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Author SHA1 Message Date
Sebastian Dröge
1af18f3028 webrtc: Require Send+Sync for signaller implementations 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
 2024-01-18 10:01:01 +02:00
Eva Pace
80b58f3b45 net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller 
The JanusVRWebRTCSink is a new plugin that integrates with the Video
Room plugin of the Janus Gateway, which simplifies WebRTC communication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
 2024-01-17 20:33:57 +00:00
Maksym Khomenko
773ebc7854 webrtcsrc: don't restrict RTP extensions to TWCC only 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1381>
 2024-01-17 07:34:01 +00:00
Sebastian Dröge
dfa95d8ed3 webrtc: Update to livekit-api / livekit-protocol 0.3 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
 2024-01-16 07:52:48 +00:00
Maksym Khomenko
fecbe01e06 webrtcsink: make 'extensions' property usage conditional 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1423>
 2024-01-16 07:13:56 +00:00
Sebastian Dröge
73a53e38c4 aws: s3: Disable remaining tests too for now 
They fail state changes, which cases `GstHarness` to abort.
 2024-01-16 09:13:41 +02:00
Arun Raghavan
fd3675aac0 aws: s3: Temporarily disable putobject tests 
Disabling while we figure out why it's failing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1429>
 2024-01-15 21:43:25 -05:00
Arun Raghavan
8b18ca15b5 Revert "aws: Disable putobjectsink tests for now" 
This reverts commit b128d127c2.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
 2024-01-11 15:38:36 -05:00
Arun Raghavan
06213714c5 aws: putobjectsink: Fix a couple of minor log typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
 2024-01-11 15:38:36 -05:00
Nirbheek Chauhan
2d85048925 webrtc/signalling: We get the address when accepting 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
 2023-12-29 13:28:48 +00:00
Nirbheek Chauhan
63b568f4a0 webrtc/signalling: Fix potential hang and FD leak 
If a peer connects via TCP and never initiates TLS, then the server
will get stuck in the accept loop. Spawn a task when accepting a TLS
connection, and timeout if it doesn't complete in 5 seconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
 2023-12-29 13:28:48 +00:00
Maksym Khomenko
17f0b61576 webrtcsink: add payloader-setup signal 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
 2023-12-23 08:02:08 +00:00
Sebastian Dröge
b128d127c2 aws: Disable putobjectsink tests for now 
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
 2023-12-22 13:25:12 +02:00
Arun Raghavan
6d47045a60 aws: s3sink: Fix spelling of debug category 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
 2023-12-18 16:13:48 -05:00
Arun Raghavan
410d104ad6 aws: s3putobjectsink: Add a flush-on-error property 
Makes sure we can send out data even if the pipeline shutdown in error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
 2023-12-18 16:13:48 -05:00
Arun Raghavan
12dbf50ddc aws: s3putobjectsink: Add some thresholds for flushing 
Lets us connect when we perform a flush

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
 2023-12-18 16:13:48 -05:00
Arun Raghavan
a54b2dd39e aws: s3: Add a new awss3putobjectsink 
When streaming small amounts of data, using awss3sink might not be a
good idea, as we need to accumulate at least 5 MB of data for a
multipart upload (or we flush on EOS).

The alternative, while inefficient, is to do a complete PutObject of
_all_ the data periodically so as to not lose data in case of a pipeline
failure. This element makes a start on this idea by doing a PutObject
for every buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
 2023-12-18 10:39:23 -05:00
Sebastian Dröge
81dd45c814 webrtc: Downgrade aws-smithy-http to 0.60 
Version 0.61 was yanked from crates.io.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1407>
 2023-12-14 09:11:07 +02:00
Sebastian Dröge
2f2bf6ca8f webrtc: Update to aws-smithy-http 0.61 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
 2023-12-09 12:21:38 +02:00
Sebastian Dröge
0bae18fe0d rtp: Update to bitstream-io 2.0 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
 2023-12-09 12:17:51 +02:00
Sebastian Dröge
181bd13103 Update to async-tungstenite 0.24 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
 2023-12-09 12:17:11 +02:00
Guillaume Desmottes
6dfd1c1496 use new debug and parse API 
Changes from https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1355

