Commit graph

1053 commits

Author SHA1 Message Date
Tim-Philipp Müller
8c1cdb7a4a win32: remove .def file with exports
They're no longer needed, symbol exporting is now explicit
via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
2017-11-26 13:14:12 +00:00
Tim-Philipp Müller
58aa58f049 rtsp-server: add missing GST_EXPORT and export deprecated funcs 2017-11-26 13:03:39 +00:00
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
7bf8c4d218 rtsp-client: Don't leak addr
CID #1422260
2017-11-21 09:53:19 +01:00
Edward Hervey
4d98bc5e55 Run gst-indent 2017-11-21 09:53:08 +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
Sebastian Dröge
d51f8abe56 rtsp-stream: Only update the RTP udpsink if it actually exists
For send-only streams it does not exist, but the RTCP udpsink might.
2017-11-15 19:56:26 +02:00
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
930a602e17 rtsp-stream: obtain stream position from pad
If no sinks have been added yet, obtain the current and
the stop position of the stream from the send_src pad.

Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +02:00
Patricia Muscalu
5ec1b80989 rtsp-session-media: add function to get a list of transports
Change-Id: I817e10624da0f3200f24d1b232cff481099278e3

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +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
Patricia Muscalu
c9605cc5e1 stream: set async=sync=false only for RTCP appsink
Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +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
Jonathan Karlsson
0f87202a71 rtsp-session: Handle the case when timeout=0
According to the documentation, a timeout of value 0 means
that the session never timeouts. This adds handling of that.
If timeout=0 we just return with a -1 from
gst_rtsp_session_next_timeout_usec ().

https://bugzilla.gnome.org/show_bug.cgi?id=785058
2017-11-15 17:20:33 +02:00
Mathieu Duponchelle
89ccaa6932 docs: add media factory transport mode accessors
and fix the documentation for the return value of the getter
2017-10-26 14:44:55 +02:00
Branko Subasic
619ac7b710 rtsp-client: unref 'pipelined_requests' in finalize
The hash table priv->pipelined_requests is not unref:ed in the
finalize funktion. Make sure it is.

https://bugzilla.gnome.org/show_bug.cgi?id=788704
2017-10-09 20:39:14 +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
8b38aa9c46 stream: Use stream duration as stream-stop if segment was not configured with a stop
Allowing client to know stream duration when no seeking happened.

https://bugzilla.gnome.org/show_bug.cgi?id=783435
2017-10-05 12:07:13 -03:00
Sebastian Dröge
c04e3b07dd rtsp-media-factory: Don't cache any media if NULL was returned as key
The docs already mentioned this, but we actually stored it in the hash
table with key==NULL and leaked its reference forever.
2017-09-25 19:41:33 +03:00
Satya Prakash Gupta
d690fbd37d sdp: fix Memory leak in error case
https://bugzilla.gnome.org/show_bug.cgi?id=787059
2017-08-31 11:04:05 +01:00
Sebastian Dröge
ffbabb1529 rtsp-client: Fix typo in debug message 2017-08-14 21:04:58 +03:00
Julien Isorce
d72284bdf8 rtsp-stream: fix connection delay due to wrong assumption on last-sample
Commit 852cc09f54 assumed that
multiudpsink's last-sample always comes from the payloader. Which
is wrong if auxiliary streams are multiplexed in the same stream.

So check the buffer's ssrc against the caps'ssrc before to use its
seqnum. If not the same ssrc just use the payloader as done prior
the commit above or when there is no last-sample yet.

https://bugzilla.gnome.org/show_bug.cgi?id=784094
2017-06-29 14:52:09 +01:00
Tim-Philipp Müller
b344248630 Mark symbols explicitly for export with GST_EXPORT 2017-05-18 10:35:18 +01:00
Thibault Saunier
b56930704f gi: Fix some annotations and docstrings 2017-04-13 14:20:10 -03:00
Thibault Saunier
133e91462a meson: Build gir 2017-04-13 14:11:43 -03:00
Sebastian Dröge
cd4e675f0c rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
If there is no Content-Length header, no body would be allocated and the
'\0' would also not be appended to the body.
2017-01-19 14:57:19 +02:00
Sebastian Dröge
ac1124efb4 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
While they logically have 0 bytes length, GstRTSPConnection is appending
a '\0' to everything making the size be 1 instead.
2017-01-19 14:24:07 +02:00
Sebastian Dröge
6e145fadf9 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
affected.
2017-01-12 19:04:23 +02:00
Patricia Muscalu
fb7833245d rtsp-stream: corrected if-statement in _get_server_port()
This bug was accidentally introduced while fixing a segfault
in _get_server_port() function.

