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Author SHA1 Message Date
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
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
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
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
Anila Balavan
18668bf495 rtsp-stream: RTCP and RTP transport cache cookies seperated 
RTCP packets were not sent because the same tr_cache_cookie was used for
both RTP and RTCP. So only one of the tr_cache lists were populated
depending on which one was sent first. If the tr_cache list is not
populated then no packets can be sent. Most often this happened to be
RTCP. Now seperate RTCP and RTP transport cache cookies are added which
resulted in both the tr_cache_lists to be populated regardless of which
one was sent first.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
 2015-01-30 18:26:44 +01:00
Tim-Philipp Müller
6987a00fa9 rtsp-stream: fix false compiler warning 
rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
 2015-01-21 14:58:19 +00:00
Göran Jönsson
0d2de69db9 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each 
Fixes crash when two threads access handle_new_sample() at the same
time, one for RTP, one for RTCP.

Otherwise, when iterating over the transports cache, it might be modified by
another thread at the same time if the transports cookie has changed.

https://bugzilla.gnome.org/show_bug.cgi?id=742954
 2015-01-16 12:52:43 +01:00
Sebastian Dröge
fe8e877dd9 rtsp-stream: Set format=TIME on our app sources for TCP 2015-01-15 19:35:01 +01:00
Sebastian Dröge
a44b564f59 rtsp-stream: Fix some minor memory leaks 2014-12-16 16:46:15 +01:00
Sebastian Dröge
06bfc0697b rtsp-stream: Fix compiler warnings 
rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
  ^

rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
  ^
 2014-12-16 16:42:13 +01:00
Matthew Waters
4f40781fff media: implement ssrc-multiplexed retransmission support 
based off RFC 4588 and the server-rtpaux example in -good
 2014-12-16 16:41:08 +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
Aleix Conchillo Flaqué
7c267928ff rtsp-stream: unref srtp decoder when leaving bin 
https://bugzilla.gnome.org/show_bug.cgi?id=739481
 2014-11-01 11:26:14 +00:00
Aleix Conchillo Flaqué
966065a018 stream: release lock even not all transports have been removed 
We don't want to keep the lock even we return FALSE because not all the
transports have been removed. This could lead into a deadlock.

https://bugzilla.gnome.org/show_bug.cgi?id=737797
 2014-10-21 10:08:44 +02:00
Srimanta Panda
376488d8c7 rtsp-media: Make sure that sequence numbers are monotonic after pause 
The sequence number is not monotonic for RTP packets after pause. The
reason is basepayloader generates a randon sequence number when the
pipeline goes from ready to pause. With this fix generation of sequence
number will be monotonic when going from pause to play request.

https://bugzilla.gnome.org/show_bug.cgi?id=736017
 2014-09-12 17:29:30 +03:00
Sebastian Dröge
1b47b6d9b0 rtsp-stream: Remove the multicast group udp sources when removing from the bin 2014-08-25 10:39:04 +03:00
Sebastian Dröge
6ba5ca447f rtsp-media: Query position and stop time only on the RTP parts of the pipeline 
The RTCP parts, in specific the RTCP udpsinks, are not flushed when
seeking and will always continue counting the time. This leads to
the NPT after a backwards seek to be something completely different
to the actual seek position.

https://bugzilla.gnome.org/show_bug.cgi?id=732644
 2014-08-12 10:54:12 +03:00
Sebastian Dröge
3159b374b9 rtsp-stream: Listen on the multicast group for RTP/RTCP packets 
When a UDP multicast transport is used it is expected that the server listens
for RTP and RTCP packets on the multicast group with the corresponding port.
Without this we will never get RTCP packets from clients in multicast mode.

https://bugzilla.gnome.org/show_bug.cgi?id=732238
 2014-07-22 14:26:49 +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
Wim Taymans
db95746f6b stream: crypto can be NULL 2014-06-27 16:55:07 +02:00
Evan Nemerson
34e6ac3b9f introspection: add (nullable) annotations to return values 
https://bugzilla.gnome.org/show_bug.cgi?id=730952
 2014-06-26 19:08:16 +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
Aleix Conchillo Flaqué
17322693f6 stream: add signals for new RTP/RTCP encoders 
New signals to allow the user to configure the dynamically created
encoders.

