Commit graph

6839 commits

Author SHA1 Message Date
Wim Taymans
1f0600ee6f Revert "rtph264pay: Restructuring to allow for adding optional caps"
This reverts commit 61666898cf.

This commit changes what the set_sps_pps() function does, not it doesn't
set caps anymore (and should have been renamed). The main problem is that
not all call sites are updated and thus leak the string.
2013-05-31 15:18:48 +02:00
Wim Taymans
1516c14881 Revert "rtph264pay/depay: Add frame dimensions a payloaded caps"
This reverts commit 3dca756a5d.

The H264 RTP spec has no attributes for width and height.
2013-05-31 15:11:12 +02:00
Wim Taymans
b79d217396 Revert "rtph264pay/depay: Add optional framerate caps for use in SDP"
This reverts commit d8825e2a5c.

There is no framerate attribute in the h264 RTP spec.
2013-05-31 15:09:51 +02:00
Wim Taymans
190b3d6688 Revert "rtpjpegpay/depay: Replace framesize caps with width/height"
This reverts commit 0075d111b4.

Extra application/x-rtp are SDP fields, which are strings.
2013-05-31 15:08:16 +02:00
Wim Taymans
f870cef8bc Revert "rtpjpegpay/depay: Replace framerate caps field with fraction"
This reverts commit 9fd25a810b.

We deal with sdp attributes in application/sdp, which are always strings.
2013-05-31 15:05:51 +02:00
Wim Taymans
25082a50b9 rtspsrc: add extra TLS url protocols
We also support TLS protocols now.
2013-05-31 12:34:22 +02:00
Sebastian Dröge
e2e1d1a158 videomixer: Add FIXME comment about the DURATION query from adder
Currently the code just takes with maximum upstream duration, which
is wrong. It should be the maximum upstream duration in running time.
2013-05-30 23:56:38 +02:00
Mathieu Duponchelle
5223868caa videomixer: Set a reference to mix->current_caps as the QUERY_CAPS result. 2013-05-30 15:36:48 -04:00
Stefan Sauer
6feaf69bec level: misc cleanups
Fix some oudated comments. Sort out some confusion of interval_frames and num_frames.
2013-05-30 17:38:55 +02:00
Stefan Sauer
52282b5faa level: fix discontinuities in timestamps 2013-05-28 19:09:12 +02:00
Wim Taymans
80850df711 rtspsrc: create and push stream-start in TCP mode 2013-05-28 15:45:49 +02:00
Wim Taymans
4fc1f3088b rtspsrc: remove some obsolete code
It is not needed to do a state change from the _play() function on
ourselves. The state change function already did that and we don't want to
interfere with that (or use hacks to avoid interference).
2013-05-28 15:10:07 +02:00
Wim Taymans
e6f850996b rtspsrc: set RTCP caps on the RTCP pads 2013-05-28 12:26:25 +02:00
Wim Taymans
63f0ecbbe7 rtpsession: send stream-start and segment events
Also send stream-start and segment event on the RTCP pad.
We don't need to send anything on the sync_src pad because we
already forwarded all incomming events.
2013-05-28 12:26:25 +02:00
Wim Taymans
779bcc093c rtspsrc: add signal to handle server requests
Add a signal to be notified of a server request. The signal handler can then
construct the response message for the server.

