Commit graph

294 commits

Author SHA1 Message Date
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
406ed190ac rtsp-server: Fix indentation 2016-03-02 11:48:49 +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
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
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
Jan Schmidt
9e92a0307c rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS 2015-11-17 01:12:28 +11:00
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
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
51ed357597 rtsp-client: Only error out in PLAY if seeking actually failed
If the media was just not seekable, we continue from whatever position we are
and let the client decide if that is what is wanted or not.

Only if the actual seek failed, we can't really recover and should error out.
2015-02-13 12:21:16 +02:00
Sebastian Dröge
98b162f54b rtsp-media: If seeking fails, don't wait forever for the media to preroll again
Instead error out properly the same way as if the SEEKING query already
failed.
2015-02-12 16:53:27 +02:00
Tim-Philipp Müller
57c21c8f9e rtsp-client: fix awkward if clause 2015-02-08 12:08:36 +00:00
Sebastian Dröge
a93ed7e5d4 rtsp-media: Use flags to distinguish between PLAY and RECORD media 2015-02-06 09:42:50 +01:00
Tim-Philipp Müller
e9ce91634c rtsp-client: fix a couple of leaks in handle_announce 2015-02-06 09:42:50 +01:00
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
Tim-Philipp Müller
cc3e0ed39b rtsp-client: log interleaved data received 2015-01-19 23:24:28 +00:00
Tim-Philipp Müller
47eaac5b9e rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data 2015-01-19 23:18:02 +00:00
Sebastian Dröge
fcef562f35 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream 2015-01-19 13:09:20 +01:00
Sebastian Dröge
69e346419a rtsp-client: Use a random session ID in the SDP
RFC4566 Section 5.2 says that it should make the username, session id,
nettype, addrtype and unicast address tuple globally unique. Always using
1188340656180883 is not going to guarantee that: https://xkcd.com/221/

Instead let's create a 64 bit random number, which at least brings us
closer to the goal of global uniqueness.

https://tools.ietf.org/html/rfc4566#section-5.2
2015-01-18 19:08:36 +01:00
Sebastian Dröge
586fe4ea4b rtsp-client: Drop trailing \0 of RTSP DATA messages
We add a trailing \0 in GstRTSPConnection to make parsing of
string message bodies easier (e.g. the SDP from DESCRIBE) but
for actual data this means we have to drop it or otherwise
create invalid data.
2015-01-16 20:06:57 +01:00
Sebastian Dröge
79e41bc2be rtsp-client: Add a send_message default signal handler
This allows subclasses to easily hook into the response sending
mechanism without doing everything from a signal, which seems
awkward from subclasses.
2014-12-29 12:06:50 +01:00
Göran Jönsson
058698c9cf rtsp: Ref transports in hash table.
Also ref streams for transports.
This solves a crash when reciving a rtcp after teardown but before
client finalize.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2014-12-02 16:29:24 +01:00
Wim Taymans
bd8b2d3fb9 client: refactor cleanup of cached media 2014-11-07 12:48:53 +01:00
Linus Svensson
088eee6590 client: Configure transport after creating session media
The default implementation of configure_client_transport() in
rtsp-client uses the session media when it chooses channels for
interleaved traffic.

https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-11-07 12:42:48 +01:00
Linus Svensson
a455181aff client: Stop caching media in client when doing setup
If the media has been managed by a session media, it should not be
cached in the client any longer. The GstRTSPSessionMedia object is now
responsible for unpreparing the GstRTSPMedia object using
gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
session media.

https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-11-07 12:34:23 +01:00
Aleix Conchillo Flaqué
ef9dc6c9e4 rtsp-client: mikey memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739383
2014-10-30 10:34:56 +00:00
Aleix Conchillo Flaqué
0aad92531d rtsp-client: add stream transport to context
We add the stream transport to the context so we can get the configured
client stream transport in the setup request signal.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2014-10-21 11:44:40 +02:00
Aleix Conchillo Flaqué
6c0c90c9d2 client: set session media to NULL without the lock
We need to set session medias to NULL without the client lock otherwise
we can end up in a deadlock if another thread is waiting for the lock
and media unprepare is also waiting for that thread to end.

https://bugzilla.gnome.org/show_bug.cgi?id=737690
2014-10-01 10:31:04 +01:00
Sebastian Rasmussen
404a80e38a rtsp-client: Remove backlog limit while processings requests
If the backlog limit is kept two cases of deadlocks may be
encountered when streaming over TCP. Without the backlog
limit this deadlocks can not happen, at the expence of
memory usage.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
2014-09-30 12:22:49 +02:00
Ognyan Tonchev
17f5785638 rtsp-client: do not free main context before rtsp watch
https://bugzilla.gnome.org/show_bug.cgi?id=737110
2014-09-24 10:42:16 +03:00
Branko Subasic
2218510cae rtsp-*: Treat sending packets to clients as keepalive
As long as gst-rtsp-server can successfully send RTP/RTCP data to
clients then the client must be reading. This change makes the server
timeout the connection if the client stops reading.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-24 10:37:59 +03:00
Branko Subasic
733ef1162b rtsp-client: Allow backlog to grow while expiring session
Allow the send backlog in the RTSP watch to grow to unlimited size while
attempting to bring the media pipeline to NULL due to a session
expiring.  Without this change the appsink element cannot change state
because it is blocked while rendering data in the new_sample callback.
This callback will block until it has successfully put the data into the
send backlog. There is a chance that the send backlog is full at this
point which means that the callback may block for a long time, possibly
forever. Therefore the media pipeline may also be prevented from
changing state for a long time.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-24 10:37:49 +03:00
Edward Hervey
980553547d rtsp-client: Make old compilers happy
rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]

