Linus Svensson
398296d978
rtspsrc: Don't include payload type in the caps for framesize
...
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2015-03-09 10:18:35 +01:00
Sebastian Dröge
c34a7cb90d
rtspsrc: Fix handling of interleaved (TCP) streams
...
We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.
https://bugzilla.gnome.org/show_bug.cgi?id=745599
2015-03-05 12:15:04 +01:00
Sebastian Dröge
b4aaa11f97
rtspsrc: Don't unref caps we don't own
2015-03-05 09:56:37 +01:00
Sebastian Dröge
297d808acc
rtspsrc: Push RTCP caps on the RTCP pads
...
Otherwise we will get not-negotiated later from rtpbin, and will never be able
to send RTCP packets back to the server. Note that error flow returns from the
RTCP pads are ignored, that's why it didn't fail more visible before.
2015-03-05 09:47:29 +01:00
Sebastian Dröge
788074733c
rtspsrc: Make sure to send SEGMENT events on all pads
2015-03-05 09:47:29 +01:00
Krzysztof Kotlenga
e3ca4d1c86
rtspsrc: improve error message when unauthorized
...
Make use of NOT_AUTHORIZED error code instead of falling back to generic
READ error.
https://bugzilla.gnome.org/show_bug.cgi?id=601733
2015-02-24 11:08:27 +02:00
Sebastian Dröge
f4b5107796
Improve and fix LATENCY query handling
...
This now follows the design docs everywhere, especially the maximum latency
handling.
https://bugzilla.gnome.org/show_bug.cgi?id=744106
2015-02-11 13:53:02 +01:00
Tim-Philipp Müller
603c1d71a1
rtspsrc: fix awkward if clause
2015-02-08 12:03:10 +00:00
Wim Taymans
852c040c89
rtspsrc: fix container handling
...
We detect a container correctly now so we need to revert the weird
check there was before.
Use gst_rtspsrc_stream_push_event() to push the caps event on the
right pad.
See https://bugzilla.gnome.org/show_bug.cgi?id=739391
2015-02-03 17:39:10 +01:00
Sebastian Dröge
d5aab81a77
Constify some static arrays everywhere
2015-01-21 09:55:53 +01:00
Aleix Conchillo Flaqué
07c5d1820a
rtspsrc: set PLAYING state after configuring caps
...
We set to PLAYING after we have configured the caps, otherwise we
might end up calling request_key (with SRTP) while caps are still
being configured, ending in a crash.
https://bugzilla.gnome.org/show_bug.cgi?id=740505
2014-12-31 12:49:11 +00:00
Sebastian Dröge
d18b893d28
rtspsrc: Don't dereference NULL if a suitable stream for the AUX element can't be found
...
CID 1258717
2014-12-18 11:51:12 +01:00
Sebastian Dröge
6b2fc2de8d
rtspsrc: Add something to the debug logs if an RTX AUX element can't be added
...
... because the application already has a signal handler set up here.
2014-12-16 16:40:08 +01:00
Matthew Waters
bf0a19bf02
rtspsrc: add retransmission support according to RFC4588
...
Based on the client-rtpaux example
2014-12-16 16:40:08 +01:00
Aleix Conchillo Flaqué
00ca83629b
rtspsrc: fix leak for mikey base64 decoded key-mgmt
...
https://bugzilla.gnome.org/show_bug.cgi?id=740392
2014-11-20 09:15:56 +01:00
Josep Torra
038cc7b004
rtsp: fix build in gst-uninstalled setup
2014-11-06 21:38:43 +01:00
Aleix Conchillo Flaqué
d15ebcbf62
rtspsrc: mikey related memory leaks
...
https://bugzilla.gnome.org/show_bug.cgi?id=739430
2014-10-31 10:03:47 +00:00
Aleix Conchillo Flaqué
bd392d72ee
rtspsrc: set full stream caps on internal src TCP pads
...
