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Author SHA1 Message Date
François Laignel 83d70d3471 webrtc: add RFC 7273 support 
This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization)
for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to
identify the clock. These attributes are handled by `rtpjitterbuffer` on the
consumer side. They MUST be part of the SDP offer.

When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the
clock's origin. Because the payloaders are not instantiated when the offer is
sent to the consumer, the RTP offset is set to 0 and the payloader
`timstamp-offset`s are set accordingly when they are created.

The `webrtc-precise-sync` examples were updated to be able to start with an NTP
(default), a PTP or the system clock (on the receiver only). The rtp jitter
buffer will synchronize with the clock signalled in the SDP offer provided the
sender is started with `--do-clock-signalling` & the receiver with
`--expect-clock-signalling`.

[RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500>
 2024-04-12 14:18:09 +02:00
François Laignel cc43935036 webrtc: add precise synchronization example 
This example demonstrates a sender / receiver setup which ensures precise
synchronization of multiple streams in a single session.

[RFC 6051]-style rapid synchronization of RTP streams is available as an option.
See the [Instantaneous RTP synchronization...] blog post for details about this
mode and an example based on RTSP instead of WebRTC.

[RFC 6051]: https://datatracker.ietf.org/doc/html/rfc6051
[Instantaneous RTP synchronization...]: https://coaxion.net/blog/2022/05/instantaneous-rtp-synchronization-retrieval-of-absolute-sender-clock-times-with-gstreamer/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1463>
 2024-04-03 19:10:40 +02:00
Guillaume Desmottes 35b84d219f webrtc: webrtcsink: set perfect-timestamp=true on audio encoders 
Chrome audio decoder doesn't cope well with not perfect ts, generating
noises in the audio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524>
 2024-04-02 14:25:51 +02:00
Sebastian Dröge b2d5ee48cd Update to async-tungstenite 0.25 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1455>
 2024-02-11 11:31:24 +02:00
Bilal Elmoussaoui 0615a16124 Use workspace features for crates metadata/deps 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1446>
 2024-02-05 15:34:31 +01:00
Sebastian Dröge 1a55c70114 Switch git dependencies to explicitly name branch 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
 2024-02-05 12:51:36 +02:00
Taruntej Kanakamalla 50e905fe4b webrtc: conditional compile for features with 1_22 dependency 
Few features being used in webrtcsink like
the signal `request-aux-sender` are introduced
to webrtcbin in gstreamer release 1.22.

Rename the feature gst1_22 to v1_22 for uniformity.

Add v1_22 to default features.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1241>
 2024-02-01 15:08:11 +05:30
Sebastian Dröge 4ad101b53b Use once_cell crate directly again 
The glib crate does not depend on it anymore and also does not re-export
it anymore.

Also switch some usages of OnceCell to OnceLock from std.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
 2024-01-31 18:07:57 +02:00
Sebastian Dröge 451d928026 webrtc: Update AWS signaller to http 1 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
 2024-01-31 18:07:57 +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
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
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 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
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
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
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
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
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 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
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 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
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 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
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
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 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 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
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
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
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
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
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
Mathieu Duponchelle a20855dfd9 webrtcsink: expose consumer-pipeline-created signal 
This signal is emitted as soon as the pipeline for each consumer
is created, and can be used by applications that require a greater
level of control over webrtcsink's internals.

An example is also provided to demonstrate usage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1220>
 2023-05-25 13:15:52 +02:00
Sebastian Dröge a27be7d054 net: Update to AWS SDK 0.28 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1224>
 2023-05-25 13:23:49 +03:00
Sebastian Dröge cb5b527d74 Update to AWS SDK 0.27 and async-tungstenite 0.22 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1199>
 2023-05-02 15:30:00 +03:00
Sebastian Dröge 5451035215 Update async-tungstenite and AWS SDK dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1187>
 2023-04-21 10:48:10 +00:00
Sebastian Dröge 315e53f064 webrtc: Update to AWS SDK 0.55/0.25 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1152>
 2023-03-31 09:12:26 +00:00
Sebastian Dröge c1bac30694 webrtc: Update to aws 0.54/0.24 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1131>
 2023-03-11 09:37:14 +02:00
Mathieu Duponchelle 584392049c net/webrtc: implement AWS KVS signaller 
And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink.

This adds support in webrtcsink for processing a consumer offer, instead
of producing one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
 2023-03-09 15:39:09 +00:00
Thibault Saunier ce3bb2f1d4 Add a webrtcsrc element 
Updating the docker image to include:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
 2023-02-28 20:50:15 -03:00
Sebastian Dröge 9fc1404415 Update minimum supported Rust version to 1.66 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
 2023-02-20 11:09:01 +02:00
Sebastian Dröge ac8afc4ac0 Update to async-tungstenite 0.20 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1087>
 2023-02-10 13:03:07 +02:00
Sebastian Dröge 1e13dbb99c Update versions to 0.11.0-alpha.1 2023-02-10 00:23:56 +02:00
Zhao, Gang 1ffeb4d44d webrtc: Move from async-std to tokio 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00