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1403>
 2023-12-04 15:58:21 +01:00
Sebastian Dröge
f13574d8ed Update further AWS SDK crates to 1.0 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1400>
 2023-11-26 10:26:02 +02:00
Mathieu Duponchelle
cf1c7600a2 webrtcsink: don't panic on failure to request pad from webrtcbin 
webrtcbin will refuse pad requests for all sorts of reasons, and should
be logging an error when doing so, simply post an error message and let
the application deal with it, the reason for the refusal should
hopefully be available in the logs to the user.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1399>
 2023-11-24 19:53:38 +01:00
Sebastian Dröge
c3ced8c7e6 Update to AWS SDK 1.0 / 0.60 / 0.39 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1397>
 2023-11-21 10:32:59 +02:00
Sebastian Dröge
1d9c89e3fe Update to AWS SDK 0.101 / 0.59 / 0.38 2023-11-20 10:13:13 +02:00
Sebastian Dröge
66c62d69b9 aws: Stop using deprecated aws_config function in the test 2023-11-18 10:16:24 +02:00
Taruntej Kanakamalla
43ee6bfc1c net/webrtc: add whipserversrc 
Implement new signaller WhipServerSignaller
 - an http server using 'warp'
 - handlers for the POST, OPTIONS, PATCH and DELETE
 - fixed path `/whip/endpoint` as the URI
 - fixed value 'whip-client' as the producer peer id
 - fixed resource url `/whip/resource/whip-client`

Derive whipserversrc element from BaseWebRTCSrc
 - implement constructed method for ObjectImpl to set
  non-default signaller, i.e., WhipServerSignaller
 - bind the properties stun-server and turn-servers to those on
   the Signaller

Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller
 - it will be emitted by the webrtcsrc when the webrtcbin element is ready
 - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state
   and perform send with the answer sdp via the channel
 - the WhipServer will hold its HTTP response in POST handler until this signal
   is received or timeout which happens early

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
 2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
ed3aa740be net/webrtc: deprecate consumer-added on the signaller 
add a new signal webrtcbin-ready in this place doing same
thing but can be used for both consumers and producers

Please note this change is only to the consumer-added
signal on the signaller interface.
The consumer-added signal on the webrtcsink is unchanged

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
 2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
2d3d03b4d3 net/webrtc: rename WhipSignaller as WhipClientSignaller 
remove generalized names to accommodate for the WhipServer
- name the Signaller for whipsink as WhipClient
- name the Settings for whipsink as WhipClientSettings
- name the State for whipsink as WhipClientState

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
 2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
a0638ec983 net/webrtc: Extract BaseWebRTCSrc 
Define a Base for all the webrtcsrc type elements
so they can all be derived from it. Similar to base
element defined for webrtcsink type elements

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
 2023-11-17 18:08:44 +00:00
Sebastian Dröge
dee27e35b7 Update to latest AWS SDK 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1395>
 2023-11-17 11:22:29 +02:00
Sebastian Dröge
58723f2a8c Update to AWS SDK 0.36 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1394>
 2023-11-15 17:20:58 +02:00
François Laignel
9250c592a7 ndi: don't accumulate meta with audio only streams 
Currently, only closed caption metadata are supported. When the next video
frame is received, pending meta are dequeued and parsed. If close captions
are found, they are attached to the video frame.

For audio only streams, it doesn't make sense to enqueue metadata. They would
accumulate in `pending_metadata` and would never be dequeued.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/460

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1392>
 2023-11-13 19:26:23 +01:00
Sebastian Dröge
39155ef81c ndisrc: Implement zerocopy handling for the received frames if possible 
Also move processing from the capture thread to the streaming thread.
The NDI SDK can cause frame drops if not reading fast enough from it.

All frame processing is now handled inside the ndisrcdemux.

Also use a buffer pool for video if copying is necessary.

Additionally, make sure to use different stream ids in the stream-start
event for the audio and video pad.

This plugin now requires GStreamer 1.16 or newer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
 2023-11-13 13:22:48 +02:00
Sebastian Dröge
2afffb39dd ndi: Don't mark private type as public 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
 2023-11-13 10:29:25 +02:00
Sebastian Dröge
99d7cce0d6 ndi: Refactor frame structs to have static lifetimes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
 2023-11-13 10:29:25 +02:00
Sebastian Dröge
eb137ec6dc ndi: Remove wrong Clone impl on RecvInstance 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
 2023-11-13 10:29:25 +02:00
Arun Raghavan
771741c10c Revert "s3: tests: Remove emoji-based tests for now" 
This reverts commit a49a5dcb11.

