Commit graph

137 commits

Author SHA1 Message Date
Olivier Crête
6fed30c48e rtph26xpay: Avoid print when there is no bundle at end of packet 2019-07-03 19:05:29 +00:00
Olivier Crête
97f2fb4cc8 rtph26xpay: Wait until there is a VCL or suffix NAL to send
With unit tests.
2019-07-03 19:05:29 +00:00
Olivier Crête
5a9b602c9e rtph264pay: Report latency when in maximal aggregation mode 2019-07-03 19:05:29 +00:00
Olivier Crête
cede4f993d rtph264pay: Default to not adding latency when aggregating
Send the bundle as soon as there is one VCL unit in the packet at
the end of an incoming buffer.

The DELTA_UNIT flag is not reliable, so ignore it.
2019-07-03 19:05:29 +00:00
Olivier Crête
13d25583db rtph265pay: Replace fragmentation while-loop with for-loop
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
b46dab13d2 rtph264pay: Support STAP-A bundling
Add a new property "do-aggregate"* to the H.264 RTP payloader which
enables STAP-A aggregation as per [RFC-6184][1]. With aggregation enabled,
packets are bundled instead of sent immediately, up until the MTU size.
Bundles also end at access unit boundaries or when packets have to be
fragmented.

*: The property-name is kept generic since it might apply more widely,
   e.g. STAP-B or MTAP.
[1]: https://tools.ietf.org/html/rfc6184#section-5.7

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/434
2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
66a3db2083 rtph264pay: Fix delta-unit/discont handling when injecting SPS/PPS
Apply the wanted delta-unit and discont to the first packet; following
packets for this frame are always delta units and not discont.
2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
2a16160b57 rtph264pay: Replace fragmentation while-loop with for-loop 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
00936a8362 rtph264pay: Calculate the right max_fragments 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
fe99982dec rtph264pay: Rename payload_len to max_fragment_size 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
5051569713 rtph264pay: Clean up _payload_nal_fragment 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
d97c3f045c rtph264pay: Clean up _payload_nal
Move determining whether we need to fragment at all into the fragmenter.
2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
166c49b800 rtph264pay: Clean up _payload_nal_single 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
b3291620ca rtph264pay: Extract sending fragments into _payload_nal_fragment 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
e493f0ba09 rtph264pay: Extract sending a single packet into _payload_nal_single 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
40c23c06b1 rtph264pay: Define and use FU_A_TYPE_ID 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
bc0018370b rtph264pay: Use snake_case variables 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
28d6dfa51f rtph264pay: Clean up whitespace and syntax 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
798f320ba7 rtph264pay: Only mark the last fragment of an AU
Commit 4add820cce removed the check for
the end of fragmentation. As a result, all fragments of an AU's last
NALU were marked.

Potential fix for https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/540
2019-01-09 15:36:40 +00:00
Nicolas Dufresne
05059ce16b rtph264pay/rtph265pay: Fix use after free
We can't assume a buffer that has been pushed in the adapter is still
valid. This fixes a use after free detect when running test on jenkins.
2018-12-19 13:54:57 -05:00
Nicolas Dufresne
5e8cab71ea rtph26xpay: Remove unused IS_ACCESS_UNIT macro
This macro is not longer used. It was secretly checking if that nal was
a slice, and confusingly name to that one may think it was checking if
the nal is an AUD.
2018-12-18 13:39:46 -05:00
Nicolas Dufresne
1f72131781 rtph264pay: Fix reading timestamps from adapter
The code was reading the timestamp from the adapter before pushing the
new buffer into it. As a side effect, if the adapter was empty, we'd end
up using an older timestamp. In alignment=au, it means that all
timestamp was likely one frame in the past, while in alignment=nal, with
multiple slices per frame, the first slice would have the timestamp of
the previous one.
2018-12-18 13:30:05 -05:00
Nicolas Dufresne
4add820cce rtph264pay: Properly set the marker bit
The marker bit is used for efficient decoding. The assumption that
it should be set on the AUD is wrong, since the AUD is conceptually
starts the frame, while the marker is to indicate the end.

So properly set the marker bit as soon as we know we are ending an
AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
flag.
2018-12-18 13:30:05 -05:00
Nicolas Dufresne
13278fbcf5 rtph264pay: Protect against use of reserved NAL types
Don't allow external encoder to use one of the reserved NAL type
implicated in NAL aggreation. These out-of-spec NAL types, if passed
from the outside world will lead to an invalid RTP payload being
created.
2018-12-18 13:30:05 -05:00
Jan Alexander Steffens (heftig)
20758215b5 rtph26*pay: Update param set timestamp even if parameters unchanged
rtph264pay and rtph265pay skip updating the parameter set timestamp if
the units they see contain no new configuration. This can result in
them injecting duplicate parameters.

https://bugzilla.gnome.org/show_bug.cgi?id=796748
2018-08-16 16:49:16 +03:00
Sebastian Dröge
f3631e6837 rtp: Use running_time instead of PTS for config-interval calculations
PTS can start again from a different offset while the running time is
increasing. The only thing that matters here is the running time.

https://bugzilla.gnome.org/show_bug.cgi?id=796807
2018-07-24 18:14:28 +03:00
Philip Craig
ec11b228a4 rtph264pay: don't add trailing zeros to PPS/SPS
This would happen if input is byte-stream with four-byte
sync markers instead of three-byte ones. The code that
scans for sync markers will place the start of the NALU
on the third-last byte of the NALU sync marker, which
means that any additional zeros may be counted as belonging
to the previous NALU instead of being part of the next sync
marker. Fix that so we don't send SPS/PPS with trailing
zeros in this case.

https://bugzilla.gnome.org/show_bug.cgi?id=732758
2017-11-23 09:36:15 +01:00
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