https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-10 10:38:13 +00:00
Patricia Muscalu
f47e6ab9f6 rtsp-stream: fixed segmenation fault in _get_server_port()
Calling function gst_rtsp_stream_get_server_port() results in
segmenation fault in the RTP/RTSP/TCP case.
Port that the server will use to receive RTCP makes only
sense in the UDP case, however the function should handle
the TCP case in a nicer way.

https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-09 15:27:40 +02:00
Aleksandr Slobodeniuk
b27e7c6b5b dosc: Fix a little typo
https://bugzilla.gnome.org/show_bug.cgi?id=777037
2017-01-09 10:19:53 +00:00
Patricia Muscalu
42f270e7f2 rtsp-stream: Fixed TCP transport case
Make sure that the appsink element is actually added to
the bin before trying to link it with the elements in it.

https://bugzilla.gnome.org/show_bug.cgi?id=776343
2016-12-22 14:21:54 +02:00
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
Edward Hervey
8317139121 media-factory: Create media objects with the proper transport mode
The function called immediately afterwards (collect_streams()) will
need it to work properly
2016-12-02 15:47:12 +01:00
Sebastian Dröge
d633c0103a rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected 2016-12-02 14:36:50 +02:00
Sebastian Dröge
708fd3c325 rtsp-media-factory: Don't create a pipeline for the media pipeline string
We're going to put a pipeline into a pipeline otherwise, which is not
exactly ideal.
2016-12-01 18:04:34 +02:00
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
Matthew Waters
b38eb8e99e stream: block the output of rtpbin instead of the source pipeline
85c52e194b introduced a more correct
detection of the srtp rollover counter to add to the SDP.

Unfortunately, it was incomplete for live pipelines where the logic
blocks the source bin before creating the SDP and thus would never have
the necessary informaiton to create a correct SDP with srtp encryption.

Move the pad blocks to rtpbin's output pads instead so that the
necessary information can be created before we need the information for
the SDP.

https://bugzilla.gnome.org/show_bug.cgi?id=770239
2016-11-23 23:08:16 +11:00
Dag Gullberg
f00ac2daf2 rtsp-client: add IDLE timeout, before session exists
The RTSP server will not timeout an idle RTSP connection
(note this is different from doing timeout on a RTSP
session).

At least for Apache this is a problem when running RTSP over
HTTPS since it uses one of the threads (there is a rather
limited number) that are available for handling requests.

https://bugzilla.gnome.org/show_bug.cgi?id=771830
2016-11-23 09:45:33 +00:00
Göran Jönsson
335d279a96 rtsp-stream: Set close-socket FALSE on UDP src:es
With this RTSP server can use the sockets independent on the udpsrc
state.
When the udp src is finalized it will unref socket and when g_socket
is finalized the socket will be closed.

https://bugzilla.gnome.org/show_bug.cgi?id=765673
2016-11-22 13:59:30 +02:00
Sebastian Dröge
927a44c55b rtsp-auth: Add support for Digest authentication
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-19 11:59:34 +02:00
Scott D Phillips
d7676bfba3 Enable building with MSVC
https://bugzilla.gnome.org/show_bug.cgi?id=774640
2016-11-19 11:58:05 +02:00
Scott D Phillips
01ef7f32b6 client: update do_send_message to match type GstRTSPClientSendFunc
This type mismatch fails building with MSVC

https://bugzilla.gnome.org/show_bug.cgi?id=774640
2016-11-17 23:38:15 +00:00
Sebastian Dröge
179eb9ae89 rtsp-sdp: Fix indentation 2016-11-11 14:42:08 +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
Branko Subasic
8425ea6969 client: emit signal in the beginning of each rtsp request
These signals let the application validate the requests, configure the
media/stream in a certain way and also generate error status code in
case of error or bad request.

https://bugzilla.gnome.org/show_bug.cgi?id=758062
2016-11-01 20:25:22 +02:00
Göran Jönsson
dbf91ab231 rtsp-client: Session filter in unwatch session
Call session filter with filter_session_media as paramer in
client_unwatch_session if using drop_backlog = FALSE.

