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124 commits

Author SHA1 Message Date
Matthew Waters
84a9f9c61f rtp/source: use extended sequence number helper 
Instead of rolling our own

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
 2024-06-24 13:13:28 +00:00
Sebastian Dröge
9b323a6519 Use Option::is_some_and(...) instead of Option::map_or(false, ...) 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
 2024-06-19 13:03:37 +00:00
Sebastian Dröge
4677948a82 rtp: av1pay: Derive Default trait for the state instead of manual implementation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
 2024-06-18 08:07:24 +03:00
Sebastian Dröge
d357a63bf9 rtp: av1pay: Correctly use N flag for marking keyframes 
The "first packet of a coded video sequence" means that this should be
the first packet of a keyframe that comes together with a sequence
header, not the first packet of a new frame.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/558

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
 2024-06-18 08:06:59 +03:00
Sebastian Dröge
5cd9e34265 rtp: av1pay: Correctly skip over ignored OBUs 
The reader is already after the header at this point so only the OBU
content has to be skipped.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
 2024-06-18 08:06:59 +03:00
Sebastian Dröge
bbe38b9599 rtp: av1: Drop padding OBUs too like Chrome does 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
 2024-06-18 08:06:59 +03:00
Sebastian Dröge
343680ffea rtp: av1depay: Don't return an error if parsing a packet fails 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
 2024-06-14 13:13:21 +00:00
Sebastian Dröge
477855789d rtp: av1depay: Also log warnings on errors 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
 2024-06-14 13:13:21 +00:00
Sebastian Dröge
93c9821cba rtp: av1depay: Drop unusable packets as early as possible 
Otherwise they would pile up until a discontinuity or until we can
actually output something.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
 2024-06-14 13:13:21 +00:00
Sebastian Dröge
0ca4a3778a rtp: av1depay: Parse internal size fields of OBUs and handle them 
They're not recommended by the spec to include in the RTP packets but it
is valid to include them. Pion is including them.

When parsing the size fields also make sure to only take that much of a
payload unit and to skip any trailing data (which should not exist in
the first place).

Pion is also currently storing multiple OBUs in a single payload unit,
which is not allowed by the spec but can be easily handled with this
code now.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/560

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
 2024-06-14 13:13:21 +00:00
Sebastian Dröge
69c3c2ae46 Fix various new clippy 1.79 warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1620>
 2024-06-14 08:33:49 +03:00
Matthew Waters
260b04a1cf rtpbin2: protoct against adding with overflow 
If jitter is really bad, then this calculation may overflow.  Protect
against that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1605>
 2024-06-06 11:43:26 +00:00
Tim-Philipp Müller
ab2f5e3d8d rtp: ac3: add some unit tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
2b68920f82 rtp: tests: add possibility to make input live 
.. for payloaders that behave differently with live
and non-live inputs (e.g. audio payloaders which by
default will pick different aggregation modes based
on that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
6597ec84eb rtp: tests: add possibility to check duration of depayloaded buffers 
.. and clarify an expect panic message.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
6b628485c5 rtp: Add AC-3 RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00
Sebastian Dröge
a7418fb483 rtp: Use released version of rtcp-types 2024-05-29 10:30:40 +03:00
Matthew Waters
df32e1ebfa rtpsend: ensure only a single rtcp pad push 
Otherwise, it can occur that multiple rtcp packets may be produced out
of order.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
525179f666 rtpbin2: handle ssrc collisions 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
1600d3b055 rtpbin2: split send and receive halves into separate elements 
There is now two elements, rtpsend and rtprecv that represent the two
halves of a rtpsession.  This avoids the potential pipeline loop if two
peers are sending/receiving data towards each other.  The two halves can
be connected by setting the rtp-id property on each element to the same
value and they will behave like a combined rtpbin-like element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
0121d78482 rtpbin2: expose session signals for new/bye ssrc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
d480c6c2d3 rtpbin2/config: add stats to session GObject 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
7d5789032a rtpbin2/config: add a new-ssrc signal 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
06f40e72cb rtpbin2: implement a session configuration object 
Currently only contains pt-map

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
48e7a2ed06 jitterbuffer: handle flush-start/stop 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
66306e32f2 jitterbuffer: remove mpsc channel for every packet 
It is very slow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
327f563e80 jitterbuffer: implement support for serialized events / queries 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
74ec83a0ff rtpbin2: implement and use synchronization context 
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
1865899621 rtpbin2: implement jitterbuffer 
The jitterbuffer implements both reordering and duplicate packet
handling.

Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
2b4ec75bc5 rtpbin2: Add support for receiving rtcp-mux packets 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
e09ad990fa rtpbin2: Implement support for reduced size RTCP (RFC 5506) 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
1e4a966c92 rtpbin2: Add support for sending NACK/PLI and FIR 
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
66c9840ad8 rtpbin2: Add handling for receiving NACK/PLI and FIR 
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Matthew Waters
2c86f18a99 rtpbin2: add support for RFC 4585 (RTP/AVPF) 
Implements the timing rules for RTP/AVPF

Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Matthew Waters
27ad26c258 rtp: Initial rtpbin2 element 
Can receive and recevie one or more RTP sessions containing multiple
pt/ssrc combinations.

Demultiplexing happens internally instead of relying on separate
elements.

Co-Authored-By: François Laignel <francois@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
984a9fe5ff rtp: Don't restrict payload types for payloaders 
WebRTC uses payload types 35-63 as dynamic payload types too to be able
to place more codec variants into the SDP offer.

Instead of allowing just certain payload types, completely remove any
restrictions and let the user decide. There's technically nothing wrong
with using any payload type, especially when using the encoding-name.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
 2024-05-27 09:34:16 +00:00
Tim-Philipp Müller
4f74cb7958 rtp: klv: add test for fragmented payloads with packet loss 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
b6e24668a7 rtp: klv: add unit test with some packet loss 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
92a1e222f4 rtp: tests: add functionality to drop RTP packets after payloading 
Add ExpectedPacket::drop() to flag RTP packets that should not
be forwarded to the depayloader.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
de71e9dadd rtp: tests: print rtp timestamp mismatch minus the initial offset 
Unit tests specify a 0-based offset, so printing that plus the
random initial offset on failure is just needlessly confusing,
so subtract the initial offset when printing expected/actual
values. The real values are still printed as part of the assert.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
be7da027f8 rtp: klv: add some basic tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
1e33926dc5 fixup: klv payloader indentation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
c2f67bd3c9 fixup: klv depay: debug log indentation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
e7d0e0702a fixup: payloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
566e6443f4 rtp: Add KLV RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-25 20:21:50 +03:00
Sebastian Dröge
58e91c154c rtp: basedepay: Reset last used ext seqnum on discontinuities 
The ext seqnum counting is reset too so keeping the old one around will
cause problems with timestamping of the next outgoing buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1584>
 2024-05-24 10:23:06 +03:00
Sebastian Dröge
dcc0b47349 rtp: basepay: Fix header extension negotiation 
Only configure header extensions from the source pad caps if they exist
already in the source pad caps, otherwise the configuration will fail.
Extensions that are added via the signals might not exist in the source
pad caps yet and would be added later.

Also, if configuring an existing extension from the new caps fails,
remove it and try to request a new extension for it.

Additionally don't remove extensions from the caps that can't be
provided. No header extensions for them would be added to the packets,
but that's not a problem. Removing them on the other hand would cause
negotiation to fail. This only affects extensions that are already
included in the caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1577>
 2024-05-20 07:53:50 +00:00
Sebastian Dröge
0d33077df6 rtp: basedepay: Clean up header extension negotiation 
If configuring an existing extension from the new caps fails, remove it
and try to request a new extension for it.

Also remove all extensions from the list that are not provided in the
caps, instead of passing RTP packets to all of them anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1577>
 2024-05-20 07:53:50 +00:00
Tim-Philipp Müller
16608d2541 rtp: opus: add multichannel depay/pay test 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
 2024-05-18 09:29:29 +00:00
Tim-Philipp Müller
bab3498c6a rtp: opus: add multichannel pay/depay test 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
 2024-05-18 09:29:29 +00:00