Now that hotdoc should work with emoji, let's bring the tests back.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1386>
 2023-11-09 11:50:53 -05:00
Maksym Khomenko
e5fd2c3568 webrtcsrc: add turn-servers property 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1380>
 2023-11-04 10:19:45 +00:00
Mathieu Duponchelle
5371eb52ad Port to AWS SDK 0.57/0.35 
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
 2023-11-03 15:13:45 +00:00
Sebastian Dröge
f7745a336f aws: Update to test-with 0.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
 2023-11-03 15:13:45 +00:00
Sebastian Dröge
16b917abb1 Update for gst::Rank API changes 2023-11-02 14:10:59 +02:00
Piotr Brzeziński
436b6d8efb gstwebrtc-api: Patch webrtc-adapter to fix Safari behaviour 
There's currently a Safari-side bug causing webrtc-adapter to be unable to correctly shim the empty-candidate scenario
which we're using. This patch is very much a workaround and should be removed as soon as Safari and/or webrtc-adapter
fixes this on their side.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/439
https://github.com/webrtcHacks/adapter/issues/1140

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1377>
 2023-10-30 16:36:11 +00:00
Sebastian Dröge
16c00ae3f5 Set sync=false in rsfilesink / s3sink 
BaseSink defaults to sync=true and that doesn't make much sense for
these elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1376>
 2023-10-30 17:38:46 +02:00
Sebastian Dröge
855b03a9ea Use let-else instead of match for weak reference upgrades 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1375>
 2023-10-30 11:34:35 +02:00
Sebastian Dröge
557b249e11 Update to AWS SDK 0.34 and tracing-log 0.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1374>
 2023-10-27 10:19:15 +03:00
Arun Raghavan
d27a04e067 hlssink3: Close the playlist giostreamsink on stop if possible 
This is a property that will be available from GStreamer 1.24, and will
ensure that we are able to flush the playlist during the READY->NULL
transition instead of when the element is freed.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/423
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1360>
 2023-10-24 21:03:14 +00:00
Arun Raghavan
a49a5dcb11 s3: tests: Remove emoji-based tests for now 
These break hotdoc, which we need to fix first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
 2023-10-24 12:52:12 -04:00
Arun Raghavan
bb26e04a55 aws: s3: Properly percent-decode GstS3Url 
We previously only percent-decoded the first fragment. This doesn't
necessarily harm anything, but for consistency we keep the structure
un-encoded, and encode when converting to a string representation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
 2023-10-24 12:52:12 -04:00
Arun Raghavan
51129febeb aws: s3sink: Fix handling of special characters in key 
Properly URL-encode the string if needed, and add some tests for a
couple of cases.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/431
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
 2023-10-24 12:52:12 -04:00
Sebastian Dröge
829469d0fe rtpav1depay: Don't push stale temporal delimiters downstream 
Only push them downstream once a complete OBU was assembled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1367>
 2023-10-24 11:13:35 +00:00
Sebastian Dröge
1f5e9a9335 rtpav1depay: Skip unexpected leading fragments 
If a packet is starting with a leading fragment but we do not expect to
receive one, then skip over it to the next OBU.

Not doing so would cause parsing of the middle of an OBU, which would
most likely fail and cause unnecessary warning messages about a
corrupted stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1367>
 2023-10-24 11:13:35 +00:00
Sebastian Dröge
73ff822d24 Update to quick-xml 0.31 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1368>
 2023-10-24 09:55:50 +03:00
Jordan Petridis
a2d7f42138 Fix compilation after glib bindings changes 
loggable_error! can now expand variables and we no longer need
the format! on our side.

https://github.com/gtk-rs/gtk-rs-core/pull/1210

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1366>
 2023-10-22 01:20:56 +03:00
Sebastian Dröge
2ce04c6a78 webrtc: Update to livekit 0.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1293>
 2023-10-18 10:30:59 +03:00
Sebastian Dröge
d468e1e4a6 Clean up usage of pad probes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1361>
 2023-10-17 08:44:06 +03:00
François Laignel
50dd519c4f net/webrtcsrc: define signaller property as CONSTRUCT_ONLY 
The "signaller" property used to be defined as MUTABLE_READY which meant that
the property was always set after `constructed()` was called.

