Commit graph

64 commits

Author SHA1 Message Date
Mathieu Duponchelle
0f498eabf4 onvif: Implement and test the Streaming Specification
https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
2019-06-06 18:45:17 +02:00
Branko Subasic
421ac85150 rtsp-media: allow specifying rate when seeking
Add new function gst_rtsp_media_seek_full_with_rate() which allows the
caller to specify the rate for the seek. Also added functions in
rtsp-stream and rtsp-media for retreiving current rate and applied rate.

https://bugzilla.gnome.org/show_bug.cgi?id=754575
2019-06-04 14:32:51 +02:00
Göran Jönsson
afb27f91cf rtsp-server: remove recursive behavior
Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2019-02-02 10:42:33 +00:00
Göran Jönsson
7cfd59820a rtsp-stream: use idle source in on_message_sent
When the underlying layers are running on_message_sent, this sometimes
causes the underlying layer to send more data, which will cause the
underlying layer to run callback on_message_sent again. This can go on
and on.

To break this chain, we introduce an idle source that takes care of
sending data if there are more to send when running callback

https://bugzilla.gnome.org/show_bug.cgi?id=797289
2018-10-23 08:18:52 +01:00
Patricia Muscalu
c394de2348 New property for socket binding to mcast addresses
By default the multicast sockets are bound to INADDR_ANY,
as it's not allowed to bind sockets to multicast addresses
in Windows. This default behaviour can be changed by setting
bind-mcast-address property on the media-factory object.

https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-28 13:27:48 +03:00
Patricia Muscalu
cbe6ae3c48 stream: Added a list of multicast client addresses
When media is shared, the same media stream can be sent
to multiple multicast groups. Currently, there is no API
to retrieve multicast addresses from the stream.
When calling gst_rtsp_stream_get_multicast_address() function,
only the first multicast address is returned.
With this patch, each multicast destination requested in SETUP
will be stored in an internal list (call to
gst_rtsp_stream_add_multicast_client_address()).
The list of multicast groups requested by the clients can be
retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
There still exist some problems with the current implementation
in the multicast case:
1) The receiving part is currently only configured with
regard to the first multicast client (see
https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2) Secondly, of security reasons, some constraints should be
put on the requested multicast destinations (see
https://bugzilla.gnome.org/show_bug.cgi?id=796916).

Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:31:42 +03:00
Patricia Muscalu
4c6cecf5d6 stream: Choose the maximum ttl value provided by multicast clients
The maximum ttl value provided so far by the multicast clients
will be chosen and reported in the response to the current
client request.

Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:31:42 +03:00
Patricia Muscalu
a7bb684e9b Add new API for setting/getting maximum multicast ttl value
Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:31:41 +03:00
Sebastian Dröge
443c2b73e5 Revert "Add new API for setting/getting maximum multicast ttl value"
This reverts commit 7f0ae77e40.

Commits where accidentially squashed together
2018-08-14 14:25:42 +03:00
Sebastian Dröge
29ae15f6f1 Revert "stream: Choose the maximum ttl value provided by multicast clients"
This reverts commit 499e437e50.

Commits where accidentially squashed together
2018-08-14 14:25:14 +03:00
Patricia Muscalu
499e437e50 stream: Choose the maximum ttl value provided by multicast clients
The maximum ttl value provided so far by the multicast clients
will be chosen and reported in the response to the current
client request.

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:10:41 +03:00
Patricia Muscalu
7f0ae77e40 Add new API for setting/getting maximum multicast ttl value
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:10:20 +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
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
Mathieu Duponchelle
a093f4442b rtsp-stream: extract handle_keymgmt from rtsp-client
rtspclientsink will also need to parse KeyMgmt headers
sent by the server to decrypt the RTCP backchannel stream

https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-30 17:55:32 +02:00
Tim-Philipp Müller
2df75442d0 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
We need different export decorators for the different libs.
For now no actual change though, just rename before the release,
and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:37:13 +00:00
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
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
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
51d670f73b rtsp-stream: add functions to get rtp and rtcp multicast sockets
Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +02:00
Tim-Philipp Müller
b344248630 Mark symbols explicitly for export with GST_EXPORT 2017-05-18 10:35:18 +01:00
Aleix Conchillo Flaqué
85c52e194b sdp: add rollover counters for all sender SSRC
We add different crypto sessions in MIKEY, one for each sender
SSRC. Currently, all of them will have the same security policy, 0.