In client_unwatch_session its allowed to grow the watchs backlog.
If using drop_backlog = FALSE and the backlog is full it will cause
a deadlock when setting session media state to NULL
if the backlog is not allowed to grow.

https://bugzilla.gnome.org/show_bug.cgi?id=771983
2016-10-25 12:55:59 +03:00
Nikita Bobkov
ff65732270 rtsp-client: Fix factory leaking in find_media() in error cases
https://bugzilla.gnome.org/show_bug.cgi?id=771488
2016-10-20 14:01:38 +03:00
Xavier Claessens
c0f24fea83 stream: Fix randomly missing streams from SDP with dynamic elements
When using dynamic elements, gst_rtsp_stream_join_bin() is called from
"pad-added" signal. In that case priv->srcpad could already have its caps,
and they'll be sent to priv->send_src[0] pad. That means that when it
connects "notify::caps" signal, that pad could already have received its
caps and the signal won't be emitted anymore.

In that case priv->caps stay to NULL and when building the SDP that stream
gets ignored. Leading to missing video or audio when playing in client side.

https://bugzilla.gnome.org/show_bug.cgi?id=772478
2016-10-06 19:05:36 +03:00
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
74c8a9f4cf rtsp-stream: Remove unused _locked() variant of a function
It was added during refactoring.
2016-09-07 18:44:34 +03:00
Xavier Claessens
e882fe9e06 stream: cosmetic cleanup
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-07 18:40:57 +03:00
Xavier Claessens
f5f350645a stream: Compare IP addresses case insensitive in more places
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-07 18:40:57 +03:00
Xavier Claessens
f90ab92547 stream: Fix leaked joined_bin
There is no need to keep a strong ref on it, and _leave_bin() was
setting it to NULL before calling g_clear_object() so it was leaked.

https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-07 18:40:57 +03:00
Sebastian Dröge
d33eca6156 rtsp-stream: Compare IP address strings case insensitive
Otherwise IPv6 addresses might fail this comparision.
2016-09-06 19:15:23 +03:00
Sebastian Dröge
e5a49efa7f rtsp-stream: Bind multicast sockets to ANY as before
https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
2016-09-06 19:10:21 +03:00
Kseniia
6136ef66d4 rtsp-session: Fix segfault when doing keep-alive after removing the session
If keep-alive happens after removing the session but before finalizing the
stream transport, we would segfault.

https://bugzilla.gnome.org/show_bug.cgi?id=750544
2016-09-05 22:57:52 +03:00
Sebastian Dröge
ca855abae1 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
Adding them later will cause deadlocks due to
1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2) adding the multicast sink
3) waiting for it to get data to preroll again

3) never happens because the queues after the tee are full.
2016-09-05 18:09:22 +03:00
Sebastian Dröge
be4b9718e3 rtsp-stream: Fix up various multicast related issues 2016-09-05 16:32:57 +03:00
Xavier Claessens
8495c47a9d stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
This is basically reverting changes introduced in commit f62a9a7,
because it was introducing various regressions:

- It introduces a leak of udpsrc elements that got wrongly fixed by adding
  an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
  ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
- If a mcast client connects, it creates a new socket in SETUP to try to respect
  the destination/port given by the client in the transport, and overrides the
  socket already set on the udpsink element. That means that if we already had a
  client connected, the source address on the udp packets it receives suddenly
  changes.
- If a 2nd mcast client connects, the destination/port in its transport is
  ignored but its transport wasn't updated.

What this patch does:

- Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
- Always have a tee+queue when udp is enabled. This could be optimized
  again in a later patch, but is more complicated. If no unicast clients
  connects then those elements are useless, this could be also optimized
  in a later patch.
- When mcast transport is added, it creates a new set of udpsrc/udpsink,
  seperated from those for unicast clients. Since we already support only
  one mcast address, we also create only one set of elements.

https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:36:17 +03:00
Xavier Claessens
aa0e60445d stream: factor our plug_src function
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:26:08 +03:00
Xavier Claessens
47a3956b48 stream: factor out plug_sink function
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:26:02 +03:00
Xavier Claessens
a44f198ffc stream: small documentation clarification
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:25:57 +03:00
Xavier Claessens
82a618c2e6 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:25:51 +03:00
Xavier Claessens
55a1df5724 stream: Keep a ref on joined bin
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:25:39 +03:00
Xavier Claessens
3ff4529a92 stream: code cleanup
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:24:06 +03:00
Xavier Claessens
2b223af792 stream: small fix in error code path
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:24:01 +03:00
Xavier Claessens
07f17c2cce Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
This partly reverts commit cba045e1b1,
but keeps unit tests.