Since `connect_signaller()` was called from `constructed()`, only the default
signaller was used.

This commit sets the "signaller" property as CONSTRUCT_ONLY. Using a builder,
this property will now be set before the call to `constructed()`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1324>
 2023-10-12 17:38:09 +00:00
François Laignel
785c9557c8 net/webrtcsink: drop State lock before calling set-local-description 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1325>
 2023-10-12 15:45:58 +00:00
François Laignel
c021e2b69f net/webrtcsink: don't miss ice candidates 
During `on_remote_description_set()` processing, current session is removed
from the sessions `HashMap`. If an ice candidate is submitted to `handle_ice()`
by that time, the session can't be found and the candidate is ignored.

This commit wraps the Session in the sessions `HashMap` so an entry is kept
while `on_remote_description_set()` is running. Incoming candidates received by
`handle_ice()` will be processed immediately or enqueued and handled when the
session is restored by `on_remote_description_set()`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1325>
 2023-10-12 15:45:58 +00:00
Sebastian Dröge
42008fb895 aws: Update to test-with 0.11 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1358>
 2023-10-12 06:57:28 +00:00
Lieven Paulissen
05aa9fa431 ndisrc: Assume input with more than 8 raw audio channels is unpositioned 
gst_audio_channel_positions_from_mask() will otherwise print warnings
all the time.

Fixes #444

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1357>
 2023-10-12 09:12:02 +03:00
François Laignel
022afa6375 ndi: use v210 encoding for cc and attach to video frame 
The NDI closed captions specifications [1] define a variation where metadata is
attached to the video frame. This requires the AFD buffer to be v210 encoded.
This commit applies this strategy.

Another difference with previous version is that when an error occurs while
encoding or decoding a meta, next meta are also tried instead of failing
immediately.

Receiving closed captions as a standalone metadata is kept for interoperability
purposes. In this case, metadata is also expected to be v210 encoded.

[1]: http://www.sienna-tv.com/ndi/ndiclosedcaptions.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1356>
 2023-10-11 21:25:29 +02:00
Maksym Khomenko
5b03f7d7b0 webrtcsrc: use @watch instead of @to-owned 
@to-owned increases refcount of the element, which prevents the object from proper destruction, as the initial refcount with ElementFactory::make is larger than 1.

Instead, use @watch to create a weak reference and unbind the closure automatically if the object gets destroyed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1355>
 2023-10-11 11:54:51 +03:00
Sebastian Dröge
3fc6220009 Update to AWS SDK 0.33 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1354>
 2023-10-09 11:28:05 +03:00
Taruntej Kanakamalla
245185d2f6 net/webrtc/whip_signaller: Use the correct URL during redirect 
Copy of 90e06dc3 for whipclientsink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1351>
 2023-10-06 13:11:46 +00:00
Maksym Khomenko
e4096b5157 webrtcsink: README: add documentation for custom signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1340>
 2023-10-06 12:58:04 +03:00
Maksym Khomenko
a9719cada2 webrtcsink: add custom signaller example 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1340>
 2023-10-06 12:58:03 +03:00
Sebastian Dröge
1c4833bc5d Update to AWS SDK 0.32 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1352>
 2023-10-06 09:11:17 +03:00
Sebastian Dröge
4569b7eca6 Fix various new 1.73 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1347>
 2023-10-03 17:47:30 +03:00
Sebastian Dröge
450ffbe452 Update for VideoFrame / GLVideoFrame API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1345>
 2023-10-02 13:25:25 +03:00
Piotr Brzeziński
fe4273ca2a webrtc: Fix paths in README 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1342>
 2023-09-29 17:05:29 +02:00
Sean DuBois
90e06dc37b net: webrtc/webrtchttp: Respect HTTP redirects 
Properly follow redirect URL. Before new request would be made, but with
original URL again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1335>
 2023-09-26 19:29:41 -04:00
Seungha Yang
22cc8c4986 hlssink3: Update README 
Mention newly added hlscmafsink element and new properties

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:34:05 +09:00
Seungha Yang
1888a2eb82 hlscmafsink: Add live recording example 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:34:05 +09:00
Seungha Yang
52117e4b11 hlsbasesink: Add enable-endlist property 
Write "EXT-X-ENDLIST" tag at the end of stream if enabled, and
default to "TRUE" which is the hlssink2's behavior as well