https://bugzilla.gnome.org/show_bug.cgi?id=730228
 2014-05-16 16:27:52 +02:00
Wim Taymans
377ca6ed0f stream: add method to set crypto info 
Make a method to configure the crypto information of a stream.
Set udpsrc in READY instead of PAUSED so that we can configure caps
later.
 2014-04-03 17:26:12 +02:00
Wim Taymans
3d6175c745 stream: add SRTP support 
Install srtp encoder and decoder elements in rtpbin
Add MIKEY in SDP
 2014-03-25 10:31:21 +01:00
Sebastian Rasmussen
b1b5301577 gobject-introspection: Add annotations to support language bindings 
In addition a few cosmetic changes:

 * Adjust the order of arguments
 * Fix typo: occured -> occurred
 * Fix indentation after Return:-clauses

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
 2014-03-24 00:36:42 +00:00
Sebastian Rasmussen
0b617dd5bd rtsp-stream: Don't mix IPv4 and IPv6 addresses 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
 2014-03-15 15:44:25 +01:00
Wim Taymans
2c7ffe97ca stream: take caps after the session manager 
Take the caps for the SDP after they leave the rtpbin so that we can
also get the properties added by rtpbin elements.
 2014-03-13 14:27:15 +01:00
Wim Taymans
50ca10e751 stream: release lock while pushing out packets 
Keep a cache of the transports and use this to iterate the transport
while pushing packets. This allows us to release the lock early.

See https://bugzilla.gnome.org/show_bug.cgi?id=725898
 2014-03-13 14:20:17 +01:00
Wim Taymans
48b6b8e3e6 stream: release some locks in error cases 2014-03-03 12:17:48 +01:00
Sebastian Rasmussen
81a2928c89 docs: Enable and fix gtk-doc warnings 
* Makefile: Enable gtk-doc warnings, like the rest of GStreamer
 * addresspool/mediafactory: Add missing annotation colon
 * stream: Annotate return value

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
 2014-03-03 09:43:05 +01:00
Göran Jönsson
a7f0feff23 stream: set ttl-mc before adding the socket 
Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
never be set on socket.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
 2014-02-18 11:10:51 +01:00
Sebastian Dröge
957a4a65c6 rtsp-server: Fix lots of compiler warnings with clang 2014-02-09 10:45:28 +01:00
Sebastian Dröge
902b59f823 Revert "rtsp-server: support build against last stable release" 
This reverts commit 099a10f61f.

Let us require 1.2.3 now, which is going to be released in a few
minutes.
 2014-02-09 10:19:50 +01:00
Wim Taymans
9048d87ff4 stream: handle NULL seqnum and rtptime arguments 2014-02-04 16:28:00 +01:00
Wim Taymans
71c45fce5a stream: add fallback for missing stats property 
Use a fallback when the payloader does not have a stats property

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
 2014-02-04 10:14:45 +01:00
Wim Taymans
036f2760bf stream: don't leak stats structure 
Don't leak the stats structure and deal with NULL stats.
 2014-01-28 14:51:26 +01:00
Sebastian Rasmussen
7edaa6ca20 stream: Get rtpinfo properties atomically from payloader 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
 2014-01-27 15:19:30 +01:00
Tim-Philipp Müller
099a10f61f rtsp-server: support build against last stable release 
Until 1.2.3 is out with the new get_type function and we
can require that.
 2014-01-12 16:55:21 +00:00
Wim Taymans
d90ce618e4 stream: fix compilation 2014-01-07 15:28:05 +01:00
Wim Taymans
ae1fe21436 stream: add property to configure profiles 2014-01-07 12:39:58 +01:00
Wim Taymans
a1202effda stream: add method to check supported transport 
Add a method to check if a transport is supported
 2014-01-07 12:39:57 +01:00
Wim Taymans
8aaa432d58 stream: return clock-rate from get_rtpinfo 
And use it to correct the rtptime to the requested start-time.

See https://bugzilla.gnome.org/show_bug.cgi?id=712198
 2013-12-26 17:14:06 +01:00
Wim Taymans
cfdc7408b5 stream: also return the running-time 
Return the running-time in the rtpinfo as well.
 2013-12-26 16:29:39 +01:00
Edward Hervey
cdd72905af rtsp-stream: Check return value of sscanf 
streamid is only valid if sscanf matched something.
 2013-12-19 14:26:34 +01:00
Wim Taymans
bdef631218 stream: add API to block streams 
Add an API to block on the streams and make it post a message.

