Tim-Philipp Müller
230cf41cc9
Fix FSF address
...
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00
Wim Taymans
adb70e89f9
rtspsrc: remove unused include
2012-10-10 12:05:34 +02:00
Tim-Philipp Müller
8b20603f8b
rtspsrc: answer URI query
...
Without this, something also answered the query
with TRUE but without setting a uri, not sure
what that was..
2012-09-21 23:33:47 +01:00
Daniela
03fbd7ec6e
rtspsrc: avoid leak
...
When setup fails, make sure to cleanup afterwards.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673509
2012-09-07 16:33:18 +02:00
Aleix Conchillo Flaque
4a200b670f
rtp: make rtp packet probation configurable (bug #682512 )
2012-08-30 21:49:57 +02:00
Tim-Philipp Müller
4bb52bbadf
docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert
2012-08-27 21:20:30 +01:00
Aleix Conchillo Flaque
8d864dbbfc
rtspsrc: make jitterbuffer drop-on-latency available ( fix #682055 )
...
Conflicts:
gst/rtsp/gstrtspsrc.h
2012-08-22 10:39:19 +02:00
Mark Nauwelaerts
a549b0bf2c
rtspsrc: manage race between connection closing and flushing
...
... where the former can happen in task thread and the latter in mainloop
upon downward state change.
2012-08-03 14:10:32 +02:00
Wim Taymans
ef38efc2d7
rtsp: go and stay in the loop function on PLAY
...
When we have a PLAY request, go into the LOOP function next. When we are
looping, keep on looping until we are told otherwise.
This fixed rtsp and TCP connections.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680551
2012-07-25 12:50:01 +02:00
Wim Taymans
943b56ff8e
rtsp: set caps after activating the pad
2012-07-25 12:49:35 +02:00
Maria Giovanna Chiossa
561b131e1a
rtspsrc: also set UDP buffer size in multicast
...
Also set the UDP buffer size in multicast mode.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675448
2012-07-19 15:26:36 +02:00
Sebastian Dröge
aeafc3a093
gst: Implement segment-done event
2012-07-05 13:13:09 +02:00
Tim-Philipp Müller
456847c66b
rtspsrc: update for gst_element_make_from_uri() changes
2012-06-23 14:57:28 +01:00
Wim Taymans
30d3dfee36
update for task api change
2012-06-20 10:33:42 +02:00
Wim Taymans
694be55c05
rtspsrc: Don't reset time in flush-stop
...
Don't reset the time in flush-stop. Live sources can do this flush in the
playing state and so the pipeline will never have a chance to update the
base_time of the elements, which only happens when going from paused to
playing.
2012-06-14 08:58:58 +02:00
Wim Taymans
935472aba7
rtspsrc: Rework the async state handling
...
Always send the flushing events to the udp elements now that basesrc supports
this. This makes sure a segment event is sent correctly after a flush.
Keep track of the currently executing command and make it possible to specify
what command you want to cancel when starting a new async command.
See https://bugzilla.gnome.org/show_bug.cgi?id=677905
2012-06-12 16:05:40 +02:00
Wim Taymans
eb982e4bbe
rtspsrc: only reset the manager object when we did a seek
...
Only reset the manager object when we used a Range header, ie. when we did a
seek. Otherwise we just paused and we can resume just fine.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677475
2012-06-07 12:11:14 +02:00
Maria Giovanna Chiossa
ff019d05f6
rtsp: add the Scale header when needed
...
Setting GST_SEEK_FLAG_SKIP when sending a seek event in rtspsrc should
set the "Scale" field in the rtsp PLAY header.
