:00
Mark Nauwelaerts
2603c2079d rtspsrc: add gtk-doc for new short-header property 2011-09-05 13:32:17 +02:00
Marc Leeman
ce276d903c rtspsrc: allow sending short RTSP requests to a server
Some encoders (Arecont) do not like the long OPTIONS sent at startup as sent by
GStreamer, but do accept the short header as sent by Live555.

This patch makes the extending the request optional by adding a property
(short-header).

Fixes #655805.

API: GstRTSPSrc:short-header
2011-09-05 13:26:06 +02:00
Edward Hervey
d08e0ccc48 rtspsrc: Properly error out if SDP contains no streams
Also fixes unitialized variable error on macosx.
2011-08-09 11:28:17 +02:00
Mark Nauwelaerts
9764b57b0a rtspsrc: set SOURCE flag at init time
Fixes #654816.
2011-07-25 12:44:38 +02:00
Mark Nauwelaerts
b98585df82 rtspsrc: fix seeking regression
... introduced when shuffling around code for the async implementation
by setting state of source (and udp sources) in _play before downstream
flushing is undone.
2011-07-12 15:13:25 +02:00
Mark Nauwelaerts
785247cfb3 rtspsrc: reset state tracking variable when appropriate
... so we don't end up interrupting an operation that should not be interrupted
based on the indication of a previous interruptable operation.
2011-06-06 12:59:23 +02:00
Miguel Angel Cabrera Moya
c39b7a5359 rtspsrc: uniform unknown message handling
Do the same processing in all the cases when an unknown message is received.
That is, give a warning.

https://bugzilla.gnome.org/show_bug.cgi?id=651059
2011-05-25 20:06:16 +02:00
Stefan Kost
be413185d0 rtspsrc: use EINVAL for missing url parameter
Fixes gcc warning about using uninitialized variable 'res'.
2011-05-18 10:22:27 +03:00
Mark Nauwelaerts
dc2ddea91b rtspsrc: also allow PAUSE to be interrupted
... as it is on the way out to NULL.

See #632504.
2011-05-17 11:56:47 +02:00
Mark Nauwelaerts
283e4e4afd rtspsrc: ensure proper closing and cleanup
... since the TEARDOWN sequence might not have had a chance to even start,
but at least connections should be closed (synchronously) and state cleaned up.

See #632504.
2011-05-17 11:56:38 +02:00
Mark Nauwelaerts
f7ddf811d7 rtspsrc: fix and improve async handling
Simplify the command handling; passing a command to thread means we really
want it to get the message, which means to always flush provided the command
can handle being interrupted.  Command thread indicates whether command
allows interruption and ensure non-flushing connection as it subsequently
needs it.

In particular, this also makes the TEARDOWN sequence interruptable
and also prevents races where _loop_ could miss a command and would
continue receiving (or at least trying to).

See #632504.
2011-05-17 11:56:22 +02:00
Mark Nauwelaerts
e6798ad54c rtspsrc: tweak post-seek loop handling 2011-05-17 11:55:40 +02:00
Wim Taymans
ddfcd8bbfd rtspsrc: open on play and pause when not done yet
With the async state changes, it is possible that we need to open the stream
before play and pause.
Also make sure we remember a previous open failure so that we don't keep trying
again.
2011-05-17 11:55:34 +02:00
Wim Taymans
6fe680934a rtspsrc: improve async handling
Simplify the command handling, only continue looping when we have not received
another command or when the previous loop was successfull.
Avoid looping on a disconnected socket.
2011-05-17 11:55:32 +02:00
Wim Taymans
2513207433 rtspsrc: rework reconnect code
Use the same async code path to implement reconnects.
Make sure we only post progress messages when doing async things.
2011-05-17 11:55:29 +02:00
Wim Taymans
c27c10f8f4 rtspsrc: small cleanups
Make sure we cancel the previous task when queuing a new one.
Move the messages to a central place so we can more easily post them.
2011-05-17 11:55:27 +02:00
Wim Taymans
852c6e11cd rtspsrc: don't post errors when interrupting 2011-05-17 11:55:24 +02:00
Wim Taymans
220e47adcf rtspsrc: implement more async handling
Remove some old locks.
Make sure we never go into the loop function when flushing.
2011-05-17 11:55:20 +02:00
Wim Taymans
2873585238 rtspsrc: first attempt at async implementation 2011-05-17 11:55:18 +02:00
Wim Taymans
547c97f590 rtspsrc: handle * control correctly
Parse session control attributes when no media control attribute is
present. Threat * control attributes as an empty string, just like the
spec says.

Fixes #646800
2011-04-05 17:12:28 +02:00
Mark Nauwelaerts
234609844e rtspsrc: perform post-flush state tricks downstream to upstream
... so downstream is set when upstream resumes data flow.
2011-04-04 11:49:00 +02:00
Mark Nauwelaerts
226a7cb32e rtspsrc: distribute new base_time to manager children following flush seek
... by forcing a state changed to PLAYING, which should otherwise be a
no-op as elements should already be in that state.