Set the complete stream caps on the TCP internal src pads. Otherwise,
ptdemux will not properly detect the caps change.
https://bugzilla.gnome.org/show_bug.cgi?id=737868
2014-10-21 11:33:01 +02:00
Sebastian Dröge
d1c7f2e4d1
rtspsrc: Fix compiler warnings
...
gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
'GstRTSPResult' [-Werror,-Wenum-conversion]
res = gst_sdp_message_new (&sdp);
~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
'GstRTSPResult' [-Werror,-Wenum-conversion]
res = gst_sdp_message_parse_uri (uri, sdp);
~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
2014-09-26 13:46:16 +03:00
Sanjay NM
f62076e49c
rtsp: clarify expression so operator precedence is clear
...
https://bugzilla.gnome.org/show_bug.cgi?id=736903
2014-09-24 00:48:09 +01:00
Sebastian Dröge
add40de469
rtspsrc: Push the correct segment in TCP mode when seeking
2014-08-05 16:28:04 +02:00
Tim-Philipp Müller
c2614e5253
rtspsrc: fix query leak
...
https://bugzilla.gnome.org/show_bug.cgi?id=733003
2014-07-10 17:19:42 +01:00
Wim Taymans
db1d9444d6
rtspsrc: fix for mikey api change
2014-07-02 16:01:47 +02:00
Wim Taymans
98a4ee0f92
rtspsrc: pass the stream id when asking for crypto params
...
This way the app can choose different parameters for each stream.
2014-06-19 16:17:23 +02:00
Aleix Conchillo Flaqué
7ce0ea3946
rtspsrc: add support for key length parameters
...
This patch adds supports for the incoming key management parameters for
encryption and authentication key lengths.
It also adds a new signal request-rtcp-key that allows the user to
provide the crypto parameters and key for the RTCP stream.
https://bugzilla.gnome.org/show_bug.cgi?id=730473
2014-06-19 16:11:19 +02:00
Aleix Conchillo Flaqué
782d65cab1
rtspsrc: always use a random ssrc for the internal session
...
Use a random SSRC different than 0 for the internal session SSRC.
https://bugzilla.gnome.org/show_bug.cgi?id=730212
2014-05-16 16:58:44 +02:00
Vincent Penquerc'h
a846e84349
rtspsrc: remove duplicate test
...
item was dereference previously.
While there, reorder some test for faster early out.
Coverity 1139844
2014-05-02 14:06:25 +01:00
Tim-Philipp Müller
c9597298f9
docs: remove outdated and pointless 'Last reviewed' lines from docs
...
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
2014-04-26 23:35:17 +01:00
Wim Taymans
5b9945e0a6
rtspsrc: update for new MIKEY API
2014-04-04 17:38:14 +02:00
Wim Taymans
6210cbe1e2
rtspsrc: send sender SSRC in the MIKEY message
...
Allocate a new SSRC for our RTCP messages back to the server and set
this in the MIKEY message.
2014-04-03 17:40:01 +02:00
Wim Taymans
4f641ef18b
rtspsrc: make random number for the CSB
...
As recommended in the RFC
2014-04-03 17:39:30 +02:00
Wim Taymans
f932da3be6
rtspsrc: don't put spaces in keymgmt header
2014-04-03 12:21:27 +02:00
Wim Taymans
2edd450369
rtspsrc: create and send the RTCP encryption key
...
Create and make a key for encrypting the RTCP packets back to the server
and wrap this in a MIKEY message that we send as a header in the SETUP
request.
2014-04-03 12:21:27 +02:00
Wim Taymans
a52b7eadfd
rtspsrc: free the srtpdec element
2014-04-03 12:18:39 +02:00
Wim Taymans
f0f9451523
rtspsrc: cleanup stream_free function
...
There is no reason to NULL all fields, we will free the stream anyway.
2014-04-03 12:16:25 +02:00
Wim Taymans
e7c8fa1127
rtspsrc: only expose streams on dataflow
...