Because the boolean "src->skip" is set after the call, "Speed" instead
of "Scale" is always set. Move the assignment before issuing the _play
request.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676618
2012-05-24 09:57:31 +02:00
Tim-Philipp Müller
e09ae5736d
Use new gst_element_class_set_static_metadata()
2012-04-10 00:51:41 +01:00
Wim Taymans
3d61d12e03
update for buffer api change
2012-03-30 18:15:34 +02:00
Wim Taymans
c44cd8f55b
Merge branch 'master' into 0.11
...
unport gdkpixbuf
not merged: https://bugzilla.gnome.org/show_bug.cgi?id=654850
Conflicts:
docs/plugins/Makefile.am
docs/plugins/gst-plugins-good-plugins-docs.sgml
docs/plugins/gst-plugins-good-plugins-sections.txt
docs/plugins/gst-plugins-good-plugins.hierarchy
docs/plugins/inspect/plugin-avi.xml
docs/plugins/inspect/plugin-png.xml
ext/flac/gstflacdec.c
ext/flac/gstflacdec.h
ext/libpng/gstpngdec.c
ext/libpng/gstpngenc.c
ext/speex/gstspeexdec.c
gst/audioparsers/gstflacparse.c
gst/flv/gstflvmux.c
gst/rtp/gstrtpdvdepay.c
gst/rtp/gstrtph264depay.c
2012-03-22 11:53:24 +01:00
Marc Leeman
b4756db358
gstrtspsrc: disable RTSP keep-alive on request
2012-03-12 15:14:21 +01:00
Sebastian Dröge
f2e569cde8
rtspsrc: Use correct enum for return values
2012-03-06 14:18:33 +01:00
Wim Taymans
ca9532ccc5
update for new memory api
2012-02-22 02:10:33 +01:00
Wim Taymans
9365f12d6e
GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING
2012-02-08 16:43:30 +01:00
Sebastian Dröge
0b517ce9fb
Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11
2012-01-25 12:49:34 +01:00
Sebastian Dröge
10554b271f
Merge branch 'master' into 0.11
...
Conflicts:
ext/flac/gstflacdec.c
ext/jpeg/gstjpegenc.c
ext/pulse/pulsesink.c
sys/v4l2/gstv4l2src.c
2012-01-25 12:49:11 +01:00
Wim Taymans
b4630dd3e0
more memory API porting
2012-01-25 12:30:29 +01:00
Mark Nauwelaerts
a224ffb971
rtspsrc: simplify internal src event debug logging
...
... which avoids almost superfluous obtaining of rtsp element.
2012-01-20 17:10:57 +01:00
Mark Nauwelaerts
018852ddc2
rtspsrc: avoid NULL string comparison
2012-01-20 17:10:54 +01:00
Wim Taymans
1584806634
port to new gthread API
2012-01-19 11:33:53 +01:00
Sebastian Dröge
305901c7cc
rtspsrc: Update for the new GIO versions of the udp elements
2012-01-17 16:49:10 +01:00
Sebastian Dröge
93e3ed5a86
Merge branch 'master' into 0.11
...
Conflicts:
ext/cairo/gsttextoverlay.c
ext/pulse/pulseaudiosink.c
gst/audioparsers/gstaacparse.c
gst/avi/gstavimux.c
gst/flv/gstflvmux.c
gst/interleave/interleave.c
gst/isomp4/gstqtmux.c
gst/matroska/matroska-demux.c
gst/matroska/matroska-mux.c
gst/matroska/matroska-mux.h
gst/matroska/matroska-read-common.c
gst/multifile/gstmultifilesink.c
gst/multipart/multipartmux.c
gst/shapewipe/gstshapewipe.c
gst/smpte/gstsmpte.c
gst/udp/gstmultiudpsink.c
gst/videobox/gstvideobox.c
gst/videocrop/gstaspectratiocrop.c
gst/videomixer/videomixer.c
gst/videomixer/videomixer2.c
gst/wavparse/gstwavparse.c
po/ja.po
po/lv.po
po/sr.po
tests/check/Makefile.am
tests/check/elements/qtmux.c
tests/check/elements/rgvolume.c
2012-01-10 14:32:32 +01:00
Wim Taymans
5fd2b7abe3
GST_FLOW_UNEXPECTED -> GST_FLOW_EOS
2012-01-03 15:26:21 +01:00
Tim-Philipp Müller
b8b8454bcb
Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly
...