Based on patch by Ognyan Tonchev <ognyan@axis.com>

See https://bugzilla.gnome.org/show_bug.cgi?id=711257
 2013-11-28 16:18:39 +01:00
Aleix Conchillo Flaque
e5332535a7 rtsp-stream: add getter for payload type 
* gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.

* gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
  element and create the stream with this one instead of the dynpay%d
  element.

  https://bugzilla.gnome.org/show_bug.cgi?id=712396
 2013-11-22 11:19:35 +01:00
Sebastian Rasmussen
d1a2853659 rtsp-*: Fix type name typos in comments 
* rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
  * rtsp-auth: Refer to part of constant name as text
  * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
  * rtsp-session-media: Fix GstRTSPSessionMedia typo
  * rtsp-stream: Fix typo when refering to GstBin

https://bugzilla.gnome.org/show_bug.cgi?id=714988
 2013-11-22 09:13:08 +00:00
Tim-Philipp Müller
33c4bdfa01 rtsp-server: sprinkle some allow-none annotations for g-i 2013-11-18 10:47:04 +00:00
Wim Taymans
a106950f70 stream: add method to filter transports 
Add a method to safely iterate and collect the stream transports

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
 2013-11-18 11:18:15 +01:00
Wim Taymans
8d5ce0d4ee stream: Add functions to get rtp and rtcp sockets 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
 2013-11-12 12:04:55 +01:00
Sebastian Pölsterl
e756324490 Fixed several GIR warnings 2013-11-12 11:15:58 +01:00
Jonas Holmberg
fcf51d3485 stream: Correct control comparison 
https://bugzilla.gnome.org/show_bug.cgi?id=709176
 2013-10-02 11:57:06 +02:00
Patrick Radizi
7b0ad7c25f addresspool: return reason of failure 
Let gst_rtsp_address_pool_reserve_address() return the reason why
the address could not be reserved.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
 2013-09-24 17:30:18 +02:00
Wim Taymans
d74cbf2911 stream: optimize pipeline for protocols 
When TCP is not an allowed protocol for the stream, avoid creating the
appsrc/appsink/queue and tee elements.
 2013-08-16 17:05:24 +02:00
Wim Taymans
04d2da4d03 media-factory: allow all protocols 2013-08-16 16:19:27 +02:00
Wim Taymans
a84b71c0f0 stream: add protocols property 2013-08-16 16:08:43 +02:00
Wim Taymans
041b1b79a1 docs: improve docs 2013-07-16 12:32:51 +02:00
Wim Taymans
0b3644a21b docs: improve docs 2013-07-11 16:57:14 +02:00
Wim Taymans
ccceb1de11 docs: update docs 2013-07-11 12:18:26 +02:00
Wim Taymans
d4e8d800c9 stream: add method to check control url of stream 2013-07-03 15:13:45 +02:00
Wim Taymans
2ffb0f69d2 stream: add methods and property to set control string 2013-07-02 14:50:30 +02:00
Wim Taymans
a7fe63298c stream: add more support for IPv6 
Rename _get_address to _get_multicast_address in GstRTSPStream to
make it clear that this function only deals with multicast.
Make it possible to have both an IPv4 and IPv6 multicast address on
a stream. Give the client an IPv4 or IPv6 address depending on the
address it used to connect to the server.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
 2013-07-01 16:46:39 +02:00
Wim Taymans
82812988a6 stream: handle failed port allocation 
Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
can't allocate any family at all. Also keep track of what port families we
allocated.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
 2013-07-01 14:47:33 +02:00
Wim Taymans
284a0a5cd1 stream: improve docs 2013-07-01 12:20:50 +02:00
Aleix Conchillo Flaque
aeaadf0e5e stream: allow access to the rtp session 
https://bugzilla.gnome.org/show_bug.cgi?id=703004
 2013-06-24 23:42:58 +02:00
Alexander Schrab
c3f8673174 dscp qos support in gst-rtsp-stream 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
 2013-06-24 14:51:44 +02:00
Alexander Schrab
3e119be829 rtspstream: handle both ipv4 and ipv6 clients 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
 2013-06-03 11:23:40 +02:00
Wim Taymans
7b880231b1 stream: keep the transport object alive 
Keep the transport object alive while we have it as qdata on the
source.
 2013-05-30 06:49:20 +02:00
Wim Taymans
0ddd98bfa6 stream: set elements to NULL before removing 
When removing a stream, set the elements to NULL first. This avoids
element-is-not-in-NULL-state errors when we dispose the elements.
 2013-04-23 10:28:34 +02:00
Ognyan Tonchev
00291e5285 stream: add method to get the srcpad 2013-04-22 17:32:31 +02:00
Ognyan Tonchev
6081f91351 stream: clear session and caps for reuse 
Set the session and caps to NULL after unref otherwise we might unref
them again later.