https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:23:53 +03:00
Tim-Philipp Müller
a353e50747 Add support for Meson as alternative/parallel build system
https://github.com/mesonbuild/meson
2016-08-31 00:04:43 +01:00
Nikita Bobkov
de3d0c4522 rtsp-client: Fix leaking of media in error cases
With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
and myself to make the media refcounting a bit easier to follow.

https://bugzilla.gnome.org/show_bug.cgi?id=755632
2016-08-02 17:46:49 +03:00
Sebastian Dröge
687301af86 rtsp-client: Fix leaking of session in error cases
https://bugzilla.gnome.org/show_bug.cgi?id=755632
2016-08-02 15:18:30 +03:00
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
Tim-Philipp Müller
fc2554404b docs: fix some typos 2016-06-07 20:44:42 +01:00
Tim-Philipp Müller
7de0d6580a g-i: pass compiler env to g-ir-scanner
It's what introspection.mak does as well. Should
fix spurious build failures on gnome-continuous
(caused by g-ir-scanner getting compiler details
via python which is broken in some environments
so passing the compiler details bypasses that).
2016-05-25 10:28:43 +01:00
Ian
178f2d6fe5 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
This works with rtspsrc and live555, but fails with e.g. ffmpeg.

https://bugzilla.gnome.org/show_bug.cgi?id=766619
2016-05-19 11:57:33 +03:00
Jake Foytik
fe5f8077c1 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
- Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
 - Create unit test for shared media.

https://bugzilla.gnome.org/show_bug.cgi?id=764744
2016-04-29 11:49:14 +03:00
Sebastian Dröge
aa9a2443a1 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
For IPv6 addresses, binding to a multicast group does not work on Linux
either. Always bind to ANY and then later join the multicast group.

https://bugzilla.gnome.org/show_bug.cgi?id=764679
2016-04-29 11:48:57 +03:00
Patricia Muscalu
f0891e2cdf rtsp-thread-pool: explained why GSource is a part of ThreadImpl
Clarified why it is necessary to add source information to
GstRTSPThreadImpl. See the reported bug in GLib:
https://bugzilla.gnome.org/show_bug.cgi?id=720186
for more information.

https://bugzilla.gnome.org/show_bug.cgi?id=761702
2016-04-06 09:46:34 +01:00
Sebastian Dröge
60dd95849f rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS) 2016-04-03 12:06:29 +03:00
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
Sebastian Dröge
8e72e69eec rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
This would get us NO_PREROLL in the bin again and break seeking.
Thanks to Carlos Rafael Giani for helping to debug this!

https://bugzilla.gnome.org/show_bug.cgi?id=740509
2016-03-16 23:36:30 +02:00
Sebastian Dröge
8b68edd138 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
Without this, RECORD pipelines are broken because
a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
added later. Previously it was there earlier and due to NO_PREROLL caused the
pipeline to preroll immediately
b) the udpsrc for the pipeline is added later and never set to PLAYING state,
as the corresponding code previously was only for PLAY pipelines.