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
2011-12-12 09:46:27 +00:00
Wim Taymans
d0b936acc7
rtspsrc: remove unused flush param
2011-12-06 13:59:52 +01:00
Wim Taymans
ac849ec2b3
fix for element flag updates
2011-11-28 16:57:24 +01:00
Vincent Penquerc'h
c0e101e93f
various: fix pad template leaks
...
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:30:27 +00:00
Tim-Philipp Müller
87aa29d2cf
rtspsrc: make connection-speed property a guint64
2011-11-24 01:19:32 +00:00
Wim Taymans
105650127e
add parent to pad functions
2011-11-17 15:02:55 +01:00
Wim Taymans
6190312214
add parent to query function
2011-11-16 17:27:13 +01:00
Tim-Philipp Müller
c27bbe4be2
Update for GstURIHandler get_protocols() changes
2011-11-13 23:44:44 +00:00
Tim-Philipp Müller
a150d1e734
soup, pushfile, rtsp, udp, v4l2: update for GstURIHandler API changes
2011-11-13 18:50:51 +00:00
Wim Taymans
c48df77320
update for probe api changes
2011-11-08 11:18:06 +01:00
Wim Taymans
de020130e6
fix for probe updates
2011-11-07 17:14:17 +01:00
Wim Taymans
768e3826ab
more template fixes
2011-11-04 17:39:15 +01:00
Wim Taymans
a95acb7122
make %u in all request pad templates
2011-11-04 11:58:22 +01:00
Wim Taymans
0560ab53c0
update for new task api
2011-11-02 09:06:37 +01:00
Wim Taymans
9a8a8e72c8
structure: fix for api update
2011-11-02 09:06:37 +01:00
Tim-Philipp Müller
9f77b02b15
Update for pad API changes
...
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
2011-11-01 00:52:28 +00:00
Wim Taymans
87fbd1e784
Merge branch 'master' into 0.11
...
Conflicts:
common
ext/pulse/pulsesink.c
ext/soup/gstsouphttpclientsink.c
gst/audioparsers/gstaacparse.c
gst/audioparsers/gstac3parse.c
gst/rtp/gstrtph264depay.c
gst/rtpmanager/gstrtpjitterbuffer.c
gst/rtpmanager/rtpjitterbuffer.c
gst/rtsp/gstrtspsrc.c
sys/ximage/gstximagesrc.c
2011-09-28 12:44:59 +02:00
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
Wim Taymans
33f18b8ea4
Merge branch 'master' into 0.11
...
Conflicts:
gst/audioparsers/gstamrparse.c
gst/isomp4/qtdemux.c
2011-09-06 16:06:25 +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
Wim Taymans
4bb2b140e9
Merge branch 'master' into 0.11
...
Conflicts:
sys/v4l2/v4l2src_calls.c
2011-08-16 18:35:53 +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
Wim Taymans
4121021bb2
Merge branch 'master' into 0.11
...
Conflicts:
ext/pulse/pulsesink.c
ext/pulse/pulsesrc.c
gst/audioparsers/gstac3parse.c
gst/rtp/gstrtph264depay.c
gst/rtp/gstrtph264pay.c
gst/rtpmanager/gstrtpssrcdemux.c
2011-08-03 18:25:30 +02:00
Mark Nauwelaerts
9764b57b0a
rtspsrc: set SOURCE flag at init time
...
Fixes #654816 .
2011-07-25 12:44:38 +02:00
Wim Taymans
9c087d7d85
Merge branch 'master' into 0.11
2011-07-15 17:06:39 +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
Wim Taymans
f0749ed617
rtsp: fix for uri changes
2011-06-22 16:41:13 +02:00
Wim Taymans
e221908169
rtsp: fix for flush_stop API change
2011-06-13 17:14:51 +02:00
Wim Taymans
eed80e2dd3
-good: update for buffer API change
2011-06-13 16:33:57 +02:00
Wim Taymans
c731cd3d95
rtsp: port to 0.11
2011-06-09 17:52:34 +02:00
Wim Taymans
710fa239d5
Merge branch 'master' into 0.11
2011-06-08 18:06:56 +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
Wim Taymans
0b1bdcf7cb
Merge branch 'master' into 0.11
...