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:34:05 +09:00
Seungha Yang
7835d78b3d hlssink3: Add hlscmafsink element 
Adding cmafmux based hls sink element

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:34:00 +09:00
Seungha Yang
5b563006f9 hlssink3: Add baseclass implementation 
Adding HlsBaseSink class to make code reusable

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:32:16 +09:00
Seungha Yang
0fe69cea9f hlssink3: Various cleanup 
* Simplify state/playlist management
* Fix a bug that segment is not deleted if location contains directory
and playlist-root is unset
* Split playlist update routine into two steps, adding segment
to playlist and playlist write

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:32:16 +09:00
Seungha Yang
d8546dd140 hlssink3: Don't remove uri from playlist if playlist-length is zero 
Behave as documented in property description

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:32:16 +09:00
Seungha Yang
8e4863e9cd hlssink3: Don't remove old files if max-files is zero 
Follow hlssink2 element's behavior

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:32:16 +09:00
Seungha Yang
a8d67cc607 hlssink3: Remove unused deps 
gstreamer-base dep is unused. And use gst::glib

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:32:16 +09:00
Seungha Yang
c4d371d163 hlssink3: Use Path API for getting file name 
Current implementation does not support Windows path separator.
Use Path API instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:32:16 +09:00
Seungha Yang
7f16ac3915 hlssink3: Use sprintf for segment name formatting 
The zero-padded naming requirement is unnecessary. Use simple
sprintf instead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
 2023-09-25 21:32:16 +09:00
Sebastian Dröge
9595c6a1e5 Update to AWS SDK 0.31 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1334>
 2023-09-25 13:36:12 +03:00
Arun Raghavan
8bbfb10cba hlssink3: Minor PDT-related naming fixups 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1332>
 2023-09-20 16:52:55 -04:00
rajneeshksoni
a7fe24a294 hlssink3: Add property track-pipeline-clock-for-pdt. 
This is required to take care of clock skew between
system time and pipeline time.
`track-pipeline-clock-for-pdt: true` mean utd time is
sampled for first segment and for subsequent segments
keep adding the time based on pipeline clock. difference
of segment duration and PDT time will match.
track-pipeline-clock-for-pdt: false` mean utd time is
sampled for each segment. system time may jump forward
or backward based on adjustments. If application needs
to synchronization of external events `false` is
recommended.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1145>
 2023-09-20 13:54:48 +03:00
rajneeshksoni
4be24fdcaf hlssink3: Allow adding EXT-X-PROGRAM-DATE-TIME tag. 
- connect to `format-location-full` it provide the first
sample of the fragment. preserve the running-time of the
first sample in fragment.
- on fragment-close message, find the mapping of running-time
to UTC time.
- on each subsequent fragment, calculate the offset of the
running-time with first fragment and add offset to base
utc time

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1145>
 2023-09-20 13:54:48 +03:00
Sebastian Dröge
b12278e334 onvifmetadataparse: Skip metadata frames with unrepresentable UTC time 
Previously we would panic, which causes the element to post an error
message. Instead, simply skip metadata frames if their UTC time since
the UNIX epoch can't be represented as nanoseconds in u64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1326>
 2023-09-16 10:59:27 +03:00
Seungha Yang
225482f7ed webrtcsink: Propagate GstContext messages 
Implement CustomBusStream so that NEED_CONTEXT and HAVE_CONTEXT
messages from session/discovery can be forwarded to parent
pipeline and also GstContext can be shared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1322>
 2023-09-15 00:26:08 +09:00
Seungha Yang
1de7754616 webrtcsink: Add support for d3d11 memory and qsvh264enc 
Adding d3d11 memory and qsvh264enc support

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1322>
 2023-09-15 00:26:04 +09:00
Robert Ayrapetyan
18967dadbf gstwebrtc-api: drop guacamole 
fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/417

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1317>
 2023-09-11 19:21:41 +00:00
François Laignel
029fa9b8dc net/ndi: improve interoperability robustness 
`quick-xml::reader::Reader::trim_text(true)` doesn't remove white spaces and
tabs from XML text. Besides, for interoperability robustness we also need to
remove carriage returns and line feeds.

