Commit graph

699 commits

Author SHA1 Message Date
Nirbheek Chauhan
95ef0a1df8 Update docs cache and fix before-send signal doc syntax
The docs for before-send were missing because of this

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>
2021-03-17 15:55:30 +05:30
Nirbheek Chauhan
73fb107a85 rtspsrc: Fix accumulation of before-send signal return values
Since glib 2.62, the accumulated return values in RUN_CLEANUP override the
accumulated return values in RUN_FIRST. Since:

1. We have a default handler that always returns TRUE, and
2. User handlers are only run in RUN_FIRST, and
3. Our accumulator just takes the latest return value

We were discarding the return value from the user handler and always
sending messages even if the user handler said not to. See
https://gitlab.gnome.org/GNOME/glib/-/issues/2352 for more details.

This signal does not need RUN_CLEANUP or RUN_FIRST, so just change it
to RUN_LAST so that it's emitted exactly once and accumulated once.

With this fix, this signal can now be used to intercept PAUSE when
going to GST_STATE_NULL so that the server does a TEARDOWN (if
necessary) and not a PAUSE, which will confuse other RTSP clients when
playing shared media.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>
2021-03-17 14:05:44 +05:30
Nirbheek Chauhan
55edd59f48 Revert unusable workaround for PAUSE being sent when going NULL
Directly setting rtspsrc to the NULL state before putting the pipeline
in the NULL state usually works, but it can cause a deadlock in some
cases, so it's not a reliable mechanism to fix this.

This reverts commit f37afdafff:
"rtspsrc: Fix state changes from PAUSED to PLAYING"

and commit 76d624b2df:
"rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/908>
2021-03-17 14:04:16 +05:30
Nirbheek Chauhan
f37afdafff rtspsrc: Fix state changes from PAUSED to PLAYING
This was accidentally broken in the last commit that touched this
because I missed the fall-through in the case immediately above this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/906>
2021-03-16 00:10:24 +05:30
Nirbheek Chauhan
76d624b2df rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL
This usually doesn't matter, but it is disruptive when streaming from
a shared media since it will pause all other clients when any client
exits.

This new behaviour is opt-in and should be safe because you need to
set the NULL state on rtspsrc directly, instead of just on the
pipeline. See the updated documentation for an explanation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/901>
2021-03-15 12:59:09 +05:30
Mathieu Duponchelle
f2d909eb4e rtspsrc: fix title of a few properties docstrings
GstRtspSrc -> GstRTSPSrc

This would have been noticed by the since checker, but those
properties were introduced prior to that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/899>
2021-03-11 22:22:15 +01:00
Marc Leeman
c7356c70ee gstrtspsrc: 551 should not result in an unhandled error
Some cameras (e.g. HikVision DS-2CD2732F-IS) return "551 Option
not supported" when a command is sent that is not implemented
(e.g. PAUSE). Instead; it should return "501 Not Implemented".

This is wrong, as previously, the camera did announce support for PAUSE
in the OPTIONS.

In this case, handle the 551 as if it was 501 to avoid throwing errors
to application level. */

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/885>
2021-03-04 13:07:49 +01:00
Tobias Ronge
706d91371c rtspsrc: Do not wait for response while flushing
Due to the may_cancel flag in GstRTSPConnection, receiving might not get
cancelled when supposed to. In this case, gst_rtsp_src_receive_response
will have to wait until timeout instead but if busy receiving RTP
data, this timeout will never occur.

With this patch, gst_rtsp_src_receive_response returns GST_RTSP_EINTR
if flushing is set to TRUE instead of continuing to receive.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/831>
2021-01-15 09:24:51 +00:00
Sebastian Dröge
39c6bc0507 rtspsrc: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/842>
2020-12-21 09:59:43 +00:00
ChrisDuncanAnyvision
d9ea3346f3 rtspsrc: Ensure same group-id used for both TCP/UDP stream-start events
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>
2020-11-10 18:18:12 +00:00
ChrisDuncanAnyvision
e5f5e712c6 rtspsrc: Use consistent URI hashed stream-id for UDP and TCP/Interleaved streams
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>
2020-11-10 16:23:17 +00:00
Ricky Tang
cfae2a37be rtspsrc: Fix push-backchannel-buffer parameter mismatch
When using python, signal parameter must match with function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/724>
2020-09-11 18:33:04 +08:00
George Kiagiadakis
d997a8d48b rtspsrc: drop stream-start message posted by the internal udp sink(s)
See #1368

