Commit graph

304 commits

Author SHA1 Message Date
Tim-Philipp Müller
2eb4d1b810 Update for g_type_class_add_private() deprecation in recent GLib 2018-06-24 12:48:11 +02:00
Tim-Philipp Müller
e82ba1e52f Fix indentation 2018-06-24 12:45:49 +02:00
Patricia Muscalu
768fb5695c Get payloader stats only for the sending streams
Get/set payloader properties only for streams that actually
contain a payloader element.

https://bugzilla.gnome.org/show_bug.cgi?id=796523
2018-06-13 10:13:12 +03:00
Mathieu Duponchelle
bfc35ae1ae Implement support for ULP Forward Error Correction
In this initial commit, interface is only exposed for RECORD,
further work will be needed in rtspsrc to support this for
PLAY.

https://bugzilla.gnome.org/show_bug.cgi?id=794911
2018-04-19 18:25:31 +02:00
Sebastian Dröge
ef878da703 gst: Run everything through gst-indent again 2018-04-04 10:06:06 +03:00
Branko Subasic
48ad01beba rtsp-media: query the position on active streams if media is complete
If the media is complete, i.e. one or more streams have been configured
with sinks, then we want to query the position on those streams only.
A query on an incomplete stream may return a position that originates from
an earlier preroll.

https://bugzilla.gnome.org/show_bug.cgi?id=794964
2018-04-04 10:05:38 +03:00
Mathieu Duponchelle
988db52016 rtsp-server: add API to enable retransmission requests
"do-retransmission" was previously set when rtx-time != 0,
which made no sense as do-retransmission is used to enable
the sending of retransmission requests, where as rtx-time
is used by the peer to enable storing of buffers in order
to respond to retransmission requests.

rtsp-media now also provides a callback for the
request-aux-receiver signal.

https://bugzilla.gnome.org/show_bug.cgi?id=794822
2018-03-30 17:55:32 +02:00
Carlos Rafael Giani
5f29712243 rtsp-media: Replace g_print() log line
https://bugzilla.gnome.org/show_bug.cgi?id=793838
2018-02-26 15:26:29 +02:00
Mathieu Duponchelle
ddb0d83844 rtsp-media: fix RECORD getting stuck
The test_record case was working because async=false had
been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
but that was incorrect, as it should not be needed.

Removing async=false made the test fail as expected, this is
fixed by not trying to preroll when preparing the media for
RECORD, as start_prepare is called upon receiving ANNOUNCE,
and our peer will not start sending media until it has received
a response to that request, and sent and received a response
to RECORD as well, thus obviously preventing preroll.

https://bugzilla.gnome.org/show_bug.cgi?id=793738
2018-02-23 16:13:56 +01:00
Ognyan Tonchev
14c511ae62 stream: Add functions for checking if stream is receiver or sender
...and replace all checks for RECORD in GstRTSPMedia which are really
for "sender-only". This way the code becomes more generic and introducing
support for onvif-backchannel later on will require no changes in
GstRTSPMedia.
2018-02-16 11:04:53 +02:00
Sebastian Dröge
231700b2bb rtsp-media: Add support for sending+receiving medias
We need to add an appsrc/appsink in that case because otherwise the
media bin will be a sink and a source for rtpbin, causing a pipeline
loop.

https://bugzilla.gnome.org/show_bug.cgi?id=788950
2018-02-16 11:04:53 +02:00
Mathieu Duponchelle
ee44f38051 set_clock functions: Add nullable annotations 2018-02-13 18:59:49 +01:00
Mathieu Duponchelle
c725ef01a4 All around: add annotations and API guards 2018-02-12 19:16:11 +01:00
Edward Hervey
64a46d47ba rtsp-server: Minor doc fixes
Mostly for g-i
2017-12-07 16:08:50 +01:00
Patricia Muscalu
0015791f8f rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-27 07:58:42 +01:00
Edward Hervey
9514f2d354 rtsp-media: Enable seeking query before pipeline is complete
SDP are now provided *before* the pipeline is fully complete. In order
to know whether a media is seekable or not therefore requires asking
the invididual streams.

API: gst_rtsp_stream_seekable

https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-25 07:53:11 +01:00
Patricia Muscalu
bb29d2e2ee rtsp-media: Fix handling in default_unsuspend()
Handle the case when streams are not blocked and media
is suspended from PAUSED.

Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040

https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-24 10:52:36 +01:00
Patricia Muscalu
132e00adfd rtsp-media: Removed fakesink elements
There is not need of adding fakesink elements to the media
pipeline in the dynamic-payloader case.
The media pipeline itself is dynamically updated with
the receiver and sender parts that are based on the client
transport information known after SETUP has been received.

Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9

https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-24 10:52:36 +01:00
Patricia Muscalu
ac6169d50a rtsp-media: Corrected ASYNC_DONE handling
Media is complete when all the transport based parts are
added to the media pipeline. At this point ASYNC_DONE is
posted by the media pipeline and media is ready to enter
the PREPARED state.

Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa

https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-24 10:52:36 +01:00
Edward Hervey
6371f2fc29 rtsp-media: Don't unblock with remaining dynamic payloaders
If we still have some dynamic paylaoders which haven't posted
no-more-pads yet, don't go to PREPARED if one of the streams
blocked.

The risk was that we would end up not exposing/using all specified
streams.

The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
then it will take a bit more time to start. But only if those 3
conditions are present.

https://bugzilla.gnome.org/show_bug.cgi?id=769521
2017-11-21 07:59:15 +01:00
Edward Hervey
d1a6418fe2 rtsp-media: Fix doc 2017-11-21 07:59:15 +01:00
Edward Hervey
0dddaba9bb rtsp-media: Don't set float on a gint64 variable
Just use 0. Fixes 'undefined' behaviour from clang
2017-11-21 07:59:15 +01:00
Edward Hervey
27d256d4ca rtsp-media: Fix previous commit
We only want to count dynamic payloaders
2017-11-21 07:59:15 +01:00
Edward Hervey
2386e91c36 rtsp-media: Handle multiple dynamic elements
If we have more than one dynamic payloader in the pipeline, we need
to wait until the *last* one emits 'no-more-pads' before switching
to PREPARED.

Failure to do so would result in a race where some of the streams
wouldn't properly be prepared

https://bugzilla.gnome.org/show_bug.cgi?id=769521
2017-11-20 09:38:49 +01:00
Patricia Muscalu
efdb795c86 rtsp-media: seek on media pipelines that are complete
Make sure that a seek is performed on pipelines that
contain at least one sink element.

Change-Id: Icf398e10add3191d104b1289de612412da326819

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 19:56:26 +02:00
Patricia Muscalu
a7732a68e8 Dynamically reconfigure pipeline in PLAY based on transports
The initial pipeline does not contain specific transport
elements. The receiver and the sender parts are added
after PLAY.
If the media is shared, the streams are dynamically
reconfigured after each PLAY.

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 19:56:15 +02:00
Patricia Muscalu
b5c3ef8d53 rtsp-media: return minimum value in query position case
The minimum position should be returned as we are interested
in the whole interval.

Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +02:00
Thibault Saunier
8608c1cae4 rtsp-media: Initialize scalar variable
CID 1418985
2017-10-09 14:44:40 +02:00
Thibault Saunier
9706199efb Start support for RTSP 2.0
This adds basic support for new 2.0 features, though the protocol is
subposdely backward incompatible, most semantics are the sames.

This commit adds:

- features:
 * version negotiation
 * pipelined requests support
 * Media-Properties support
 * Accept-Ranges support

- APIs:
  * gst_rtsp_media_seekable

The RTSP methods that have been removed when using 2.0 now return
BAD_REQUEST.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-10-05 13:23:48 -03:00
Thibault Saunier
b56930704f gi: Fix some annotations and docstrings 2017-04-13 14:20:10 -03:00
Edward Hervey
dea000f2e3 media: Fix pt map caps
Since decryption is handled within rtpbin, all outcoming stream
caps will be application/x-rtp (i.e. regular rtp)

Fixes RECORD with SRTP streams
2016-12-02 15:47:12 +01:00
Kseniia Vasilchuk
09e499387d media: Fix race condition around finish_unprepare() if called multiple time
https://bugzilla.gnome.org/show_bug.cgi?id=755329
2016-12-01 16:39:00 +02:00
Neha Arora
166a903594 rtsp-media: Only signal "new-state" if the state has actually changed
https://bugzilla.gnome.org/show_bug.cgi?id=774173
2016-11-10 13:16:23 +02:00
Ian Jamison
34389831cb rtsp-server: Hint that set_multicast_iface expects the name of the interface
To prevent any possibly confusion with IPs or anything else.

https://bugzilla.gnome.org/show_bug.cgi?id=771530
2016-09-18 10:00:29 -04:00
Sebastian Dröge
800bed8c9c rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
2016-09-18 09:58:55 -04:00
Sebastian Dröge
9fab555cc5 rtsp-server: Implement clock signalling according to RFC7273
For NTP and PTP clocks we signal the actual clock that is used and signal
the direct media clock offset.