Also improve the default capacities for the `SmallVec`s.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1321>
 2023-09-11 06:22:41 +00:00
Mathieu Duponchelle
2381558169 webrtcsink: fix codec selection discoveries 
Since ab1ec12698:

webrtcsink: Add support for pre encoded streams

Discovery pipelines for remote offers were no longer fed any buffers.

While some encoders could already produce caps with no input buffers,
others, such as x264enc, simply hung forever. This resulted in no answer
getting produced if for instance video-caps were constrained to H264.

Fix this by tracking discovery pipelines at the State rather than the
InputStream level, removing the useless distinction of Initial vs.
CodecSelection discoveries, and always feeding all the current
discovery pipelines with incoming buffers.

For reference, the issue here was that codec selection discoveries were
assigned to local clones of InputStreams, not tracked anywhere, and thus
not iterated for discoveries when queuing incoming buffers from the
chain function, as it only looked at the original instance of
InputStream's in state.streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1319>
 2023-09-08 12:58:08 +00:00
François Laignel
9604dea90a net/ndi: add closed caption support 
Closed caption support in NDI is described as a proposal in [1] & [2].

The proposal consists in encapsulating c608 or c708 closed caption in ADF
packets and pushing them in an XML tag as part of NDI Metadata.

This commit implements this proposal.

[1]: http://www.sienna-tv.com/ndi/ndiclosedcaptions.html
[2]: http://www.sienna-tv.com/ndi/ndiclosedcaptions608.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1320>
 2023-09-07 14:28:24 +02:00
Robert Ayrapetyan
e83238b681 webrtcsink: fix TWCC extension adding 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1310>
 2023-09-04 18:27:51 +00:00
Sebastian Dröge
b0b63e58f8 ndi: Comment out empty Opus handling and add FIXME comment 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
 2023-08-29 12:21:38 +00:00
Sebastian Dröge
8d433761d1 Fix indentation of let-else blocks 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
 2023-08-29 12:21:38 +00:00
Taruntej Kanakamalla
de6d2e7f40 net/webrtc: rename whipwebrtcsink as whipclientsink 
add a deprecation warning in whipsink to indicate it
should be used only for RTP content
add documentation in whipsink code regarding usage and
deprecation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
 2023-08-26 10:53:30 +05:30
Sebastian Dröge
905da44958 Update to AWS SDK 0.30 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1313>
 2023-08-25 09:46:52 +03:00
Andoni Morales Alastruey
3c1f05cdc3 webrtcsrc: document how to use the element for remote control 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
 2023-08-10 17:43:51 +00:00
Andoni Morales Alastruey
3000b08ec7 webrtcsrc: add support for navigation events 
This provides support GstNavigation events handling in webrtcsrc so that
a GStreamer client can be used to control remotely a GStreamer server,
similar to how the web client is capable of controlling a wpesrc.
This is part of a larger set of patches that require more work on the
sinks and sources.
server: d3d11screencapturesrc ! webrtcsink enable-data-channel-navigation=true
client: webrtcsrc enable-data-channel-navigation=true ! d3d11videosink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
 2023-08-10 17:43:51 +00:00
Loïc Le Page
e5e3dc6e19 net/webrtc/signaller: add property to get the connection client ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
 2023-08-10 17:30:21 +02:00
Loïc Le Page
7af2ff0843 net/webrtc/signaller: advertise running producers in Listener mode 
When starting a webrtcsrc-signaller client in Listener mode, only the producers
started after the client connection were advertised. All currently
running producers were ignored unlike the gstwebrtc-api behavior. This
commit now lists all running producers when the client Listener connects
and advertises them through the "producer-added" signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
 2023-08-10 17:30:21 +02:00
Sebastian Dröge
d688aeb184 Update versions to 0.12.0-alpha.1 2023-08-10 17:21:11 +03:00
Sebastian Dröge
3b41f206bc Don't generate .def files for plugins 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
 2023-08-09 13:54:34 +03:00
Sebastian Dröge
b3826c108d webrtc: Update to async-tungstenite 0.23 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
 2023-08-09 13:18:44 +03:00
Sebastian Dröge
5ee46a214c webrtc: Use #[repr(C)] to get a C-compatible layout for the Signaller struct 
This is required by GObject for class/interface and instance structs and
the reason why implementing the `glib::ObjectInterface` trait is unsafe.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/397

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1297>
 2023-08-09 10:32:44 +03:00
Sebastian Dröge
cac791a6ca aws/webrtc: Update to AWS SDK 0.56/0.29 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1295>
 2023-08-07 20:03:51 +03:00
Sebastian Dröge
2591feb72e Update a couple of dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1294>
 2023-08-07 11:42:32 +03:00
Sanchayan Maity
5b60ecbb18 net: webrtc/webrtchttp: Fix canceller usage 
Commit 08b6251a added the check to ensure only one canceller at a time for net/webrtc.