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/685>
2020-07-29 14:06:55 +03:00
Ognyan Tonchev
adb044c9ed rtspsrc: Fix segfault with illegal free
set_get_param_q is not a pointer so it is illegal to call g_queue_free_full().
Freeing the requests by popping them from the queue instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/669>
2020-07-15 13:19:38 +00:00
Seungha Yang
8b4f18d53b rtspsrc: Don't return TRUE for unhandled query
Expected return value for unhandled query is FALSE

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/629>
2020-06-16 19:35:30 +09:00
Mathieu Duponchelle
f63299ff2f plugins: uddate gst_type_mark_as_plugin_api() calls 2020-06-06 00:42:25 +02:00
Mathieu Duponchelle
37c619f995 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types 2020-06-03 22:44:09 -04:00
Sebastian Dröge
b94b9988fa rtspsrc: Use the correct type for storing the max-rtcp-rtp-time-diff property
It's an integer property and rtpbin also expects an integer. Passing it
as a GstClockTime (guint64) to g_object_set() will cause problems, and
on big endian MIPS apparently causes crashes.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/737

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/605>
2020-05-27 22:33:31 +03:00
Thibault Saunier
3fdae346ca rtspsrc: Error out when failling to receive message response
And let it rety twice.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/717

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/601>
2020-05-25 20:13:06 -04:00
Thibault Saunier
fd7ecac793 rtspsrc: Properly set segments seqnums after seeks 2020-04-09 14:03:04 -04:00
Thibault Saunier
00539e1277 rtspsrc: Avoid stack overflow recursing waiting for response
Instead of recursing, simply implement a loop with gotos, the same
way it was done before 8121752887

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/710
2020-04-08 09:49:49 -04:00
Stefano Buora
2d3dccdba7 rtspsrc: remove useless function calls
Comparing gst_rtspsrc_loop_interleaved and gst_rtspsrc_loop_udp, and investigating on timeout issues, it sounds like a piece of code has been originally copied from udp to the interleaved one. The timeout variable is never used inside the interleaved one. No side effect has been seen in the removed function calls.

The debug message removed is pointless as the timeout used is "src->tcp_timeout" that is fixed.

The presence of the two timeout drove my team in investigating if the reference to the tcp_timeout was correct (it is). Hence we removed the misleading reference to the local timeout variable.
2020-02-20 08:27:35 +00:00
Mathieu Duponchelle
1471100f37 rtspsrc: fix requested range
When the server replies with a range "now-", it is presumed to
be a "live" stream and we should request a similar range.

This was the case prior to my refactoring to make use of
gst_rtsp_range_to_string in 5f1a732bc7,
this commit restores the behaviour for that case.
2020-02-12 05:47:54 +00:00
Alexander Lapajne
54c4ba82f8 rtspsrc: Fix for segmentation fault when handling set/get_parameter requests
gstrtspsrc uses a queue, set_get_param_q, to store set param and get
param requests. The requests are put on the queue by calling
get_parameters() and set_parameter(). A thread which executs in
gst_rtspsrc_thread() then pops requests from the queue and processes
them. The crash occured because the queue became empty and a NULL
request object was then used. The reason that the queue became empty
is that it was popped even when the thread was NOT processing a get
parameter or set parameter command. The fix is to make sure that the
queue is ONLY popped when the command being processed is a set
parameter or get parameter command.
2020-02-10 09:43:17 +01:00
Aaron Boxer
09d4514814 rtspsrc: improved handling of control concatenation with base
Also, `control_url` variable has been renamed to `control_path`,
as it is actually a path.
2019-12-30 16:52:45 +00:00
Aaron Boxer
ed6b5a3a63 rtspsrc: append aggregate control string to base URL before query string
Appending control string to end of query changes meaning of query string
Fixes #650
2019-12-30 16:52:45 +00:00
Aaron Boxer
4155c59cc4 rtspsrc: avoid seek DISCONT when only rate changes in same direction
Not setting DISCONT avoids a noticable delay when seeking
with only rate changing, in the same direction as current
rate.
2019-12-19 05:54:38 +00:00
Olivier Crête
9db1d740e8 rtspsrc: Remove deprecated GTimeVal
GTimeVal won't work past 2038
2019-12-18 19:48:34 +00:00
Joakim Johansson
4d7d577496 gstrtspsrc: Add missing lock on free set_get_param_q
Otherwise is it possible to get a crash in gst_rtspsrc_set_parameter.
2019-12-16 13:13:00 +01:00
Niels De Graef
7cf4ab6229 Don't pass default GLib marshallers for signals
By passing `NULL` to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.

Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
2019-11-17 15:32:30 +00:00
Tim-Philipp Müller
c9a47c0c8d Remove autotools build system 2019-10-14 11:04:18 +01:00
Aaron Boxer
46989dca96 documentation: fix a number of typos 2019-10-05 22:38:11 +00:00
Mathieu Duponchelle
c5e8a8f320 rtspsrc: fix git diff indentation 2019-09-02 16:33:05 +02:00
Mathieu Duponchelle
3bc5d3d3b5 rtspsrc: normalize variable to boolean 2019-08-30 22:42:58 +02:00
Mathieu Duponchelle
37eca8a12c rtspsrc: clip output segment on accurate seeks
The output segment is only used in ONVIF mode.

The previous behaviour was to output a segment computed from
the Range response sent by the server.

In ONVIF mode, servers will start serving from the appropriate
synchronization point (keyframe), and the Range in response will
start at that position.

This means rtspsrc can now perform truly accurate seeks in that
mode, by clipping the output segment to the values requested in
the seek. The decoder will then discard out of segment buffers
and playback will start without artefacts at the exact requested
position, similar to the behaviour of a demuxer when an accurate
seek is requested.
2019-08-30 14:50:21 +00:00
Mathieu Duponchelle
5c7423d73c rtspsrc: expose and implement is-live property
This is useful to support the ONVIF case: when is-live is set to
FALSE and onvif-rate-control is no, the client can control the
rate of delivery and arrange for the server to block and still
keep sending when unblocked, without requiring back and forth
PAUSE / PLAY requests. This enables, amongst other things, fast
frame stepping on the client side.

When is-live is FALSE, we don't use a manager at all. This case
was actually already pretty well handled by the current code. The
standard manager, rtpbin, is simply no longer needed in this case.

Applications can instantiate a downloadbuffer after rtspsrc if
needed.
2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
75f53631e5 rtspsrc: reset_time when flush stopping 2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
5f1a732bc7 rtspsrc: expose and implement onvif-mode property
Refactor the code for parsing and generating the Range, taking
advantage of existing API in GstRtspTimeRange.

Only use the TCP protocol in that mode, as per the specification.

Generate an accurate segment when in that mode, and signal to the
depayloader that it should not generate its own segment, through
the "onvif-mode" field in the caps, see
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/merge_requests/328>
for more information.

Translate trickmode seek flags to their ONVIF representation

Expose an onvif-rate-control property
2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
544f8fecf4 rtspsrc: improve handling of rate in seeks 2019-08-06 22:45:37 +00:00
Sebastian Dröge
86ec5c1031 rtspsrc: Use new GstRTSPMessage API to set message body from a buffer directly 2019-08-05 19:35:36 +03:00
Mathieu Duponchelle
33277da781 rtspsrc: unref the event in element seek handler 2019-07-01 13:54:13 +02:00
Mathieu Duponchelle
bcd367b81d rtspsrc: handle seek event on the element
Without this, the user has to wait for rtspsrc to have sent a PLAY
request and exposed its pads before seeking it.
2019-06-29 00:25:26 +02:00
Nicolas Dufresne
301a46bd2d rtspsrc: Remove uneeded keep-alive hack
The rtsp connection code has been fixed now.

https://bugzilla.gnome.org/show_bug.cgi?id=744209
2019-05-27 16:04:23 +02:00
Mathieu Duponchelle
d704790519 doc: fix element section documentations
Element sections were not rendered anymore after the hotdoc
port, fixing this revealed a few incorrect links.
2019-05-25 16:57:31 +02:00
Thibault Saunier
af01988534 doc: Port documentation to hotdoc 2019-05-13 11:34:56 -04:00
Thibault Saunier
0a6a62aa76 docs: Port all docstring to gtk-doc markdown 2019-05-13 10:24:40 -04:00
Thiago Santos
135e12565b rtspsrc: do not try to send EOS with invalid seqnum
The second udpsrc (rtcp) might not have seen the segment event if it was
not enabled or if rtcp is not available on the server. So if the
application tries to send an EOS event it will try to set an invalid
seqnum to the event.
2019-05-02 22:14:35 -07:00
Sebastian Dröge
a676c17259 rtspsrc: Don't remove udpsrc/sink from rtspsrc if they were not added to it
This can happen in various error cases that could happen between the
creation of the element in question and the adding to the rtspsrc.

It causes an ugly critical warning right now but is otherwise harmless.
2019-03-15 08:21:11 +00:00
Antonio Ospite
2dfe228740 docs: fix typos s/recieve/receive/ 2019-03-07 12:41:40 +01:00
Mathieu Duponchelle
6ed7ddebf9 rtspsrc: use the correct segment seqnum 2019-02-04 13:14:37 +00:00