Commit graph

110 commits

Author SHA1 Message Date
Tim-Philipp Müller
4a28e649c3 rtp: cache meta tag quarks and add more utility functions for metas
Every g_quark_from_static_string() is a hash table lookup serialised
on the global quark lock in GLib. Let's just look up the two quarks
we need once and cache them locally for future use. While we're at it,
add new utility functions for the two most commonly used tags
(audio + video). Make first argument a gpointer so we don't have to
cast and make the code ugly. These are used for logging purposes
only anyway.
2017-05-24 13:32:10 +01:00
Jan Schmidt
a1838927f7 rtph264pay: Intersect with filter caps in getcaps function.
Always intersect with the filter caps in the getcaps function
to make sure we return a subset of what was requested.

Other payloaders also have this problem and need fixing
in future commits.
2016-07-20 00:31:59 +10:00
Sebastian Dröge
5f2b32e642 rtph264pay: Deprecated sprop-parameter-set property
This is supposed to be either in the codec_data (avc stream format) or inside
the stream, and we extract it from there. It should not be set from a
property as it's stream specific.

https://bugzilla.gnome.org/show_bug.cgi?id=767789
2016-06-21 10:03:04 +03:00
Vineeth TM
1071309870 good: use new gst_element_class_add_static_pad_template()
https://bugzilla.gnome.org/show_bug.cgi?id=763076
2016-03-24 14:32:20 +02:00
Anton Bondarenko
453a618a9d rtph264pay: add "send SPS/PPS with every key frame" mode
It's not enough to have timeout or event based SPS/PPS information sent
in RTP packets. There are some scenarios when key frames may appear
more frequently than once a second, in which case the minimum timeout
for "config-interval" of 1 second for sending SPS/PPS is not sufficient.
It might also be desirable in general to make sure the SPS/PPS is
available with every keyframe (packet loss aside), so receivers can
actually pick up decoding immediately from the first keyframe if
SPS/PPS is not signaled out of band.

This patch adds the possibility to send SPS/PPS with every key frame. This
mode can be enabled by setting "config-interval" property to -1. In this
case the payloader will add SPS and PPS before every key (IDR) frame.

https://bugzilla.gnome.org/show_bug.cgi?id=757892
2015-11-27 13:30:07 +00:00
Tim-Philipp Müller
3026d1094b rtph264pay: change config-interval property type from uint to int
This way we can use -1 as special value, which is nicer than MAXUINT.
This is backwards compatible even with the GValue API, as shown by
a unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=757892
2015-11-27 12:48:09 +00:00
Sebastian Dröge
b1089fb520 rtp: Copy metadata in the (de)payloader, but only the relevant ones
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
2015-08-11 12:47:23 +02:00
Sebastian Dröge
978903cd87 rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING() 2015-07-01 11:58:26 +02:00
Luis de Bethencourt
671b4d25cd remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
2015-04-24 17:01:12 +01:00
Patrick Radizi
0a359cdbdc rtph264pay: fix potential crash when shutting down
A race condition in the state change function may cause buffers
to be unreffed while they are still used by the streaming thread
in gst_rtp_h264_pay_send_sps_pps() resulting in a crash. Chain
up to the parent class first in the state change function to
make sure streaming has stopped and only then free those buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=741381
2014-12-11 14:00:19 +00:00
Patrick Radizi
fef1a8d88a rtph264pay: Fixes buffer leak when using SPS/PPS
Fixes a buffer leak that would occurr if the pipeline was shutdown
while a SPS/PPS header was being created.

https://bugzilla.gnome.org/show_bug.cgi?id=741271
2014-12-09 09:47:23 +01:00
Srimanta Panda
421b00cd17 rtph264pay: append packetization mode parameter to SDP
Append packetization-mode parameter to SDP description.
Packetization mode signals the properties of an RTP payload type.

https://bugzilla.gnome.org/show_bug.cgi?id=733556
2014-08-08 13:41:36 +01:00
Mark Nauwelaerts
d5d28055c1 rtph264pay: unbreak au aligned byte-stream payloading 2014-08-03 14:42:45 +02:00
Srimanta Panda
dd9f716892 rtph264pay: append profile-level-id to SDP
Append profile-level-id to SDP if available.

https://bugzilla.gnome.org/show_bug.cgi?id=733539
2014-08-01 16:01:07 +01:00
Guillaume Desmottes
f00c2b7155 rtph264pay: propagate the GST_BUFFER_FLAG_DISCONT flag
Similarly to what we did with the DELTA_UNIT flag, this patch
propagates the DISCONT flag to the first RTP packet being used to transfer a
DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-19 12:22:49 +02:00
Guillaume Desmottes
4be99ec7d5 rtph264pay: propagate the GST_BUFFER_FLAG_DELTA_UNIT flag
Downstream elements may be interested knowing if a RTP packet is the start
of a key frame (to implement a RTP extension as defined in the
ONVIF Streaming Spec for example).

