Commit graph

1117 commits

Author SHA1 Message Date
Göran Jönsson
d1d404912e rtsp-stream: Not wait on receiver streams when pre-rolling
Without this patch there are problem pre-rolling when using audio back
channel.

Without this patch a probe will be created for all streams including
the stream for audio backchannel. To pre-roll all this pads have to
receive data. Since the stream for audio backchannel is a receiver this
will never happen.

The solution is to never create any probes for streams that are for
incomming data and instead set them as blocking already from beginning.
2019-06-28 13:31:34 +02:00
Tim-Philipp Müller
7b2adb015d onvif-media: fix "void function returning a value" compiler warning 2019-06-25 13:19:44 +01:00
Mathieu Duponchelle
ab37286300 rtsp-media: make sure streams are blocked when sending seek
The recent ONVIF work exposed a race condition when dealing with
multiple streams: one of the sinks may preroll before other streams
have started flushing. This led to the pipeline posting async-done
prematurely, when some streams were actually still in the middle
of performing a flushing seek. The newly-added code looks up a
sticky segment event on the first stream in order to respond to
the PLAY request with accurate Scale and Speed headers. In the
failure condition, the first stream was flushing, and thus had
no sticky segment event, leading to the PLAY request failing,
and in turn the test.
2019-06-12 22:19:27 +02:00
Michael Bunk
b545c10e8f Fix typos 2019-06-07 13:42:24 +02:00
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
Mathieu Duponchelle
7640cb8f21 rtsp-client: add gst_rtsp_client_get_stream_transport()
This will be used in the onvif tests in order to validate the
data transmitted over TCP: for streaming to continue after a
data message has been provided to client->send_func, the client
is responsible for marking the message as sent on the relevant
stream transport.
2019-06-06 18:45:17 +02:00
Mathieu Duponchelle
52d20df0f4 client: Scale implies TRICK_MODE 2019-06-04 14:32:51 +02:00
Mathieu Duponchelle
b9cc3e867a client: compare booleans, not pointers to them 2019-06-04 14:32:51 +02:00
Nikita Bobkov
f31f79f60e Reverse playback support
GStreamer plays segment from stop to start when doing reverse playback.
RTSP implies that media should be played from start of Range header to
its stop. Hence we swap start and stop times before passing them to
gst_element_seek.

Also make gst_rtsp_stream_query_stop always return value that can be
used as stop time of Range header.
2019-06-04 14:32:51 +02:00
Branko Subasic
bc74589601 rtsp-client: add support for Scale and Speed header
Add support for the RTSP Scale and Speed headers by setting the rate in
the seek to (scale*speed). We then check the resulting segment for rate
and applied rate, and use them as values for the Speed and Scale headers
respectively.

https://bugzilla.gnome.org/show_bug.cgi?id=754575
2019-06-04 14:32:51 +02:00
Branko Subasic
65d9aa327c rtsp-client: allow sub classes to adjust the seek
Adds a new virtual function, adjust_play_mode(), that allows
sub classes to adjust the seek done on the media. The sub class can
modify the values of the the seek flags and the rate.

https://bugzilla.gnome.org/show_bug.cgi?id=754575
2019-06-04 14:32:51 +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
Mathieu Duponchelle
da22533780 doc: remove xml from comments 2019-05-29 23:20:09 +02:00
Thibault Saunier
f4b20b010d doc: Fix some docstrings 2019-05-13 17:00:00 -04:00
Thibault Saunier
5b039416db docs: Port to hotdoc 2019-05-13 11:38:39 -04:00
Sebastian Dröge
abf6be1d7a rtsp-server: Fix various Since markers 2019-04-23 15:09:34 +03:00
Sebastian Dröge
8d3bef4c1e rtsp-server: Add various Since: 1.14 markers 2019-04-23 15:01:32 +03:00
Sebastian Dröge
802a648723 rtsp-server: Add various missing Since: 1.16 markers 2019-04-23 14:56:42 +03:00
Sebastian Dröge
640c5bba41 rtspclientsink: Set async-handling=false for the internal bins
Without this we can easily run into a race condition with async state changes:
- the pipeline is doing an async state change
- we set the internal bins to PLAYING but that's ignored because an
  async state change is currently pending
- the async state change finishes but does not change the state of the
  internal bins because of locked_state==TRUE
- the internal bins stay in PAUSED forever
2019-04-22 10:38:08 +00:00
Sebastian Dröge
c4653ef38b rtspclientsink: Use write_messages() API to send buffer lists in one go
And to write messages with multiple memories also via writev().
2019-04-22 10:38:08 +00:00
Kristofer Bjorkstrom
b578628dc1 rtsp-client: Handle Content-Length limitation
Add functionality to limit the Content-Length.
API addition, Enhancement.

