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2024-01-18webrtchttp: Allow audio or video caps to be specified as None with WHEPHEADmainSanchayan Maity
We were setting audio and video caps by default even when the user might have requested only video or audio. This would then result in a `Could not reuse transceiver` error from the webrtcbin. Fix this by allowing the user to specify audio or video caps as None. This allows us to maintain the earlier behaviour for backward compatibility while allowing the user to not request audio or video as need be. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1433>
2024-01-18webrtc: Remove unnecessary manual `Send+Sync` implementations for signallersSebastian Dröge
These are automatically implemented. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/483 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18webrtc: Require `Send+Sync` for signaller implementationsSebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-17net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signallerEva Pace
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>
2024-01-17webrtcsrc: don't restrict RTP extensions to TWCC onlyMaksym Khomenko
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1381>
2024-01-16webrtc: Update to livekit-api / livekit-protocol 0.3Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
2024-01-16webrtcsink: make 'extensions' property usage conditionalMaksym Khomenko
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1423>
2024-01-16aws: s3: Disable remaining tests too for nowSebastian Dröge
They fail state changes, which cases `GstHarness` to abort.
2024-01-16aws: s3: Temporarily disable putobject testsArun Raghavan
Disabling while we figure out why it's failing. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1429>
2024-01-11Revert "aws: Disable putobjectsink tests for now"Arun Raghavan
This reverts commit b128d127c25e42f4908a096cad6829c6f60962b9. Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2024-01-11aws: putobjectsink: Fix a couple of minor log typosArun Raghavan
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2023-12-29webrtc/signalling: We get the address when acceptingNirbheek Chauhan
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29webrtc/signalling: Fix potential hang and FD leakNirbheek Chauhan
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>
2023-12-23webrtcsink: add payloader-setup signalMaksym Khomenko
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
2023-12-22aws: Disable putobjectsink tests for nowSebastian Dröge
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
2023-12-19aws: s3sink: Fix spelling of debug categoryArun Raghavan
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-19aws: s3putobjectsink: Add a flush-on-error propertyArun Raghavan
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>
2023-12-19aws: s3putobjectsink: Add some thresholds for flushingArun Raghavan
Lets us connect when we perform a flush Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18aws: s3: Add a new awss3putobjectsinkArun Raghavan
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>
2023-12-14webrtc: Downgrade aws-smithy-http to 0.60Sebastian Dröge
Version 0.61 was yanked from crates.io. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1407>
2023-12-09webrtc: Update to aws-smithy-http 0.61Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09rtp: Update to bitstream-io 2.0Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09Update to async-tungstenite 0.24Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-04use new debug and parse APIGuillaume Desmottes
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>
2023-11-26Update further AWS SDK crates to 1.0Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1400>
2023-11-24webrtcsink: don't panic on failure to request pad from webrtcbinMathieu Duponchelle
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>
2023-11-21Update to AWS SDK 1.0 / 0.60 / 0.39Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1397>
2023-11-20Update to AWS SDK 0.101 / 0.59 / 0.38Sebastian Dröge
2023-11-18aws: Stop using deprecated aws_config function in the testSebastian Dröge
2023-11-17net/webrtc: add whipserversrcTaruntej Kanakamalla
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>
2023-11-17net/webrtc: deprecate consumer-added on the signallerTaruntej Kanakamalla
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>
2023-11-17net/webrtc: rename WhipSignaller as WhipClientSignallerTaruntej Kanakamalla
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>
2023-11-17net/webrtc: Extract BaseWebRTCSrcTaruntej Kanakamalla
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>
2023-11-17Update to latest AWS SDKSebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1395>
2023-11-15Update to AWS SDK 0.36Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1394>
2023-11-13ndi: don't accumulate meta with audio only streamsFrançois Laignel
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>
2023-11-13ndisrc: Implement zerocopy handling for the received frames if possibleSebastian Dröge
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>
2023-11-13ndi: Don't mark private type as publicSebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13ndi: Refactor frame structs to have static lifetimesSebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13ndi: Remove wrong `Clone` impl on `RecvInstance`Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-09Revert "s3: tests: Remove emoji-based tests for now"Arun Raghavan
This reverts commit a49a5dcb11c736897ddadc04173cc46a5806aa00. 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>
2023-11-04webrtcsrc: add turn-servers propertyMaksym Khomenko
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1380>
2023-11-03Port to AWS SDK 0.57/0.35Mathieu Duponchelle
Co-authored-by: Sebastian Dröge <sebastian@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
2023-11-03aws: Update to test-with 0.12Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
2023-11-02Update for `gst::Rank` API changesSebastian Dröge
2023-10-30gstwebrtc-api: Patch webrtc-adapter to fix Safari behaviourPiotr Brzeziński
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>
2023-10-30Set sync=false in rsfilesink / s3sinkSebastian Dröge
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>
2023-10-30Use let-else instead of match for weak reference upgradesSebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1375>
2023-10-27Update to AWS SDK 0.34 and tracing-log 0.2Sebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1374>
2023-10-25hlssink3: Close the playlist giostreamsink on stop if possibleArun Raghavan
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>
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-20 21:41:58 +0300