In `whipsink` and since `whipwebrtcsink` picked up the same implementation, there exists a
bug around the use of canceller. `whipsink` calls `wait_async` while passing the canceller
as an argument. The path `send_offer -> do_post -> parse_endpoint_response` results in the
canceller being replaced in each subsequent call to `wait_async`. Since `wait_async` call
does not ensure one canceller, with the async call the use of canceller/abort was subtly
broken. Similarly, for `whepsrc`.

We really don't need to use `wait_async` inside `do_post` for any `await` calls. If the
root future viz. `do_post` with `wait_async` is aborted, the child futures will be taken
care of.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1290>
 2023-08-04 10:01:11 +05:30
Mathieu Duponchelle
9680805bdb webrtcsink: don't forget to setup encoders for discoveries 
The "encoder-setup" signal must also be emitted for the encoders
used in discovery pipelines in order for the default settings to
be applied.

This otherwise meant that for instance the x264 encoder would
use a 60 frames latency, greatly delaying startup.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1289>
 2023-08-01 00:28:52 +02:00
Mathieu Duponchelle
dbeb65da06 webrtc/utils: fix typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1289>
 2023-08-01 00:28:32 +02:00
Sebastian Dröge
d4b3827efa webrtcsink: NVIDIA V4L2 encoders always require NVMM memory 
And if the input is not like that then a corresponding converter must be
inserted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1283>
 2023-07-24 10:14:59 +00:00
Sebastian Dröge
31b1cb8ca6 Update minimum supported Rust version to 1.70 
gtk-rs will update soonish too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1280>
 2023-07-19 09:19:34 +03:00
Mathieu Duponchelle
9707bb89e6 webrtcsink: fix pipeline when input caps contain max-framerate 
GstVideoInfo uses max-framerate to compute its fps, but this leads
to issues in videorate when framerate is actually 0/1.

Fix this by stripping away max-framerate from input caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1276>
 2023-07-13 22:18:08 +02:00
Sebastian Dröge
0331522128 webrtcsink: Configure only 4 threads for x264enc 
More threads can cause more slices to be created, and Chrome simply falls
apart if there are more than a few slices and fails decoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1275>
 2023-07-13 16:59:43 +03:00
Sebastian Dröge
ca51cf2509 webrtcsink: Translate force-keyunit events to force-IDR action signal for NVIDIA encoders 
NVIDIA's v4l2 encoder elements don't handle the force-keyunit events but
instead provide a custom action signal based API for requesting a
keyframe.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1274>
 2023-07-12 10:09:32 +00:00
Sebastian Dröge
bbd3d9ffe0 Remove unnecessary mut everywhere 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1273>
 2023-07-11 10:09:35 +03:00
Sebastian Dröge
ee4aca3010 webrtcsink: Set config-interval=-1 and aggregate-mode=zero-latency on rtph26[45]pay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
 2023-07-10 19:48:37 +03:00
Sebastian Dröge
957a28f239 webrtcsink: Set VP8/VP9 payloader based on payloader element factory name 
Instead of checking the encoder's name. There are more VP8/VP9 encoders
than the ones from the vpx plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
 2023-07-10 19:45:17 +03:00
Mathieu Duponchelle
1dd13c4812 webrtcsink: fix session_id / peer_id confusion 
In a few places, for instance parameter names, peer_id was still used
when session_id was actually getting passed.