Define an appropriate request size limit and reject requests
exceeding the limit with response status 413 Request Entity Too Large

Related to !182
2019-04-22 09:17:13 +00:00
Sebastian Dröge
0f0a10525a rtspclientsink: Notify the stream transport about each written message
Otherwise it will never try to send us the next one: it tries to keep
exactly one message in-flight all the time.

In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
in the client sink we always write data out synchronously.
2019-04-15 17:34:56 +00:00
Göran Jönsson
3cfe88632f rtsp_server: Free thread pool before clean transport cache
If not waiting for free thread pool before clean transport caches, there
can be a crash if a thread is executing in transport list loop in
function send_tcp_message.

Also add a check if priv->send_pool in on_message_sent to avoid that a
new thread is pushed during wait of free thread pool. This is possible
since when waiting for free thread pool mutex have to be unlocked.
2019-04-11 08:02:52 +02:00
Ulf Olsson
d09b7c8e4f rtsp-stream: Add support for GCM (RFC 7714)
Follow-up to !198
2019-04-10 08:43:29 +00:00
Erlend Eriksen
a9d3bcc03e session pool: fix missing klass-> in klass->create_session 2019-03-28 00:27:37 +01:00
Göran Jönsson
1fd49d36d1 rtsp-media: Handle set state when preparing.
Handle the situation when  a call to gst_rtsp_media_set_state is done
when media status is preparing.

Also add unit test for this scenario.

The unit test simulate on a media level when two clients share a (live)
media.
Both clients have done SETUP and got responses. Now client 1 is doing
play and client 2 is just closing the connection.

Then without patch there are a problem when
client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
And client2 is doing closing connection we can end up in a call
to gst_rtsp_media_set_state when
priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
shut down media is jumped over .

With this patch and this scenario we wait until
priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
execute after that and now we will execute the logic for
shut down media.
2019-03-20 12:26:50 +01:00
Göran Jönsson
7e01dfd151 rtsp-media: Fix multicast use case with common media
Use case
client 1: SETUP
client 1: PLAY
client 2: SETUP
client 1: TEARDOWN
client 2: PLAY
client 2: TEARDOWN
2019-02-19 12:12:34 +01:00
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
Sebastian Dröge
4be7424de5 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
And route all messages through the send_func if no send_messages_func
was provided.

We otherwise break backwards compatibility.
2019-01-30 14:40:09 +02:00
Sebastian Dröge
c372643e1e rtsp-client: Add support for sending buffer lists directly
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2019-01-30 14:40:09 +02:00
Sebastian Dröge
d708f9736b rtsp-server: Add support for buffer lists
This adds new functions for passing buffer lists through the different
layers without breaking API/ABI, and enables the appsink to actually
provide buffer lists.

This should already reduce CPU usage and potentially context switches a
bit by passing a whole buffer list from the appsink instead of
individual buffers. As a next step it would be necessary to
  a) Add support for a vector of data for the GstRTSPMessage body
  b) Add support for sending multiple messages at once to the
    GstRTSPWatch and let it be handled internally
  c) Adding API to GOutputStream that works like writev()

Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2019-01-30 14:39:50 +02:00
Benjamin Berg
621e140a8e client: Fix crash in close handler
The close handler could trigger a crash because it invalidated the
watch_context while still leaving a source attached to it which would be
cleaned up at a later point.
2019-01-29 18:34:08 +00:00
Edward Hervey
a48711fa9d rtsp-stream: Use cached address when allocating sockets
If an address/port was previously decided upon (ex: multicast in the
SDP), then use that instead of re-creating another one

Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2019-01-29 14:42:35 +01:00
Lars Wiréen
ae32203cb0 rtsp-media: Fix race codition in finish_unprepare
The previous fix for race condition around finish_unprepare where the
function could be called twice assumed that the status wouldn't change
during execution of the function. This assumption is incorrect as the
state may change, for example if an error message arrives from the
pipeline bus.