Only probe on buffers, we don't want to expose the streams on events.
2014-03-25 11:44:27 +01:00
Wim Taymans
3b497bf7d5
rtspsrc: copy sticky events to ghostpad
...
When we expose internal pads as ghostpads, first copy the sticky events
so that we have the caps and segment etc.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724712
2014-03-25 11:36:40 +01:00
Wim Taymans
67f3113759
rtspsrc: srtp handling
2014-03-25 10:23:24 +01:00
Wim Taymans
4846be1491
rtspsrc: set SSRC on caps if known
2014-03-25 10:23:00 +01:00
Wim Taymans
5ec8c96966
rtspsrc: put caps on udpsrc instead of using the signals
...
Try to avoid using the request-pt-map to get caps but set them directly
on the udpsrc element. That way, the caps get nicely transformed as they
pass through the different elements in the rtpbin, including the AUX and
decoder/encoder elements.
2014-03-24 17:07:06 +01:00
Wim Taymans
2b59828e0b
rtspsrc: use profile to set rtcp caps
...
Use the negotiated profile to set x-rtcp or x-srtcp caps
2014-03-24 15:35:09 +01:00
Wim Taymans
a7b55d7687
rtspsrc: set udpsrc to READY
...
READY is enough to allocate ports now
2014-03-24 15:34:26 +01:00
Wim Taymans
a4f6f963ec
rtspsrc: free caps in ptmap array
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726696
2014-03-24 14:35:01 +01:00
Wim Taymans
d6c5fbc87c
rtspsrc: handle NULL rtpmap and parse error better
2014-03-20 11:12:51 +01:00
Wim Taymans
224239096d
rtspsrc: skip streams with same control url
...
Keep track of what streams we did the SETUP for. We only need to
configure caps, wait for pads and push events on setup streams. We can
remove the disabled state of the stream and simplify some checks.
After we setup a stream, skip the other streams that have the same
control url. Use a skipped flag to mark streams that should be skipped.
2014-03-06 12:30:54 +01:00
Wim Taymans
3b27fc2f0f
rtspsrc: remove obsolete code
2014-03-06 12:30:54 +01:00
Wim Taymans
27d883fe64
rtspsrc: just use the SDP index as the stream id
...
Use the index of the media stream in the SDP as the stream id instead of
keeping a separate counter.
2014-03-06 12:30:54 +01:00
Wim Taymans
99a9d2873c
rtspsrc: handle NULL control urls better
2014-03-05 15:44:25 +01:00
Alessandro Decina
c4bf6e8b7e
rtspsrc: fix seeking
...
Call gst_rtspsrc_connection_flush (src, FALSE) to reset connections as
non-flushing before sending PAUSE and PLAY with the new npt range. Without this
patch, those commands would fail with EINTR as the connections were still
flushing.
2014-03-05 11:39:09 +01:00
Wim Taymans
70de0e4e99
rtspsrc: Add support for multiple payload types
...
A media stream can have multiple payload types. Parse all the payload
types and collect the caps information. We then have to store the
pt<->caps mapping instead of 1 pt and 1 caps.
Parse the profile from the SDP and use that to negotiate the transport
instead of always using AVP.
Rework how we do some tweaks for ASF and Realmedia.
2014-03-04 16:40:34 +01:00
Wim Taymans
dbe92c9147
rtspsrc: refactor payload handling
2014-03-04 11:34:39 +01:00
Santiago Carot-Nemesio
b9a953161f
rtspsrc: Fix deadlock when task creation is no successful
...
https://bugzilla.gnome.org/show_bug.cgi?id=725124
2014-02-25 10:10:31 +01:00
Aleix Conchillo Flaqué
62f5a27416
rtspsrc: add tls-database property
...
Add support for a new property: tls-database. If the property is set,
the certificate database will be given to the rtsp connection if TLS
protocol is being used. If the server certificate can't be verified with
the default database, this additional database will be used.
https://bugzilla.gnome.org/show_bug.cgi?id=724396
2014-02-20 20:03:40 +01:00
Sebastian Dröge
8054cd5df3
Revert "rtspsrc: Proxy rtpjitterbuffer do-retransmission property"
...