Conflicts:
sys/ximage/ximageutil.c
2011-06-02 18:51:29 +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
Wim Taymans
d89790d545
Merge branch 'master' into 0.11
...
Conflicts:
gst/avi/gstavidemux.c
gst/rtp/gstrtpac3depay.c
gst/rtp/gstrtpg726depay.c
gst/rtp/gstrtpmpvdepay.c
gst/videofilter/gstgamma.c
2011-05-24 17:34:19 +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
Wim Taymans
e15651816e
Merge branch 'master' into 0.11
2011-05-17 16:13:59 +02:00
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
77acc618e1
use G_DEFINE_TYPE some more
2011-04-19 17:35:47 +02:00
Wim Taymans
c124ba1489
Merge branch 'master' into 0.11
...
Conflicts:
gst/rtsp/gstrtspsrc.c
2011-04-05 17:20:08 +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
Wim Taymans
f67c95d826
rtsp/udp: port to 0.11
2011-04-05 17:06:41 +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
Wim Taymans
8f22a09dc4
Merge branch 'master' into 0.11-fdo
2011-03-28 20:50:59 +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
Wim Taymans
759a3507d7
Merge branch 'master' into 0.11
...
Conflicts:
configure.ac
2011-02-28 11:58:05 +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
Wim Taymans
1d57ec6a6e
rtspsrc: use _object_ref_sink() when we can
2010-12-07 11:42:15 +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
Wim Taymans
8f2d254e24
rtspsrc: don't clear sdp when set as uri
...
when we set the SDP with an uri, don't clear it when we go to READY.
2010-09-10 18:06:48 +02:00
Wim Taymans
7698d8bc4a
rtspsrc: use sdp uri parse method
...
Use the sdp parse method that does proper uri escaping.
2010-09-10 18:02:04 +02:00
Wim Taymans
ae84ae1b36
rtspsrc: add rtsp-sdp protocol support
...
Allow setting an SDP with the rtsp-sdp:// url.
Based on patch from Marco Ballesio.
See #628214
2010-09-10 12:14:21 +02:00
American Dynamics
5999e8e716
rtspsrc: Add property to configure udpsrc buffer size
...
Add a new udp-buffer-size property to configure the buffer-size on the udpsrc
elements.
Fixes #628058
2010-09-06 12:22:11 +02:00
Wim Taymans
e4f8144bbf
rtspsrc: implement custom event handler
...
Extend the _push_event() function so that it can also send events to the udp
sources when asked.
Implement a custum send_event function that correctly dispatches the downstream
events in TCP mode. This fixes sending EOS to rtspsrc and have it push the EOS
downstream.
2010-09-06 10:45:23 +02:00
Sebastian Dröge
d224251df4
rtspsrc: Don't use GST_FLOW_IS_FATAL() and GST_FLOW_IS_SUCCESS()
2010-09-04 14:52:10 +02:00
Wim Taymans
9dcfed0a5b
rtspsrc: don't reuse udp sockets
...
Don't reuse sockets but make the udpsrc element fail the state change when the
socket is already in use. If we don't prevent reuse, we might end up using the same
port for different streams in some cases.
Fixes #622017
2010-08-04 10:40:23 +02:00
Wim Taymans
e39d7f7359
rtspsrc: improve error and warning message
...
Improve error and warning message.
Fixes #622577
2010-08-04 10:39:44 +02:00
Arnaud Vrac
c6f47c34fb
rtspsrc: add port-range property to rtspsrc
...
To support setups with firewall/ipsec, it is useful for an rtsp client to be
able to set the range of ports that can be used for rtp/rtcp reception.
Allows this by adding a "port-range" property to the rtspsrc element.
Fixes #625153
2010-07-26 17:47:35 +02:00
Wim Taymans
8696d10a5b
rtspsrc: fix memory leak in server request reply
...
The RTSP server rtspsrc is communicating with, sends a GET_PARAMETER request
periodically as a ping. The code in gst_rtspsrc_handle_request forms an OK
response and sends, but doesn't call gst_rtsp_message_unset to free the memory
after sending the response. This results in a constant slow memory leak.