In particular, jitterbuffer needs new base_time as soon as possible to perform
proper timing (e.g. eos timeout handling) and can't wait for the new base_time
that will be distributed when the whole pipeline returns to PLAYING.

See bug #646397.
2011-04-04 11:49:00 +02:00
Mark Nauwelaerts
2738917852 rtspsrc: improve recovery from failed seek
In case server-side fails to perform seek, i.e. PLAY at non-zero requested
position, recovery so far would arrange for streaming to continue, albeit
having lost position tracking in the process.  So, query position prior
to seek and use upon failed seek.
2011-03-09 17:18:09 +01:00
Miguel Angel Cabrera Moya
3cca27ced1 rtspsrc: fix minor leaks when handling server requests.
https://bugzilla.gnome.org/show_bug.cgi?id=640163
2011-02-14 11:33:18 +01:00
Stefan Kost
6f6b2a7efc rtspsrc: strip trailing spaces 2011-02-07 17:08:47 +02:00
Stefan Kost
5e071d51f2 rtpsrc: set multiple properties in one go
There is no need for separate g_object_set() calls here.
2011-02-07 17:07:42 +02:00
Tim-Philipp Müller
08855b45b6 rtspsrc: don't leak url string
https://bugzilla.gnome.org/show_bug.cgi?id=640064
2011-01-20 13:46:44 +00:00
Wim Taymans
bc0824181b rtspsrc: don't confuse return values
Return a return value of the right type.
2011-01-05 18:33:41 +01:00
Stefan Kost
c9e0db6469 rtspsrc: remove unused variables when debug-logging disabled 2011-01-03 20:17:47 +02:00
Wim Taymans
dc221c0219 rtspsrc: increase udp buffer size
Set a bigger UDP buffer size by default to reduce packet loss with
high bitrate streams.
2011-01-03 15:40:11 +01:00
Tim-Philipp Müller
96830324a5 rtspsrc: serialise/deserialise floats without changing locale
Use g_ascii_dtostr() and g_ascii_strtod() to serialise/deserialise
floating point numbers, instead of ugly hacks that switch locale
before and after calling libc functions (which is not a good idea
in a multi-threaded application).
2010-12-29 15:54:46 +00:00
Wim Taymans
2a49d34c3e rtspsrc: on-npt-stop is a manager signal 2010-12-23 16:25:15 +01:00
Wim Taymans
12bc7258b9 rtspsrc: improve RTP session handling
Store the RTP session in the stream so that we can more efficiently
perform actions on the stream based on RTP signals.
2010-12-23 15:24:29 +01:00
Tim-Philipp Müller
7759ad0db2 docs: update rtspsrc docs, rtpbin is not in -bad any more 2010-12-22 13:04:42 +00:00
Mark Nauwelaerts
287894a89a rtspsrc: mark DISCONT when resuming PLAY
In particular, when streaming interleaved, this arranges for setting a new
timestamp on outgoing buffer so downstream can appropriate reset
to a change in (rtp)time.
2010-12-10 12:11:15 +01:00
Mark Nauwelaerts
c25625c31c rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response 2010-12-10 12:09:49 +01:00
Mark Nauwelaerts
52b5929a2b rtspsrc: add and use auto buffering mode
... which selects BUFFER for a non-live stream, and otherwise SLAVE.

Fixes #633088.
2010-12-10 12:09:32 +01:00
Mark Nauwelaerts
0f2373cbd1 rtspsrc: reset session manager base time when flushing
... as rtpbin uses running time to handle rtpjitterbuffer's buffer mode pauses.
2010-12-03 15:50:17 +01:00
Mark Nauwelaerts
148af2235e rtspsrc: include range request for all streams with non-aggregate control 2010-12-03 15:50:17 +01:00
Mark Nauwelaerts
dedf145316 rtspsrc: fix debug statement 2010-12-03 15:50:17 +01:00
Wim Taymans
7ed250c793 rtspsrc: select multicast transports in a smarter way
When we see a multicast address in the SDP connection, only try to negotiate a
multicast transport with the server.

Fixes #634093
2010-12-02 19:16:47 +01:00
Mark Nauwelaerts
b6b0de0c49 rtspsrc: handle stale digest authentication session data
In particular, handle Unauthorized server response when trying to convey
keep-alive.

Fixes #635532.
2010-11-29 17:34:28 +00:00
Mark Nauwelaerts
ca7870de49 rtspsrc: fix duration reporting
Init segment prior to storing duration info in it.

Fixes #632548.
2010-10-19 16:47:20 +02:00
Wim Taymans
ee7207aa3e rtspsrc: mark as a source
Mark the rtspsrc element as a source.
Requires 0.10.31.1 now
2010-10-11 15:12:51 +02:00
René Stadler
0cfe24d132 rtspsrc: fix missing null-terminator in protocols array
Fixes random crash regression from commit ae84ae.
2010-09-28 16:21:48 +03:00
Wim Taymans
ef29a59903 rtspsrc: don't add /UDP in the transport, it's the default
don't add the default UDP lower-transport, some servers don't seem to like it.

Fixes #630500
2010-09-24 16:26:20 +02:00