Go through all instances of peer_id in webrtcsink/imp.rs and address
those mix-ups.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1269>
 2023-07-07 05:33:30 +00:00
Bilal Elmoussaoui
dd2d7d9215 Use re-exported once_cell 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Bilal Elmoussaoui
2cc98bf410 Adapt to glib::Continue rename 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Sebastian Dröge
58adebb325 Fix a couple of typos 2023-07-06 13:50:17 +03:00
Olivier Crête
08b6251a7a webrtc-utils: Ensure there is only one cancellable call at a time 
Since we only have one canceller at a time, panic if one try to
use it twice in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête
817b60a758 webrtc: Value.get() is already type checks in the property calls 
GObject will have ensured we get a GValue of the right type.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête
793ee66afa webrtcsink: Add LiveKit WebRTC sink and signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Seungha Yang
1f0ce101eb awstranscriber: Tone down log message 
It's not an ERROR case at all

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1263>
 2023-06-28 23:57:54 +09:00
Sebastian Dröge
c350f3c2af webrtcink: Use correct property types for nvvideoconvert 
These are enums and not plain integers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1256>
 2023-06-26 14:48:58 +00:00
Mathieu Duponchelle
84a33ca7b9 webrtcsink: bring in signalling code from whipsink as a signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
 2023-06-16 00:32:56 +02:00
Mathieu Duponchelle
f00a169081 webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1245>
 2023-06-15 20:41:53 +00:00
Mathieu Duponchelle
1200ae0ee6 webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Mathieu Duponchelle
64056c5527 net/webrtc: improve documentation layout 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Sebastian Dröge
8a7a1f519c webrtc: Update to fastrand 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1240>
 2023-06-09 09:36:51 +03:00
Mathieu Duponchelle
81ae675f2d webrtcsink: don't try to use cudaconvert if not present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
 2023-06-08 15:32:49 +02:00
Mathieu Duponchelle
7f78a8428e webrtcsink: dump discovery pipelines on state changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
 2023-06-08 15:32:49 +02:00
Mathieu Duponchelle
7447d95f1b webrtc/signalling: fix race condition in message ordering 
Spawning one task per message to send out instead of sending them out
sequentially from the one task used to poll the handler sometimes
resulted in peers receiving ICE candidates before SDP offers, triggering
hard to understand errors in the browser.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 13:24:45 +02:00
Mathieu Duponchelle
de0f7a08fe gstwebrtc-api: fix firefox errors about more than two stun servers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 13:24:45 +02:00
Mathieu Duponchelle
cd4b90fef4 webrtcsink/utils: remove unused decoders field in DecodingInfo 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 01:54:13 +02:00
Mathieu Duponchelle
271b583876 webrtcsink: avoid panic on unprepare from an async tokio context 
.. and log an error with advice on how to dispose of elements properly
from a tokio runtime.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1218>
 2023-06-07 19:57:19 +00:00
Sebastian Dröge
c65b3429ad Use MPL as license specifier for plugins only requiring GStreamer < 1.20 
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1235>
 2023-06-07 19:13:55 +03:00
Mathieu Duponchelle
fda5aed89f webrtcsink: encoded streams: address last review comments 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 16:05:28 +02:00
Thibault Saunier
ab1ec12698 webrtcsink: Add support for pre encoded streams 
This is a first step where we try to replicate encoding conditions from
the input stream into the discovery pipeline. A second patch will
implement using input buffers in the discovery pipelines.

This moves discovery to using input buffers directly. Instead of trying
to replicate buffers that `webrtcsink` is getting as input with testsrc,
directly run discovery based on the real buffers. This way we are sure
we work with the exact right stream type and we don't need encoders to
support encoding streams inputs.

We use the same logic for both encoded and raw input to avoid having
several code paths and makes it all more correct in any case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:32:40 +02:00
Thibault Saunier
059cdecf7d webrtc: Unify the Codec structure between sink and source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Thibault Saunier
cf32d9d668 webrtc: Move make_element to the utils 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Thibault Saunier
ce42723ad2 webrtc: Minor documentation enhancement 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Mathieu Duponchelle
6346d5608e net/aws/transcriber: track discont offset in input stream 
and add it up to subsequent transcripts.

This ensures synchronization is maintained even after the input stream
experiences a discontinuity and a gap in its timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1230>
 2023-06-02 08:55:11 +00:00
Mathieu Duponchelle
80582923bb aws_kvs_signaller: don't force us-east-1 region 
Instead use default region provider, with a fallback to us-east-1

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1228>
 2023-05-30 16:04:27 +00:00
Edward Hervey
31b06e52ea rtpgccbwe: Improve packet handling 
Both the delay-based *and* loss-based estimates should be computed instead of
just one. This ensures faster adaptation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1179>
 2023-05-29 08:20:36 +00:00