Commit graph

824 commits

Author SHA1 Message Date
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
Sebastian Dröge
1c30c60e64 rtsp-media: Mark some more functions static 2015-05-05 16:46:57 +02:00
Sebastian Dröge
bbdf0a47d1 rtsp-media: Only unblock the media in suspend() when actually changing the state
Otherwise we're going to lose a few packets for live streams during DESCRIBE.
2015-05-05 16:46:19 +02:00
Sebastian Dröge
ec2c500a9d rtsp-sdp: Only add RTX to the SDP when using a feedback profile 2015-05-04 16:31:20 +02:00
Hyunjun Ko
4ff22ef6d2 rtsp-stream: get valid clock-rate from last-sample
clock-rate in last-sample's caps is integer, not unsigned.
To get this value properly, variable needs to be type-casted to int.

https://bugzilla.gnome.org/show_bug.cgi?id=747614
2015-04-27 12:41:59 +02:00
Hyunjun Ko
de590b4b2a rtsp-stream: fix to get valid each stream data for request-aux-sender signal
Because of duplicated g_signal_connect for request-aux-sender signal,
wrong stream pointer is passed to the signal handler.
Instead of passing each stream, pass stream array and get the relevant stream.

https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-15 15:14:04 +02:00
Sebastian Dröge
ef3bfd757b rtsp-stream: Limit the queues to 1 buffer
We only need them to be able to pre-roll, queueing up more data here
is only going to harm latency and memory usage.
2015-03-23 21:04:43 +01:00
Sebastian Dröge
357af7aea6 rtsp-stream: Update comment and ASCII art to the latest code
We have a queue in front of the udpsink too to prevent the pipeline from
locking up.
2015-03-23 20:59:52 +01:00
Nicolas Dufresne
dfb053add3 rtsp-media: Properly return first rtptime
Instead we where returning first GstBuffer timestamp. This would result
in clock skew and unwanted behaviour in RTSP playback.

https://bugzilla.gnome.org/show_bug.cgi?id=746479
2015-03-21 11:04:05 -04:00
Nicolas Dufresne
01562286ba rtsp-stream: Don't leave buffer mapped
If the seq is NULL, the RTP buffer was left mapped. We should always
unmap the buffer.
2015-03-18 16:44:19 -04:00
Tim-Philipp Müller
896767b041 Fix double semicolons 2015-03-10 09:39:22 +00:00
Sebastian Dröge
852cc09f54 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
This gives more accurate values than asking the payloader. There might be
queueing happening between the payloader and the sink.

https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 16:00:38 +01:00
Sebastian Dröge
b58af93d83 rtsp-media: Don't seek for PLAY if the position will not change
https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 13:00:25 +01:00
Sebastian Dröge
93bdbb6acd rtsp-media: Don't include payload type in the caps for framesize
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335

Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
2015-03-09 10:21:49 +01:00
Linus Svensson
9dadaed2fd rtsp-sdp: add payload type to the sdp framesize attribute
The sdp framesize attribute is desribed in RFC6064. It is specified
for payloading of H263 and has the following form
a=framesize:<payload type> <width>-<height>. The <width>-<height> part
should be added to the caps in a payloader and the <payload type> should
be added by the rtsp-server.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
2015-03-09 09:26:38 +01:00
Jan Schmidt
db42945c2c rtsp-media-factory: Add functions to set/get the media gtype
Allow specifying the GType of a GstRtspMedia subclass to create
as a simpler way to get the factory to create a custom
GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
2015-03-03 11:53:16 +11:00
Gregor Boirie
bc7765eee7 rtsp-media: fix double unlock in _get_buffer_size()
Fixes an abort when calling gst_rtsp_media_get_buffer_size()
because of double g_mutex_unlock () usage.

https://bugzilla.gnome.org/show_bug.cgi?id=745434
2015-03-02 10:50:57 +00:00
Kent-Inge Ingesson
d2f1997c4b rtsp-session: Use monotonic time for RTSP session timeout
Changed RTSP session timeout handling to monotonic time
and deprecating the API for current system time.

This fixes timeouts when the system time changes.

https://bugzilla.gnome.org/show_bug.cgi?id=743346
2015-02-19 10:43:30 +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
Andreas Frisch
bac59c52f1 rtsp-stream: Add necessary queues between tee and multiudpsink
https://bugzilla.gnome.org/show_bug.cgi?id=744379
2015-02-13 11:28:43 +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
dc43f427a9 rtsp-stream: minor code formatting fix 2015-02-11 17:25:35 +00:00
Luis de Bethencourt
ec7bf5379e rtsp-media: fix logic for collect_streams
Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
all streams it knows if it got any, and can check if the transport mode is OK.

CID #1268400
2015-02-10 16:45:23 +00:00
Sebastian Dröge
8405cfad3a rtsp-media: Don't set the transport mode based on what elements we find
Just print a warning if the one that was set before disagrees with what
elements we found. It must already be set to something before as this
function is called after we received the SDP from ANNOUNCE in RECORD mode,
and we would reject ANNOUNCE if the RECORD flag was not set.
2015-02-09 10:21:50 +01:00
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
35b2b10cf4 rtsp-media: Expose latency setting for setting the rtpbin latency 2015-02-06 09:42:50 +01:00
Sebastian Dröge
844add610d rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer 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