This reverts commit 9f7b1128b1
.
This should be handled automatically be rtspsrc if the AVPF profile
is used, and manual enabling of it can be done with the new-manager
signal.
2014-01-24 12:37:39 +01:00
Wim Taymans
43feb82feb
rtspsrc: add signal to notify of new manager
...
So that you can configure and connect to signals on the rtpbin.
See https://bugzilla.gnome.org/show_bug.cgi?id=722866
2014-01-24 10:22:59 +01:00
Aleix Conchillo Flaqué
9f7b1128b1
rtspsrc: Proxy rtpjitterbuffer do-retransmission property
...
https://bugzilla.gnome.org/show_bug.cgi?id=722866
2014-01-24 09:14:59 +01:00
Wim Taymans
2e9e80badf
rtspsrc: use new method to get media-type
...
Use the new method to get the media type of a transport.
2014-01-07 15:04:02 +01:00
Wim Taymans
bf878d75d1
rtspsrc: use aggregate control for PLAY/PAUSE/TEARDOWN
...
Use the aggregate control instead of the original request url to perform
PAUSE/PLAY and TEARDOWN.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721003
2013-12-26 11:27:30 +01:00
Tim-Philipp Müller
28f524a551
rtspsrc: fix 'make check'
...
Fix generic/states check. Also, g_return_if_fail() is
not for internal state checking.
2013-11-18 17:13:49 +00:00
Tim-Philipp Müller
d506409af5
docs: get rid of 'Since: 0.10.x' markers
...
And some gtk-doc markup fixes.
2013-11-18 14:47:35 +00:00
Sebastian Dröge
9ae6981578
rtspsrc: Use the synced buffer mode in auto mode if a clock provider is in the SDP
2013-11-13 10:54:19 +01:00
Rico Tzschichholz
b137f79581
rtsp: Add missing gio-2.0 deps and includes
2013-11-02 23:12:13 +01:00
Aleix Conchillo Flaque
82b8374af8
rtspsrc: allow setting tls certificate validation flags
...
Added a new property "tls-validation-flags". If the url transport is
TLS, the validation flags will be set to the rtsp connection.
https://bugzilla.gnome.org/show_bug.cgi?id=711230
2013-11-01 16:47:36 +01:00
Wim Taymans
e96f8f519c
rtspsrc: proxy new buffer mode
2013-10-31 10:38:35 +01:00
Wim Taymans
8c5ce0dbdc
rtspsrc: also go into the loop function after connect
...
When we have opened the stream, go into the loop function so that we can
receive messages from the server.
2013-09-27 15:08:31 +02:00
Wim Taymans
6095e2e859
rtspsrc: disable checks when linking pads
...
We know the pad links will work (and we don't check the return value
anyway).
2013-09-25 17:42:02 +02:00
Wim Taymans
9f9bcbc405
rtspsrc: only wait if we flushed
...
Only wait for the STREAM_LOCK when we flushed something when sending
a command for PAUSED or PLAYING.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611
2013-09-09 15:13:46 +02:00
Wim Taymans
7b2e002879
rtspsrc: return when a flush was issued
...
Make gst_rtspsrc_loop_send_cmd() return TRUE when the current
action has been flushed
2013-09-09 15:13:46 +02:00
Tim-Philipp Müller
1dfc1f2686
Don't use setlocale in plugins()
...
Only apps should call setlocale(), not libraries.