https://bugzilla.gnome.org/show_bug.cgi?id=763281
2016-03-10 19:47:13 +02:00
Jan Schmidt
4a6f63ad03 rtsp-stream: Fix typo in the docstring
gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
2016-03-11 01:23:15 +11:00
Sebastian Dröge
206d2ded09 rtsp-stream: Disable multicast loopback for all our sockets
On Windows this is a receiver-side setting, on Linux a sender-side setting. As
we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
loopback setting on the socket... while udpsink does which unfortunately has
no effect here on Windows but on Linux.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-05 10:53:15 +02:00
Sebastian Dröge
9794822549 rtsp-stream: Only bind multicast sockets to ANY on Windows
On Linux it is still needed to bind to the multicast address
to filter out random other packets, while on Windows binding
to multicast addresses just fails.
2016-03-04 13:51:12 +02:00
Sebastian Dröge
a7ced98346 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
Otherwise we fail to allocate UDP ports if the pool only contains multicast
addresses, which is something that used to work before. For unicast addresses
if the pool contains none, we just allocate them as if there is no pool at
all.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-03 10:43:13 +02:00
Sebastian Dröge
406ed190ac rtsp-server: Fix indentation 2016-03-02 11:48:49 +02:00
Sebastian Dröge
bcee3202d3 rtsp-stream: Don't bind the sockets to multicast addresses
This works on Linux but fails completely on Windows. You're supposed
to bind to ANY and then join the multicast group.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-02 11:47:47 +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
Patricia Muscalu
d10ba734cd rtsp-stream: postpone the creation of the UDP sources
Code refactoring: allocate the UDP ports after the sender and
the reciver parts have been created.
We postpone the creation of the UDP sources until the UDP
ports have been allocated.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 17:05:15 +02:00
Patricia Muscalu
66389cb900 rtsp-stream: added function for setting UDP sources to PLAYING state
Code refactoring: Introduced a function for setting UDP sources
to PLAYING state.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 17:05:15 +02:00
Patricia Muscalu
c0cadc6ec3 rtsp-stream: added function for creating and configuring UDP sources
Code refactoring: create and configure UDP sources in a separate function.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 17:05:15 +02:00
Patricia Muscalu
b26c16c824 rtsp-stream: added function for RTP/RTCP socket configuration
Code refactoring: configure RTP and RTCP sockets for UDP sinks
in a separate function.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 17:05:15 +02:00
Patricia Muscalu
6b6970ab23 rtsp-stream: added function for creating and configuring UDP sinks
Code refactoring: create and configure UDP sinks in a separate function.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 17:05:15 +02:00
Patricia Muscalu
89bc8009dd rtsp-stream: added helper function for creating the sender/receiver parts
Code refactoring: introduced helper function for creating
the receiver and the sender parts of the streaming pipeline.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 17:05:15 +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
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
Jan Schmidt
192a1eea34 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
A new function that adds info from a GstRTSPStream into an SDP message.

https://bugzilla.gnome.org/show_bug.cgi?id=758180
2016-01-29 01:44:26 +11:00
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
Tim-Philipp Müller
ac1d35b147 rtsp-media-factory: add missing break in "clock" property setter
CID 1348453
2016-01-15 07:01:37 +00:00
Srimanta Panda
fdbda049c6 rtsp-stream: fixed assert during update transport
When RTSP server trying update transport during multicast, it throws an
assert. The assert is thrown because it is trying to get the parent of
an non-existing funnel element.

https://bugzilla.gnome.org/show_bug.cgi?id=760150
2016-01-07 14:31:03 +02:00
Tim-Philipp Müller
bec94861b0 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
gtk-doc can handle static inline functions just fine these days,
there's no need for this stuff any more.
2016-01-03 17:26:31 +00:00
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 Dröge
6b76c02552 rtsp-media-factory: Add FIXME for 2.0 2015-12-30 16:30:38 +02:00
Sebastian Dröge
3d6b93bcd3 rtsp-stream: Fix indentation 2015-12-30 16:29:45 +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
Olivier Crête
ee3a7b61ef rtsp-session-pool: Avoid dollar sign ($) in session ids
Live555 in VLC strips off dollar signs and then gets very confused,
we don't loose too much entropy by just skipping it.
2015-12-15 16:57:37 -05:00
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
Srimanta Panda
f96947b350 rtsp-stream: fixed valgrind error
Fixed the valgrind error in unit test. The UDP source created during
gst_rtsp_stream_join_bin() was not released while destroying the rtp
bin.

https://bugzilla.gnome.org/show_bug.cgi?id=759010
2015-12-08 09:47:53 +02:00
Srimanta Panda
ed70572c6c rtsp-client: suspend media during setup request
SETUP request from clients needs to suspend the media to clear the
prerolled buffers. Otherwise it will not affect the prerolled buffer
and the prerolled buffers will be incorrect (for example block-size
from setup request will not affect the prerolled buffer unless the
media is suspended).

https://bugzilla.gnome.org/show_bug.cgi?id=758268
2015-12-04 15:48:23 +02:00
Srimanta Panda
82dffd17b3 rtsp-stream: create stream pipeline based on transport
Based on the protocol, create the rtsp stream pipeline. If only TCP or
only UDP is set as the transport protocol, it will not add the extra tee
or queue element to the pipeline. Both these elements will be added, if
it supports both TCP and UDP protocols. This improves the pipeline
performance when one protocol is present.

https://bugzilla.gnome.org/show_bug.cgi?id=758179
2015-12-04 14:13:10 +02:00
Sebastian Dröge
61772cb326 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
Adding them when not needed will start some logic inside rtpbin that might be
problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
would start up a rtpjitterbuffer and behave in weird ways.