For all other clocks we at least signal that it's the local sender clock.

This allows receivers to know which clock was used to generate the media and
its RTP timestamps. Receivers can then implement network synchronization,
either absolute or at least relative by getting the sender clock rate directly
via NTP/PTP instead of estimating it from RTP timestamps and packet receive
times.

https://bugzilla.gnome.org/show_bug.cgi?id=760005
2016-04-03 11:22:31 +03:00
Sebastian Dröge
69d04f3838 rtsp-media: Add support for setting the multicast interface
https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-25 12:52:12 +02:00
Steven Hoving
aea624b6f8 rtsp-media: fix state_lock not locked again when preroll fails
https://bugzilla.gnome.org/show_bug.cgi?id=761399
2016-02-02 10:36:05 +00:00
Steven Hoving
fefc011dfb rtsp-media: Fix mutex beeing unlocked while they should be locked
https://bugzilla.gnome.org/show_bug.cgi?id=761226
2016-01-28 09:34:32 +01:00
Hyunjun Ko
924f914172 sdp: replace duplicated codes to call new base sdp apis
https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-31 17:13:39 +02:00
Sebastian Dröge
7a41d396ae rtsp-media: Add API to directly configure a clock on the media pipelines 2015-12-30 18:40:43 +02:00
Sebastian Dröge
cbf3f3888f rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency() 2015-12-30 16:43:17 +02:00
Sebastian Rasmussen
b2abb97043 rtsp-media: Do not prepare media after media times out
Deferred calls to start_prepare() can be deferred past the point until
which wait_preroll() and by proxy gst_rtsp_media_get_status() is
prepared to wait. Previously there was no lock and no check for this
situation. This meant that a media could be prepared and unprepared
simultaneously by two different threads. Now a lock is in place and a
suitable check is done.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
2015-12-28 14:08:09 +02:00
Sebastian Dröge
c8f179948e rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
Without TEARDOWN it might be desireable to keep the media running and continue
sending data to the client, even if the RTSP connection itself is
disconnected.

Only do this for session medias that have only UDP transports. If there's at
least on TCP transport, it will stop working and cause problems when the
connection is disconnected.

https://bugzilla.gnome.org/show_bug.cgi?id=758999
2015-12-28 10:51:56 +02:00
Sebastian Rasmussen
6f1cad9237 rtsp-media: Take reference to media that will be prepared
default_prepare() takes a transfer-none reference GstRTSPMedia object.
Later on a g_idle_source_new() is created and a pointer to the media
object is passed as user data. If the media is freed before the idle
source is dispatched the media object pointer is invalid, but the idle
source callback expects it to still be valid. To fix this a reference to
the media object is taken when registering the source callback function
and a corresponding release of the reference is done when the souce is
destroyed.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
2015-09-29 11:23:06 +01:00
Jan Schmidt
315c2f93bb rtsp-media: Don't crash on encrypted RTX SDP
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).

https://bugzilla.gnome.org/show_bug.cgi?id=754753
2015-09-09 17:57:15 +10:00
Jan Schmidt
27736d406e rtsp-stream: Implement UDP buffer size setting.
Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
UDP TX buffer size.

Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-09-03 22:19:40 +10:00
Nicolas Dufresne
707ac9c487 media: Only add fakesink once per pipeline
The intention is to prevent going PLAYING state before pads are created.
If there was mutilple dynamic payload, it would leak few fakesink and
actually prevent from ever reaching playing state.

https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-08-08 09:46:40 -04:00
Nicolas Dufresne
160b87430f Revert "rtsp-media: Only add 1 fakesink per pipeline"
This reverts commit 22bf61f16c.
2015-08-08 09:08:37 -04:00
Nicolas Dufresne
22bf61f16c rtsp-media: Only add 1 fakesink per pipeline
There should be only one fakesink per pipeline, not per dynpay. This
would lead to element naming clash.
2015-08-07 09:33:55 -04:00