2013-09-01 21:18:38 +01:00
Youness Alaoui
e22f7e91c4
rtspsrc: Fix response argument in handle-request signal
2013-08-21 09:06:02 +02:00
Youness Alaoui
6636efd31a
rtspsrc: Add sdes property and proxy it to rtpbin
2013-08-21 09:06:02 +02:00
Sebastian Dröge
282afae244
rtspsrc: Only free GCheckSum after its last usage
...
https://bugzilla.gnome.org/show_bug.cgi?id=705760
2013-08-13 12:44:11 +02:00
Tim-Philipp Müller
7272dec5fe
rtpdec: use generic marshaller
2013-08-04 11:20:41 +01:00
Sebastian Dröge
169b490664
rtspsrc: Add support for group-id in the stream-start event
2013-07-22 15:30:13 +02:00
Wim Taymans
ab24598443
rtspsrc: avoid some strdup
2013-07-02 11:13:25 +02:00
Wim Taymans
7c950ef3f2
rtspsrc: add select-stream signal
...
Add a signal to let the app select what streams will be selected.
See https://bugzilla.gnome.org/show_bug.cgi?id=634419
2013-07-02 10:40:35 +02:00
Wim Taymans
2d276e1bcb
rtspsrc: avoid strdup
2013-07-02 10:40:35 +02:00
Wim Taymans
1db7e62060
rtspsrc: add signal to notify of the SDP
...
This way, the app can look and modify the SDP.
2013-07-01 17:31:30 +02:00
Wim Taymans
3289a2963b
rtspsrc: reset-sync before play
...
Call reset-sync on the rtpbin before we go to playing. This makes us require SR
packets for all streams again before we attempt to sync them. If we don't reset,
it might be that we combine SR packets from before and after the PAUSE/PLAYING
state change and end up with huge bogus offsets.
2013-06-27 17:02:14 +02:00
Wim Taymans
bb9d42b976
rtspsrc: avoid some flushes
2013-06-26 14:58:53 +02:00
Wim Taymans
f39ef2ab68
rtspsrc: handle data message when waiting for reply
...
When we are waiting for a server reply, handle data messages instead of
ignoring them.
2013-06-26 14:41:36 +02:00
Wim Taymans
61219dc6ed
rtspsrc: handle data messages in separate method
...
Refactor and make a method to handle a data message.
2013-06-26 14:41:36 +02:00
Wim Taymans
a4be0c6de3
rtspsrc: add some more docs to handle-request signal
...
See https://bugzilla.gnome.org/show_bug.cgi?id=702705
2013-06-25 20:36:18 +02:00
Youness Alaoui
52e440c91b
Send a clock_provide message on the bus when we get a netclock
2013-06-25 14:50:47 +02:00
Youness Alaoui
547df8e14f
rtspsrc: Expose use-pipeline-clock property
2013-06-25 14:50:33 +02:00
Youness Alaoui
95906b8f1c
rtsp: go back into the loop after doing pause
...
After we do a pause request, go back to loop mode so that we can listen
for server messages again.
See https://bugzilla.gnome.org/show_bug.cgi?id=702705
2013-06-21 10:42:20 +02:00
Wim Taymans
b96d931bf4
rtspsrc: fix race in state change to paused
...
When we go to paused, we first flush the connection and then send the pause
command. As a result of the flushing, the scheduled paused command can get
lost. Wait until the connection is completely flushed and the rtsp task is
waiting before issuing the paused or playing request.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702705
2013-06-20 14:43:47 +02:00
Wim Taymans
d9bc48edc9
rtspsrc: manage element state ourselves
...
Lock the state of the all our elements and manage their states
outselves. Because we are working async, we can't rely on the state
change function to set the state at the right time or to return the
right return value from the state change function.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702046
2013-06-16 05:40:13 +02:00
Wim Taymans
25082a50b9
rtspsrc: add extra TLS url protocols
...
We also support TLS protocols now.
2013-05-31 12:34:22 +02:00
Wim Taymans
80850df711
rtspsrc: create and push stream-start in TCP mode
2013-05-28 15:45:49 +02:00
Wim Taymans
4fc1f3088b
rtspsrc: remove some obsolete code
...
It is not needed to do a state change from the _play() function on
ourselves. The state change function already did that and we don't want to
interfere with that (or use hacks to avoid interference).