We still set up the UDP sources for RTP receiving for a sender media to be
able to receive any packets sent by the client for NAT traversal. They will
all go to a fakesink though.

Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
NO_PREROLL, which will cause deadlocks when seeking the media as it will never
receive ASYNC_DONE after a seek.

https://bugzilla.gnome.org/show_bug.cgi?id=758319
2015-12-01 15:32:45 +02:00
Sebastian Dröge
cdc0849dfe rtsp-stream: Disable multicast loopback for the multicast udp sources too
On POSIX this setting is for sender sockets, on Windows for receiver sockets.
Previously we were only setting this for sender sockets, which caused looped
back packets to be received on Windows if a multicast transport was used.
2015-11-17 12:45:58 +02:00
Jan Schmidt
9e92a0307c rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS 2015-11-17 01:12:28 +11:00
Marcus Prebble
b90d4ba917 rtsp-server: Change the logic so we don't pop a NULL context
When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
will sometimes fail. This call is made before any context is pushed
resulting in an attempt to pop a NULL context.

https://bugzilla.gnome.org/show_bug.cgi?id=757949
2015-11-11 15:58:27 +01:00
David Svensson Fors
81ae320383 rtsp-stream: Always unref return value of gst_object_get_parent()
Fixes a leak of a GstBin in the udp-mcast case.

https://bugzilla.gnome.org/show_bug.cgi?id=756968
2015-10-22 19:28:15 +03:00
Hyunjun Ko
a51337974c stream: listen to sender ssrc signals
https://bugzilla.gnome.org/show_bug.cgi?id=746747
2015-10-02 16:40:31 +03:00
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
Tim-Philipp Müller
da8a31ac88 stream: fix docs for recently-added get/set_buffer_size API
https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-09-17 20:07:34 +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
Jan Schmidt
9bfcdba42b rtsp-media: Fix small typo causing gtk-doc to complain 2015-09-03 22:16:30 +10:00
Hyunjun Ko
4c6b1faa6a media-factory: get port number through gst_rtsp_url_get_port
https://bugzilla.gnome.org/show_bug.cgi?id=753473
2015-08-16 12:08:49 +02:00
Xavier Claessens
8511ffe178 Document that source keeps a ref on server until it's destroyed
https://bugzilla.gnome.org/show_bug.cgi?id=749227
2015-08-10 12:18:53 -04:00
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
Vineeth TM
3920e21cd0 rtsp-media: assertion error due to wrong condition check
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j

https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-30 15:52:08 +03:00
Sebastian Dröge
ae7bec97cb rtsp-media: Strip keys from the fmtp that we use internally in our caps
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps

https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-29 11:28:21 +01:00
Xavier Claessens
5585dc5878 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
https://bugzilla.gnome.org/show_bug.cgi?id=752640
2015-07-20 16:47:05 -04:00
Ognyan Tonchev
8922afb88d rtsp-client: allow application to decide what requirements are supported
Add "check-requirements" signal and vfunc to allow application
(and subclasses) to check the requirements.

Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>

https://bugzilla.gnome.org/show_bug.cgi?id=749417
2015-06-23 14:38:29 +01:00
Ognyan Tonchev
fb71b9c4e9 rtsp-media: Always use real payloader when creating streams
A bin that contains the real payloader might be used as payloader. In this
case we have to get the real payloader for the various properties it provides.

Example use cases for this are bins that payload some media and then have
additional elements that add metadata or RTP extension headers to the stream.

https://bugzilla.gnome.org/show_bug.cgi?id=750800
2015-06-16 11:09:37 +02:00
Hyunjun Ko
2a3dd3d38f rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-11 11:37:03 +01:00
Xavier Claessens
6ec8fe44b2 GstRTSPAuth: Add client certificate authentication support
https://bugzilla.gnome.org/show_bug.cgi?id=750471
2015-06-09 19:51:46 -04:00
Göran Jönsson
08e0c79cee rtsp-client: No flush during Teardown.
When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
backlog is empty it can happen that just a part of a message will be
sent and rest is in backlog queue. If then flush during teardown
just a part of message will be sent.This can lead to client miss
teardown response since it expect to get the last part of message.

The flushing during teardown was introduced to fix a deadlock that now
is fixed more generally in handle_request by temporary  setting backlog
size to unlimited.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
2015-06-03 15:09:10 +02:00
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