See https://bugzilla.gnome.org/show_bug.cgi?id=632207
2013-05-28 12:26:24 +02:00
Nicolas Dufresne
cd30a81ee3 videomixer: Maintain z-order when new pad are added
https://bugzilla.gnome.org/show_bug.cgi?id=701109
2013-05-27 22:43:25 -04:00
Thibault Saunier
7a3df1ab31 videomixer: Always handle flush_stop_pending atomically
It is not protected with the COLLECT_PADS_STREAM_LOCK anymore
2013-05-25 12:20:08 -04:00
Thibault Saunier
608bd3e2db videomixer: Do not take COLLECT_PADS_STREAM_LOCK when unnecessary
Collectpad takes the lock itself when receiving serialized events
and we should not take it for not serialized ones
2013-05-25 11:03:31 -04:00
Sebastian Dröge
1b5a8ac41c flxdec: Properly skip non-frame chunks 2013-05-24 19:34:05 +02:00
Sebastian Dröge
ae3ee32f42 flxdec: Flush data from adapter after reading it
Otherwise we're going in an infinite loop, reading the same data
over and over again.
2013-05-24 19:31:14 +02:00
Andoni Morales Alastruey
a62af107ae goom2k1: fix more duplicated symbols 2013-05-24 09:29:23 +02:00
Sebastian Rasmussen
9fd25a810b rtpjpegpay/depay: Replace framerate caps field with fraction
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
2013-05-23 21:05:49 +02:00
Sebastian Rasmussen
0075d111b4 rtpjpegpay/depay: Replace framesize caps with width/height
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.

Keep parsing a-framerate, x-framerate and x-dimensions in rtpjpegdepay
to be backwards compatible with previous payloaders.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
2013-05-23 21:05:43 +02:00
Sebastian Rasmussen
d8825e2a5c rtph264pay/depay: Add optional framerate caps for use in SDP
This allows for applications to format SDP attributes and still do SDP
offer/answer based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
2013-05-23 21:04:17 +02:00
Sebastian Rasmussen
3dca756a5d rtph264pay/depay: Add frame dimensions a payloaded caps
This allows for applications to format SDP attributes and still do SDP
offer/answer based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
2013-05-23 21:04:11 +02:00
Sebastian Rasmussen
61666898cf rtph264pay: Restructuring to allow for adding optional caps
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
2013-05-23 21:04:00 +02:00
Sebastian Dröge
e26b8c2832 (dyn|multi)udpsink: Add properties to specify the bind address and port
By default we use the any addresses and a random port for binding the socket.
2013-05-23 18:42:09 +02:00
Sebastian Dröge
5b79b8ff3c (dyn|multi)udpsink: Bind socket before using it
https://bugzilla.gnome.org/show_bug.cgi?id=700878
2013-05-23 18:05:07 +02:00
Sebastian Dröge
1ed7f7a6a8 (multi)udpsink: Add missing getters for socket-v6 and used-socket-v6 properties 2013-05-23 17:26:31 +02:00
Nicolas Dufresne
d8c5e31657 videomixer: Don't hold stream-lock while pushing non-serialized events
https://bugzilla.gnome.org/show_bug.cgi?id=700868
2013-05-23 09:20:04 -04:00
Nicolas Dufresne
a7e0f251ca videomixer: Don't hold object lock while sending events
https://bugzilla.gnome.org/show_bug.cgi?id=700868
2013-05-23 09:20:04 -04:00
Sebastian Dröge
ecc6c607ff deinterlace: The return value of gst_pad_set_caps() is not relevant anymore
Caps can fail to be set because the pad is not linked yet for example.
2013-05-22 17:34:07 +02:00
David Schleef
318cd39c3e qtdemux: Add error if file has playready drm 2013-05-21 18:21:49 -07:00
Thibault Saunier
18ef4f18d0 videomixer: Send a reconfigure event upstream if sinkpad caps are not usable
https://bugzilla.gnome.org/show_bug.cgi?id=684237
2013-05-21 12:15:36 -04:00
Alexander Schrab
a1df050356 mulawdec: Handle NULL buffers in handle_frame
https://bugzilla.gnome.org/show_bug.cgi?id=698894
2013-05-21 15:18:04 +02:00
Sebastian Rasmussen
2361567bae rtpjpegpay/depay: Add framesize caps for use in SDP
The format of the value adheres to RFC6064 and it is meant to be parsed
and included in the SDP sent by gst-rtsp-server to its clients.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
2013-05-21 09:09:03 +02:00
Sebastian Rasmussen
919eed0787 rtpjpegpay: Add optional framerate caps for use in SDP
The format of the value adheres to RFC4566 and it is meant to be parsed
and included in the SDP sent by gst-rtsp-server to its clients.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
2013-05-21 09:08:21 +02:00
Mathieu Duponchelle
2d3910fc79 videomixer: When all sinkpads are eos, update output segment stop and forward it
https://bugzilla.gnome.org/show_bug.cgi?id=699793
2013-05-20 21:06:56 +02:00
Mathieu Duponchelle
521c9a7b5d videomixer: Don't reset the output segment on flush stop
Only init it when getting from READY to PAUSED, and change it on seek events.