2013-05-28 15:10:07 +02:00
Wim Taymans
e6f850996b
rtspsrc: set RTCP caps on the RTCP pads
2013-05-28 12:26:25 +02:00
Wim Taymans
779bcc093c
rtspsrc: add signal to handle server requests
...
Add a signal to be notified of a server request. The signal handler can then
construct the response message for the server.
See https://bugzilla.gnome.org/show_bug.cgi?id=632207
2013-05-28 12:26:24 +02:00
Tim-Philipp Müller
643450c9b8
Revert "gstrtspsrc: set buffer-size for multicast buffers"
...
This reverts commit 2481e95d03
.
This is already done five lines above, it was added a year
ago in commit 561b131e
.
2013-05-09 09:09:59 +01:00
Aha Unsworth
2481e95d03
gstrtspsrc: set buffer-size for multicast buffers
...
For receiving video data via RTSP when the video is sent via
multicast there is no way to specify the udpsrc buffer-size.
On windows the native network buffer is not large and with video
i-frames being huge the buffer is to small and you get i-frame corruption,
it looks terrible, and there is no (easy) way to set the udpsrc buffer-size.
https://bugs.freedesktop.org/show_bug.cgi?id=52264
2013-05-08 16:57:53 -03:00
Sebastian Dröge
b0b0557c48
gst: Add better support for static plugins
2013-04-15 15:54:11 +02:00
Sebastian Dröge
b17750ed9e
rtspsrc: Proxy the ntp-sync property of rtpbin
2013-04-12 12:58:50 +02:00
Sebastian Dröge
53dae1585e
rtspsrc: Give the manager always the name "manager"
...
This allows to use the GstChildProxy interface to adjust
properties on it.
2013-04-12 12:51:05 +02:00
Wim Taymans
f8013487c9
rtspsrc: add support for NetClientClock
...
When the server suggests a GstNetTimeProvider in the SDP, set up a
GstNetClientClock that slaves to the remote clock and suggest this clock in
provide_clock.
2013-04-11 15:00:05 +01:00
Sebastian Dröge
d80ff8e7f3
rtspsrc: Proxy the multicast-iface property of udpsrc
2013-04-03 17:53:13 +02:00
Wim Taymans
640de61740
rtspsrc: only EOS when our source sends BYE
...
Only EOS when we receive a BYE event from the SSRC of our stream.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453
2013-02-06 14:01:16 +01:00
Wim Taymans
0540492ab2
rtspsrc: save the stream SSRC
...
Conflicts:
gst/rtsp/gstrtspsrc.c
2013-02-06 14:00:56 +01:00
Wim Taymans
c8fb1c720c
rtspsrc: flush connection when stopping
...
When we stop, we can flush all pending commands so that we can stop and
join the task.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924
2013-02-06 13:18:18 +01:00
Tim-Philipp Müller
95a37196b3
rtspsrc: add "proxy-id" and "proxy-pw" properties
...
to match souphttpsrc. user/password passed via the URI
will still take precedence though.
https://bugzilla.gnome.org/show_bug.cgi?id=395427
2012-12-31 00:22:27 +00:00
Wim Taymans
8cfec6a88d
rtspsrc: fix cmd comparison
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690476
2012-12-20 17:12:30 +01:00
Wim Taymans
75616fac9a
rtspsrc: add some more debug
2012-12-20 17:12:20 +01:00
Wim Taymans
a858bf46db
rtspsrc: fix TCP reconnect
...
Ignore other commands when reconnecting, otherwise the loop function would pause
and the reconnection would not happen. Continue looping after doing a reconnect
so that we have a chance to actually read the new data.
2012-12-13 09:30:59 +01:00
Wim Taymans
b1dc816772
rtspsrc: timeout on udpsrc is in nanoseconds
2012-12-12 11:09:42 +01:00
Aleix Conchillo Flaque
3503aef946
rtspsrc: do not change state to PLAYING if currently chaning state
...
* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be
happening in the application thread, so we don't change the state to
PLAYING in the gstrtspsrc thread unless it is safe.
A specific case is when chaning the state to NULL from the application
thread. This will synchronously try to stop the task (with the element
state lock acquired), but we will try a gst_element_set_state from
gstrtspsrc thread which will block on the element state lock causing a
deadlock.
https://bugzilla.gnome.org/show_bug.cgi?id=684312
2012-12-10 15:13:22 +01:00
Wim Taymans
64cdbb77a9
rtspsrc: use new option parser function
2012-11-27 11:13:37 +01:00
Wim Taymans
5d0507c09e
rtspsrc: pause the task instead of spinning
...
Actually pause the loop task instead of spinning forever.
2012-11-22 11:34:31 +01:00
Wim Taymans
c28bfa8902
rtspsrc: handle segment event
...
Make a segment event when we send a new range header to a client (first PLAY
request or after a seek). Send the segment event in interleaved mode.
Clean the segment event on cleanup
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382
2012-11-16 15:38:29 +01:00
Wim Taymans
bd91bd3193
rtspsrc: fix check for active streams
...
A stream can be active without a srcpad yet and we want to send
events on those streams as well.
2012-11-16 15:22:46 +01:00
Wim Taymans
11cf4d4fd3
rtspsrc: create and add pads outside of lock
...
Create and add the ghostpad for the new stream outside of the lock because it
is not needed and causes deadlocks.
2012-11-16 13:33:44 +01:00
Aleix Conchillo Flaque
6c855edf03
rtspsrc: allow client to disable reconnection
...
* gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before,
rtspsrc always tried to reconnect to the server when the RTSP
connection was closed by the server. This property lets the user
decide whether it wants rtspsrc to reconnect or not.
https://bugzilla.gnome.org/show_bug.cgi?id=683912
2012-11-16 12:55:10 +01:00
Wim Taymans
e2a4d28c1f
rtspsrc: clear variables before retrying
...
Else we might unref an old udpsrc twice in cleanup.
2012-11-16 12:17:37 +01:00
Wim Taymans
cc9cb26be1
rtspsrc: propose ports in multicast
...
When the user configured a port-range, propose ports from this range
as the multicast ports. The server is free to ignore this request but if it
honours it, increment our ports so that we suggest the next port pair for the
next stream.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420
2012-11-16 12:17:37 +01:00
Wim Taymans
5025b3f1b3
rtspsrc: add more debug
2012-11-16 12:17:37 +01:00
Marc Leeman
7cbca3dcd1
rtsp: the RTCP port number is inclusive
...
The configured port number pair has its upper bound set to the maximum
allowed RTCP port, inclusive.
See https://bugzilla.gnome.org/show_bug.cgi?id=639420
2012-11-06 13:22:58 +01:00
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
Wim Taymans
51371d26ee
update for RTP buffer api changes
2012-07-17 16:38:27 +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
Sebastian Dröge
a1948e34d2
elements: Use gst_pad_set_caps() instead of manual event fiddling
2012-06-08 15:54:42 +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
Sebastian Dröge
d99eb6d2cb
Update everything for the removal of the interface library and mixer/tuner interfaces
2012-04-13 13:15:11 +02:00
Tim-Philipp Müller
e09ae5736d
Use new gst_element_class_set_static_metadata()
2012-04-10 00:51:41 +01:00
Sebastian Dröge
aa2cd462da
gst: Update for GST_PLUGIN_DEFINE() API changes
2012-04-05 17:36:38 +02:00
Sebastian Dröge
5cdd49bf25
gst: Update versioning
2012-04-04 14:37:47 +02:00
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
27ee8931dd
autodetect, rtsp: gst_registry_get_default() -> gst_registry_get()
2012-01-02 14:32:40 +00:00
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
71b615515a
update for clock provider API change
2011-11-28 17:52:06 +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