Just in case that guint8 doesn't fit in a pointer. Just in case ...
2014-09-22 09:30:39 +02:00
Göran Jönsson
23b9d8fbb0 client: raise the backlog limits before pausing
We need to raise the backlog limits before pausing the pipeline or else
the appsink might be blocking in the render method in wait_backlog() and
we would deadlock waiting for paused.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
2014-09-16 11:41:52 +02:00
Göran Jönsson
ebd9be59fe client: make define for the WATCH_BACKLOG
See https://bugzilla.gnome.org/show_bug.cgi?id=736322
2014-09-16 11:29:38 +02:00
Wim Taymans
0292be09ea client: simplify session transport handling
link/unlink of the transport in a session was done to keep track of all
TCP transports and to send RTP/RTCP data to the streams. We can simplify
that by putting all the TCP transports in a hashtable indexed with the
channel number.

We also don't need to link/unlink the transports when we pause/resume
the streams. The same effect is already achieved when we pause/play the
media. Indeed, when we pause the media, the transport is removed from
the media and the callbacks will not be called anymore.

See https://bugzilla.gnome.org/show_bug.cgi?id=736041
2014-09-16 10:46:13 +02:00
Göran Jönsson
09bf2025f8 rtsp-client: Protect saved clients watch with a mutex
Fixes a crash when close() is called while merging clients
in handle_tunnel(). In that case close() would destroy the
watch while it is still being used in handle_tunnel().

https://bugzilla.gnome.org/show_bug.cgi?id=735570
2014-09-04 10:35:56 +03:00
Wim Taymans
ced406cc28 client: expose _close() method
Expose a previously internal close method to close the client
connection.
2014-07-10 17:05:13 +02:00
Wim Taymans
945c93fde0 filter: Release lock in filter functions
Release the object lock before calling the filter functions. We need to
keep a cookie to detect when the list changed during the filter
callback. We also keep a hashtable to make sure we only call the filter
function once for each object in case of concurrent modification.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2014-07-10 11:36:55 +02:00
Ognyan Tonchev
6543082d2b client: check if watch is set in handle_teardown()
The unit tests run without a watch
2014-07-09 16:17:00 +02:00
Ognyan Tonchev
e0bc97e40c client: keep ref to client for the session removed handler
This extra ref will be dropped when all client sessions have been
removed. A session is removed when a client sends teardown, closes its
endpoint of the TCP connection or the sessions expires.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-07-09 16:16:50 +02:00
Wim Taymans
5aec4af1b9 client: manage media in session as a last step
Once we manage a media in a session, we can't unmanage it anymore
without destroying it. Therefore, first check everything before we
manage the media, otherwise if something is wrong we have no way to
unmanage the media.
If we created a new session and something went wrong, remove the session
again. Fixes a leak in the unit test.
2014-07-08 14:46:13 +02:00
Wim Taymans
99f670d8bc rtsp: fix for MIKEY api change 2014-07-02 16:04:53 +02:00
Wim Taymans
72a57e792f client: free watch context only once
The watch context is freed when the source is destroyed. Avoids
a CRITICAL when we try to unref the context twice.
2014-07-01 16:12:13 +02:00
Wim Taymans
517bb78ae3 client: fix build 2014-07-01 15:02:15 +02:00
Wim Taymans
5e2afcefdd client: protect sessions with lock
Protect the list of sessions with the lock.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-07-01 14:41:14 +02:00
Wim Taymans
fe081e7301 Client: keep a ref to the session
Don't just keep a weak ref to the session objects but use a hard ref. We
will be notified when a session is removed from the pool (expired) with
the new session-removed signal.
Don't automatically close the RTSP connection when all the sessions of
a client are removed, a client can continue to operate and it can create
a new session if it wants. If you want to remove the client from the
server, you have to use gst_rtsp_server_client_filter() now.

Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>

See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-07-01 12:28:41 +02:00
Evan Nemerson
cecc2cb4ff introspection: add missing allow-none annotations
https://bugzilla.gnome.org/show_bug.cgi?id=730952
2014-06-26 19:08:56 +02:00
Evan Nemerson
d08b46f4b7 gi: improve annotations
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2014-06-24 09:48:45 +02:00
Wim Taymans
661f4d928f signals: use generic marshal function
Use the generic C marshal function.
Use more explicit type instead of G_TYPE_POINTER
2014-06-24 09:43:44 +02:00
Wim Taymans
d676c56888 sdp: hide key length defines
They don't have a namespace.
2014-06-24 09:34:50 +02:00