https://bugzilla.gnome.org/show_bug.cgi?id=699793
2013-05-20 21:03:03 +02:00
Thibault Saunier
86b106091c videomixer: Send caps event from the streaming thread
This way we avoid races in caps negotiation and we make sure
that the caps are sent after stream-start.

https://bugzilla.gnome.org/show_bug.cgi?id=684237
2013-05-19 09:28:04 -04:00
Thibault Saunier
718f9004d0 videomixer: Do not send flush_stop when receiving a seek
There is no reason to send a flush-stop when receiving a seek event.
In the case of a flushing seek, we could eventually want to, but in
the code path were we check if the seek is "flushing", we have the
following comment that makes sense:

"we can't send FLUSH_STOP here since upstream could start pushing data
after we unlock mix->collect.
We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
forwarding the seek upstream or from gst_videomixer_collected,
whichever happens first."

https://bugzilla.gnome.org/show_bug.cgi?id=684237
2013-05-19 09:28:04 -04:00
Thibault Saunier
85b6852deb videomixer2: Protect flush_stop_pending with the collectpad stream lock
And make sure to expect a flush-stop after a flush-start

https://bugzilla.gnome.org/show_bug.cgi?id=684237
2013-05-19 09:28:04 -04:00
Michael Olbrich
d1c56376d6 rtpmp4apay: clear config buffer before using it
This is necessary because parts of the memory are only modified with "|="

https://bugzilla.gnome.org/show_bug.cgi?id=700514
2013-05-18 10:57:56 +01:00
Thiago Santos
55caa99ccd qtdemux: Do not expect EOS after a segment event if upstream is mss
In case qtdemux is handling a mss stream, do not mark the stream to wait
for EOS after a segment. Even if it seems to be the last one according to
the current streams information.

MSS handling is different here because there is another demuxer driving
the pipeline
2013-05-16 16:50:49 -03:00
Thiago Santos
5517e352ab qtdemux: only set channels and rate if qtdemux knows it
Setting both of those to 0 is pointless and means that qtdemux
doesn't know the real value. Avoid setting it in this case.
2013-05-16 16:50:49 -03:00
Arnaud Vrac
6edcc564ba qtdemux: set alac caps using info from codec buffer
The samplerate field in the STSD atom is not right for some ALAC files
(usually when audio is 96kHz/24bits), so the audio caps must be
extracted from the codec data.

https://bugzilla.gnome.org/show_bug.cgi?id=700382
2013-05-15 18:42:11 +01:00
Arnaud Vrac
8ed611cdbc avidemux: do not push discont buffers if they aren't discont
https://bugzilla.gnome.org/show_bug.cgi?id=682110
2013-05-15 13:16:11 +01:00
Joshua M. Doe
837dcfb363 videocrop: Add support for GRAY16_LE/GRAY16_BE
https://bugzilla.gnome.org/show_bug.cgi?id=700331
2013-05-15 09:29:30 +02:00
Sebastian Dröge
41e1af3751 rgvolume: Send all events through the proxypads instead of just sending to the target
Otherwise the sticky events are missing on the proxypads.
2013-05-14 17:29:58 +02:00
Sebastian Dröge
4fdbf88a65 matroskaparse: Make sure to send a segment event before dataflow 2013-05-14 13:52:18 +02:00