Commit graph

801 commits

Author SHA1 Message Date
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
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
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
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 Rasmussen
94f3e18c5b Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
This reverts commit 935e8f852d.

RFC 2326 states that session IDs may consist of alphanumeric as well as
the safe characters $-_.+ -- N.B. the percent character is not allowed.

Previously the session ID was URI-escaped, this meant that any character
which was not alphanumeric or any of the characters +-._~ would be
percent encoded. While the RFC (surprisingly) mentions that linear white
space in session IDs should be URI-escaped, it does not say anything
about other characters. Moreover no white space is allowed in the
session ID. Finally the percent character which is the result of
URI-escaping is not allowed in a session ID.

So there is no reason to do any URI-escaping, and now it is removed.

https://bugzilla.gnome.org/show_bug.cgi?id=742869
2015-01-14 18:43:37 +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
Sebastian Dröge
a44b564f59 rtsp-stream: Fix some minor memory leaks 2014-12-16 16:46:15 +01:00
Sebastian Dröge
8ae3566591 rtsp-media: Some minor cleanup 2014-12-16 16:46:06 +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
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
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
ef9dc6c9e4 rtsp-client: mikey memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739383
2014-10-30 10:34:56 +00:00
Vincent Penquerc'h
f803be2dc8 rtsp-media: deactivate media when shutting down from paused
This was only done when going directly from playing.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2014-10-21 11:52:27 +02:00
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