Instead a flag keeping track on whether the finish_unprepare function
is currently executing is introduced and checked.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2019-01-25 12:44:23 +00:00
Patricia Muscalu
3be1b9bba8 Add source elements to the pipeline before activation
In plug_src we changed the element state before adding it to
the owner container. This prevented the pipeline from intercepting
a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
to assign a custom task pool.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2018-12-06 08:59:04 +00:00
Patricia Muscalu
4370f3a901 rtsp-media: Update priv->blocked when linked streams are unblocked.
Media is considered to be blocked when all streams that belong to
that media are blocked.
This patch solves the problem of inconsistent updates of
priv->blocked that are not synchronized with the media state.
2018-11-19 11:35:26 +00:00
Patricia Muscalu
146b3da174 rtsp-media: Don't block streams before seeking
Before the seek operation is performed on media, it's required that
its pipeline is prepared <=> the pipeline is in the PAUSED state.
At this stage, all transport parts (transport sinks) have been successfully
added to the pipeline and there is no need for blocking the streams.
2018-11-19 11:35:26 +00:00
Linus Svensson
185385924d rtsp-stream: Use seqnum-offset for rtpinfo
The sequence number in the rtpinfo is supposed to be the first RTP
sequence number. The "seqnum" property on a payloader is supposed to be
the number from the last processed RTP packet. The sequence number for
payloaders that inherit gstrtpbasepayload will not be correct in case of
buffer lists. In order to fix the seqnum property on the payloaders
gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
"seqnum-offset" from the "stats" property contains the value of the
very first RTP packet in a stream. The server will, however, try to look
at the last simple in the sink element and only use properties on the
payloader in case there no sink elements yet, and by looking at the last
sample of the sink gives the server full control of which RTP packet it
looks at. If the payloader does not have the "stats" property, "seqnum"
is still used since "seqnum-offset" is only present in as part of
"stats" and this is still an issue not solved with this patch.

Needed for gst-plugins-base!17
2018-11-14 12:29:58 +00:00
Linus Svensson
1c4d3b36fb rtsp-stream: Plug memory leak
Attaching a GSource to a context will increase the refcount. The idle
source will never be free'd since the initial reference is never
dropped.
2018-11-14 12:29:58 +00:00
Mathieu Duponchelle
f43df76ee7 meson: add new onvif types 2018-11-01 14:20:36 +01:00
Sebastian Dröge
961de06be2 Add ONVIF subclass headers to the installed headers in meson.build too 2018-11-01 12:49:51 +02:00
Sebastian Dröge
5392cbba8a rtsp-server: Declare GstRTSPServer struct before anything else
It's needed by all kinds of other headers, including the ones that are
required for defining the GstRTSPServer struct itself and its API.
2018-11-01 11:29:01 +02:00
Sebastian Dröge
d525000f6a Mark all ONVIF-specific subclasses as Since 1.14 2018-11-01 10:23:02 +02:00
Sebastian Dröge
84a7f459a1 Include ONVIF types from single-include rtsp-server.h
... by actually making it a single-include header and moving everything
related to the GstRTSPServer type to rtsp-server-object.h instead.
Otherwise there are too many circular includes.

https://bugzilla.gnome.org/show_bug.cgi?id=797361
2018-11-01 10:18:22 +02:00
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
Edward Hervey
ebafccb65a rtsp-client: Remove timeout GSource on cleanup
Avoids ending up with races where a timeout would still be around
*after* a client was gone. This could happen rather easily in
RTSP-over-HTTP mode on a local connection, where each RTSP message
would be sent as a different HTTP connection with the same tunnelid.

If not properly removed, that timeout would then try to free again
a client (and its contents).
2018-10-22 09:35:54 +02:00
Tim-Philipp Müller
22ced50da5 autotools: fix distcheck 2018-10-04 14:31:59 +01:00
Ognyan Tonchev
7b5e232a9e onvif: encapsulate onvif part into a bin
...and thus do not let onvif affect pipelines latency

https://bugzilla.gnome.org/show_bug.cgi?id=797174
2018-10-03 13:26:36 +03:00
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
Tim-Philipp Müller
62d4c0b179 libs: fix API export/import and 'inconsistent linkage' on MSVC
Export rtsp-server library API in headers when we're building the
library itself, otherwise import the API from the headers.

This fixes linker warnings on Windows when building with MSVC.

Fix up some missing config.h includes when building the lib which
is needed to get the export api define from config.h

https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-24 09:36:21 +01:00