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:09:22 +02:00
Olivier Crête
5a39e25949 stream: Select unicast address from pool if appropriate 2013-03-11 11:07:20 +01:00
Olivier Crête
a797cbde06 stream: Properties are always there in Gst 1.0 2013-03-11 11:07:20 +01:00
Olivier Crête
d06e68abd1 address-pool: Add unicast addresses 2013-03-11 11:07:20 +01:00
Olivier Crête
773c48e22f client: Check client provided addresses against the address pool 2013-03-11 11:07:19 +01:00
Wim Taymans
ad00c5e792 rtsp: make object details private 
Make all object details private
Add methods to access private bits
 2012-11-29 11:11:05 +01:00
Wim Taymans
d3d74ab77b stream: improve debug 2012-11-28 12:40:18 +01:00
Wim Taymans
1826844ee4 stream: set udp sources to PLAYING 
Set the UDP sources to PLAYING and locked state before we add it to the pipeline
so that it doesn't cause our pipeline to produce ASYNC-DONE.
 2012-11-20 12:24:13 +01:00
David Svensson Fors
0996266342 rtsp-stream: plug socket leak 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
 2012-11-20 09:26:28 +01:00
Wim Taymans
ba21661ce4 rtsp: improve debug 2012-11-15 16:15:20 +01:00
Wim Taymans
44a2855eb3 stream: add methods to deal with address pool 
Add methods to get and set the address pool for the stream
Add method to allocate and get the multicast addresses for this stream.
 2012-11-15 15:36:21 +01:00
Wim Taymans
6c2947e68b stream: share src and sink sockets 
the allocated socket is in the used-socket property, not socket.
 2012-11-15 13:52:07 +01:00
Wim Taymans
4753588b09 stream: add locking 2012-11-13 11:14:49 +01:00
Wim Taymans
c7d20e5603 stream-transport: add keep-alive method 2012-11-12 17:11:18 +01:00
Wim Taymans
75473fc88d stream-transport: add method to handle RTP/RTCP 
Call new methods instead of poking into the structures directly.
 2012-11-12 17:06:42 +01:00
Tim-Philipp Müller
4dba434f16 Fix FSF address 2012-11-04 00:14:25 +00:00
Wim Taymans
6a838fd5c8 stream: transports must already have been removed 2012-10-26 17:29:30 +02:00
Wim Taymans
6f7d755894 stream: improve join and leave of the pipeline 
simplify code
Do the cleanup properly
Add some docs
 2012-10-26 17:28:10 +02:00
Wim Taymans
6b7ff45ca6 rtsp: fix MTU setting 
Fix setting of the MTU. There is no need for a vmethod.
 2012-10-26 12:35:20 +02:00
Wim Taymans
de7c72dec2 rtsp: massive refactoring 
Make GObjects from the remaining simple structures.
Remove GstRTSPSessionStream, it's not needed.
Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
  a GstRTSPStream should be transported to a client.
Rename GstRTSPMediaFactory::get_element -> create_element because that
  more accurately describes what it does.
Make nice methods instead of poking in the structures.
Move some methods inside the relevant object source code.
Use GPtrArray to store objects instead of plain arrays, it is more
  natural and allows us to more easily clean up.
Move the allocation of udp ports to the Stream object. The Stream object
  contains the elements needed to stream the media to a client.
Improve the prepare and unprepare methods. Unprepare should now undo
  everything prepare did. Improve also async unprepare when doing EOS on
  shutdown. Make sure we always unprepare correctly.
 2012-10-25 21:29:58 +02:00