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Author SHA1 Message Date
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
Arun Raghavan b4b56eb282 aws: s3: Re-enable tests 
These seem to have stopped working due to bad/rotated creds. Should work
fine now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1623>
 2024-06-17 06:08:18 -04:00
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
Sebastian Dröge 3d4d785a2a webrtchttp: Fix race condition when unlocking 
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.

To solve this remember if cancellation should happen and reset this
later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
 2024-06-10 07:38:29 +00:00
Sebastian Dröge 51f6d3986f aws: Fix race condition when unlocking 
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.

To solve this remember if unlock() was called and reset this in
unlock_stop().

Also implement actual unlocking in s3hlssink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
 2024-06-10 07:38:29 +00:00
Sebastian Dröge 00aaecad07 quinn: Fix race condition when unlocking 
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.

To solve this remember if unlock() was called and reset this in
unlock_stop().

Also actually implement unlock() / unlock_stop() for the sink, and don't
cancel in stop() as unlock() / unlock_stop() would've been called before
that already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
 2024-06-10 07:38:29 +00:00
Sebastian Dröge 9945b702b8 reqwesthttpsrc: Fix race condition when unlocking 
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.

To solve this remember if unlock() was called and reset this in
unlock_stop().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
 2024-06-10 07:38:29 +00:00
Sebastian Dröge f68655b5e2 Update for gst::BufferList API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1610>
 2024-06-08 09:58:10 +03:00
Sebastian Dröge 30252a1b2e ndi: Add support for loading NDI SDK v6 
The library name and environment variable name have changed but the ABI
is completely compatible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1607>
 2024-06-06 14:51:09 +00:00
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
Sebastian Dröge ba70bb1154 deny: Add override for older tungstenite 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
 2024-06-06 10:34:12 +00:00
Sebastian Dröge 85c38107cf webrtc: Update to async-tungstenite 0.26 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
 2024-06-06 10:34:12 +00:00
Sanchayan Maity 8171a00943 net/quinn: Fix pad template naming typo 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1601>
 2024-06-05 13:44:40 +05:30
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
Tamas Levai 802ff6a67c net/quinn: Make QUIC role configurable 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
 2024-05-31 23:20:38 +02:00
Francisco Javier Velázquez-García 8fc652f208 webrtcsink: Refactor value retrieval to avoid lock poisoning 
When setting an incorrect property name in settings,
start_stream_discovery_if_needed would panic because it attempts to
unwrap a poisoned lock for settings.

This refactor avoids that situation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
 2024-05-31 08:10:23 +00:00
Francisco Javier Velázquez-García 568e8533fa webrtcsink: Fix typo in property name for av1enc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
 2024-05-31 08:10:23 +00:00
Arun Raghavan 04e9e5284c webrtc: signaller: A couple of minor doc fixups 
The expectation is `Returns:`, not `Return:`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
 2024-05-30 22:16:46 +03:00
Arun Raghavan 1c54c77840 webrtcsink: Add a mechanism for SDP munging 
Unfortunately, server implementations might have odd SDP-related quirks,
so let's allow clients a way to work around these oddities themselves.
For now, this means that a client can fix up the H.264 profile-level-id
as required by Twitch (whose media pipeline is more permissive than the
WHIP implementation).

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/516
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
 2024-05-30 22:16:46 +03:00
Taruntej Kanakamalla 83f76280f5 net/webrtc: Example for whipserver 
rudimentary sample to test multiple WHIP client connections

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
 2024-05-29 21:03:27 +00:00
Taruntej Kanakamalla 712d4757c3 net/webrtc/whip_signaller: multiple client support in the server 
- generate a new session id for every new client
use the session id in the resource url

- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer

- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
 2024-05-29 21:03:27 +00:00
Taruntej Kanakamalla de726ca8d2 net/webrtc: multi producer support in webrtcsrc 
- Add a new structure Session
  - manage each producer using a session
  - avoid send EOS when a session terminates, instead keep running
    waiting for any new producer to connect

- Maintain a bin element per session
  - each session bin encapsulates webrtcbin and the decoder if needed
    as well as the parser and filter if requested by the application
    (through request-encoded-filter)
  - this will be helpful to cleanup the session's respective elements
    when the corresponding producer terminates the session

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
 2024-05-29 21:03:27 +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
Nirbheek Chauhan 9485265769 rtspsrc2: Update rtpbin2 support to use rtprecv and rtpsend 
USE_RTPBIN2 is now USE_RTP2 because there is no "rtpbin2" now.

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