Fixes #624770
2010-07-26 15:33:44 +02:00
Wim Taymans
5534c7d91d
rtspsrc: fix locking after moving things around
2010-06-18 20:04:08 +02:00
Wim Taymans
651c82a01f
rtspsrc: make some errors as warnings
...
Avoid spamming the testsuite with these error debug lines.
2010-06-18 16:56:19 +02:00
Wim Taymans
966ced2208
rtspsrc: factor out the connections
...
Keep a global connection for aggregate control but also keep stream connections
for non-aggregate control.
Add some helper methods to connect/close/flush the connections.
2010-06-18 15:13:06 +02:00
Wim Taymans
ddc214d322
rtspsrc: add non-aggregate control
...
Add non-aggregate control.
Separate retrieving thr SDP from parsing and setting up the streaming from the
SDP.
2010-06-18 15:13:06 +02:00
Wim Taymans
e6ec5cce2e
rtspsrc: respect aggregate control attributes
...
when the SDP specifies an aggregate control url, use that for playback
control.
Fixes #619531
2010-06-14 19:24:14 +02:00
Wim Taymans
cb8252275d
rtsp: try all ranges from the sdp
...
Try all ranges in the SDP before giving up.
2010-06-04 13:58:38 +02:00
Wim Taymans
6fbca707bb
rtspsrc: make parse_range return result
...
Make the parse_range function return if the parsing succeeded or failed.
2010-06-04 13:58:38 +02:00
Wim Taymans
a50cd7c27d
rtspsrc: don't leak the session
2010-05-07 19:02:21 +02:00
Wim Taymans
bc72d8250c
rtsp: configure bandwidth properties in the session
2010-05-07 18:59:42 +02:00
Wim Taymans
db3c4e7f46
rtspsrc: fall back to SDP ports instead of server_port
...
In multicast, fall back to the ports in the SDP instead of the server_port
attribute as this is more in line with the RFC.
2010-05-07 12:51:05 +02:00
Wim Taymans
4e1ced0a77
rtspsrc: refactor collecting the transport info
...
Make a method to collect the ports and destination address.
2010-05-07 12:24:51 +02:00
Wim Taymans
05352d7ea8
rtspsrc: handle servers that send broken Transports
...
Handle servers that send their port pairs with the wrong name.
Fixes #617537
2010-05-07 11:28:36 +02:00
Wim Taymans
ef4d2901aa
rtspsrc: use the SDP connection info in multicast
...
Parse the connection info from the SDP.
When we need to configure the multicast destination, fall back to the SDP
connection info when the transport did not specify a destination and ttl.
Fixes #617537
2010-05-06 16:52:26 +02:00
Wim Taymans
d6579912cb
rtspsrc: make setup url in a smarter way
...
Make sure we always separate the base and control url parts with a / when
creating the setup url.
2010-05-04 16:36:15 +02:00
Alessandro Decina
c8a02a91a6
rtspsrc: handle SEEKING queries.
2010-05-04 16:05:13 +02:00
Wim Taymans
b84bf10455
rtspsrc: add property to control the buffering method
...
Add a property to control how the jitterbuffer performs timestamping and
buffering.
2010-04-05 15:26:03 +02:00
Benjamin Otte
3f511ec361
Add -Wwrite-strings to the configure flags
...
... and fix all warnings
2010-03-21 14:17:47 +01:00
Wim Taymans
ef804589ca
rtsp: use GType from -base and bump required version
...
Use the transport flags GType from -base and bump the required version of -base
because of this.
2010-03-19 15:03:43 +01:00
Benjamin Otte
cccfeaa59c
gst_element_class_set_details => gst_element_class_set_details_simple
2010-03-18 14:32:00 +01:00
Benjamin Otte
1055aaa9cb
Add -Wredundant-decls warning flag
...
Also fix compile issues
2010-03-17 19:35:10 +01:00
Benjamin Otte
3342b1679e
Add -Wmissing-declarations -Wmissing-prototypes warning flags
...
And fix all the warnings.
2010-03-17 18:23:28 +01:00