The rollover counters are obtained from the srtpenc element using the
"stats" property.

https://bugzilla.gnome.org/show_bug.cgi?id=730539
2016-06-14 11:14:48 +02:00
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
Patricia Muscalu
f62a9a7eb9 rtsp-stream: postpone UDP socket allocation until SETUP
Postpone the allocation of the UDP sockets until we know
what transport has been chosen by the client.
Both unicast and multicast UDP sources are created in one
function.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 17:05:15 +02:00
Jan Schmidt
b6ca057c72 rtsp-stream: Add functions for using rtsp-stream from the client
Add a boolean to indicate that the rtsp-stream is running on the
'client' side of an RTSP connection, for sending streams via
RECORD. In that case, the roles of the client/server ports
in transport setup are swapped.

https://bugzilla.gnome.org/show_bug.cgi?id=758180
2016-01-29 01:44:26 +11:00
Xavier Claessens
0ea68a1b0f rtsp-server: Add g_autoptr() support to all types
https://bugzilla.gnome.org/show_bug.cgi?id=754464
2015-12-14 13:52:17 -05:00
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
Sebastian Dröge
8700468499 rtsp-server: Use single-include rtsp header to make sure we get all definitions 2015-05-20 17:05:47 +03:00
Hyunjun Ko
de590b4b2a rtsp-stream: fix to get valid each stream data for request-aux-sender signal
Because of duplicated g_signal_connect for request-aux-sender signal,
wrong stream pointer is passed to the signal handler.
Instead of passing each stream, pass stream array and get the relevant stream.

https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-15 15:14:04 +02:00
Tim-Philipp Müller
dc43f427a9 rtsp-stream: minor code formatting fix 2015-02-11 17:25:35 +00:00
Sebastian Dröge
ccf6c6eb53 Add initial support for RECORD
We currently only support media that is RECORD or PLAY only, not both at once.

https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-02-06 09:42:42 +01:00
Matthew Waters
4f40781fff media: implement ssrc-multiplexed retransmission support
based off RFC 4588 and the server-rtpaux example in -good
2014-12-16 16:41:08 +01:00
Srimanta Panda
376488d8c7 rtsp-media: Make sure that sequence numbers are monotonic after pause
The sequence number is not monotonic for RTP packets after pause. The
reason is basepayloader generates a randon sequence number when the
pipeline goes from ready to pause. With this fix generation of sequence
number will be monotonic when going from pause to play request.

https://bugzilla.gnome.org/show_bug.cgi?id=736017
2014-09-12 17:29:30 +03:00
Sebastian Dröge
6ba5ca447f rtsp-media: Query position and stop time only on the RTP parts of the pipeline
The RTCP parts, in specific the RTCP udpsinks, are not flushed when
seeking and will always continue counting the time. This leads to
the NPT after a backwards seek to be something completely different
to the actual seek position.

https://bugzilla.gnome.org/show_bug.cgi?id=732644
2014-08-12 10:54:12 +03:00
Wim Taymans
377ca6ed0f stream: add method to set crypto info
Make a method to configure the crypto information of a stream.
Set udpsrc in READY instead of PAUSED so that we can configure caps
later.
2014-04-03 17:26:12 +02:00
Wim Taymans
ae1fe21436 stream: add property to configure profiles 2014-01-07 12:39:58 +01:00
Wim Taymans
a1202effda stream: add method to check supported transport
Add a method to check if a transport is supported
2014-01-07 12:39:57 +01:00
Wim Taymans
8aaa432d58 stream: return clock-rate from get_rtpinfo
And use it to correct the rtptime to the requested start-time.

See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2013-12-26 17:14:06 +01:00
Wim Taymans
cfdc7408b5 stream: also return the running-time
Return the running-time in the rtpinfo as well.
2013-12-26 16:29:39 +01:00
Tim-Philipp Müller
91fac8eb29 rtsp-server: add padding to many public structures
Not mini objects though, since they are not subclassable
anyway, nor kept on the stack or inlined in a structure.
2013-12-12 00:36:07 +00:00
Wim Taymans
bdef631218 stream: add API to block streams
Add an API to block on the streams and make it post a message.

Based on patch by Ognyan Tonchev <ognyan@axis.com>

See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 16:18:39 +01:00
Aleix Conchillo Flaque
e5332535a7 rtsp-stream: add getter for payload type
* gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.

* gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
  element and create the stream with this one instead of the dynpay%d
  element.

  https://bugzilla.gnome.org/show_bug.cgi?id=712396
2013-11-22 11:19:35 +01:00
Wim Taymans
a106950f70 stream: add method to filter transports
Add a method to safely iterate and collect the stream transports

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2013-11-18 11:18:15 +01:00
Wim Taymans
8d5ce0d4ee stream: Add functions to get rtp and rtcp sockets
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 12:04:55 +01:00
Wim Taymans
a84b71c0f0 stream: add protocols property 2013-08-16 16:08:43 +02:00
Wim Taymans
d357fc55af docs: more updates 2013-07-11 12:24:33 +02:00
Wim Taymans
d4e8d800c9 stream: add method to check control url of stream 2013-07-03 15:13:45 +02:00
Wim Taymans
2ffb0f69d2 stream: add methods and property to set control string 2013-07-02 14:50:30 +02:00