We do this by checking the GST_BUFFER_FLAG_DELTA_UNIT flag we receive from
upstream and propagate it to the *first* RTP packet outputted to transfer this
buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-19 12:22:38 +02:00
Tim-Philipp Müller
01ee993d8d rtph264pay: pre-allocate bufferlist of the right size
To avoid unnecessary re-allocs.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
c7c72c00b1 rtph264pay: push single buffer directly, no need to wrap it in a bufferlist
No point in a buffer list if we just have one single
buffer to push. Fix up unit test to handle that case
as well.
2014-06-18 14:54:58 +01:00
Göran Jönsson
80967c7638 gstrtph264pay: Reset sps pps variable when state change.
Reset last_spspps and sps/pps arrays  when state transition
GST_STATE_CHANGE_PAUSED_TO_READY.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726015
2014-04-21 12:07:20 +02:00
Göran Jönsson
53ffd9e1ca rtph264pay: only update last_spspps time if all sps/pps got sent successfully
This fixes an issue with gst-rtsp-server where no sps and pps are
sent for the first intra frame, because the payloader starts working
already when receiving DESCRIBE but there is no transports so it tries
to send sps and pps, but that fails with a FLUSHING flow. But the time
for last sent sps and pps would still be set, so when PLAY arrives and
the first intra frame is to be sent there is no sps and pps sent due to
that time since last sps pps is less than spspps_interval.

https://bugzilla.gnome.org/show_bug.cgi?id=724213
2014-02-25 10:48:24 +00:00
Jonas Holmberg
0ab0421759 rtph264pay: Map inbuffer once only
Do not call gst_buffer_extract() twice since each call will map and
unmap the biffer.

https://bugzilla.gnome.org/show_bug.cgi?id=719434
2013-11-28 16:08:40 -05:00
Paul HENRYS
8eceb8f327 Add call to gst_rtp_h264_pay_clear_sps_pps() when receiving a STREAM_START event
https://bugzilla.gnome.org/show_bug.cgi?id=692787
2013-11-04 14:36:28 -05:00
Olivier Crête
4c6e636720 rtph264pay: Use the SPS/PPS handling function from the depayloader
Remove duplicated copies

https://bugzilla.gnome.org/show_bug.cgi?id=705553
2013-08-13 10:38:23 -04:00
David Svensson Fors
692206d3a7 rtph264pay: avoid double buffer unmap on error
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703171
2013-06-27 17:14:11 +02:00
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
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
ae05ed4f05 rtph264pay: If the adapter is empty on EOS don't try to map its content
https://bugzilla.gnome.org/show_bug.cgi?id=699314
2013-05-01 15:49:45 +02:00
Ognyan Tonchev
3f8ad30cee rtph264pay: Don't use upstream caps with peer_query_caps ()
Calling gst_pad_peer_query_caps () on the src pad with the caps
upstream can produce as a filter from gst_rtp_h264_pay_getcaps ()
is wrong and makes caps negotiation fail if upstream caps are not
NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=695629
2013-03-11 16:55:13 -04:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00
Patricia Muscalu
7a863e4d8d rtph264pay: do not push unmapped data
Also do not use a GstBuffer after it has been pushed into the adapter.

https://bugzilla.gnome.org/show_bug.cgi?id=685213
2012-10-04 09:22:50 +01:00
Olivier Crête
bc252d29ee rtph264pay: Make sure the caps don't have duplicated sps/pps 2012-09-21 17:36:12 -04:00
Mark Nauwelaerts
fa90dfc4df rtph264pay: avoid crashing on NULL access in debug message 2012-09-07 15:25:52 +02:00
Olivier Crête
264bcf7d6f rtph264pay: Make it actually work after cleanups 2012-08-08 19:49:05 -07:00
Patricia Muscalu
d38ac43a27 rtph264pay: use buffer lists
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679994
2012-07-17 10:10:14 +02:00
Tim-Philipp Müller
c6224443a4 rtph264pay: avoid some relocations 2012-07-06 19:11:02 +01:00
Tim-Philipp Müller
cffbf8cfc3 rtph264pay: remove deprecated and non-functional "profile-level-id" property
This is now optionally taken from downstream caps, so can be
specified via a capsfilter after the payloader.
2012-07-06 14:46:22 +01:00
Wim Taymans
6d158775bb rtph264pay: cleanups
Use the caps properties for alignment and format.
Remove some old properties, we always want to use bufferlists when we can now.
2012-06-28 12:00:09 +02:00
Wim Taymans
429bda6923 h264pay: prefer AVC, it's easier to parse etc 2012-06-28 11:32:03 +02:00
Luis de Bethencourt
c81fff0471 rtp: fix build issue in gstrtph264pay.c 2012-05-24 09:29:25 +01:00
Jonas Holmberg
7bf3a1bf95 rtph264pay: Add unrestricted caps
If there are no profile restrictions downstream, return caps with
profile=constrained-baseline in the first structure and append
unrestricted caps as the last structure.

Fixes bug #672019
2012-05-24 10:01:19 +02:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Mark Nauwelaerts
e5ab3cc0a0 rtph264pay: ensure output caps are set when pushing output data
... even if some SPS/PPS has not passed by yet.
2012-03-26 18:38:34 +02:00
Wim Taymans
d65de434f5 rtph264pay: do DTS and PTS correctly 2012-03-13 18:07:18 +01:00
Wim Taymans
a32d944a38 fix for caps api changes 2012-03-11 19:06:37 +01:00
Olivier Crête
18899cf94d rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-21 10:51:43 +01:00
Wim Taymans
225e98d623 Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacenc.c
	ext/jack/gstjackaudioclient.c
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	ext/pulse/plugin.c
	ext/shout2/gstshout2.c
	gst/matroska/matroska-mux.c
	gst/rtp/gstrtph264pay.c
2012-02-10 16:23:14 +01:00