Commit graph

947 commits

Author SHA1 Message Date
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