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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>
2023-12-23webrtcsink: add payloader-setup signalMaksym Khomenko
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
2023-12-22Update plugin documentation cacheSebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
2023-12-19aws: Add plugin docsArun Raghavan
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
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: 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>
2023-11-04webrtcsrc: add turn-servers propertyMaksym Khomenko
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1380>
2023-10-12net/webrtcsrc: define signaller property as CONSTRUCT_ONLYFrançois Laignel
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>
2023-10-04fmp4mux: specify the fragment duration unitStéphane Cerveau
The fragment duration is expressed in nanoseconds. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1348>
2023-10-03Update plugins cache for updated raw video capsSebastian Dröge
2023-09-25hlssink3: Update plugin docsSeungha Yang
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-20hlssink3: Add property track-pipeline-clock-for-pdt.rajneeshksoni
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>
2023-09-20hlssink3: Allow adding EXT-X-PROGRAM-DATE-TIME tag.rajneeshksoni
- 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>
2023-09-20gtk4: Only support RGBA/RGB in the GL code pathSebastian Dröge
For all other component orderings a shader is necessary to re-order the components for what GTK expects. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1312>
2023-09-14webrtcsink: Add support for d3d11 memory and qsvh264encSeungha Yang
Adding d3d11 memory and qsvh264enc support Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1322>
2023-08-29Update docs for new order of raw video formatsSebastian Dröge
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
2023-08-26net/webrtc: rename whipwebrtcsink as whipclientsinkTaruntej Kanakamalla
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>
2023-08-14generic: expose inter pluginMathieu Duponchelle
This new plugin exposes two elements, intersink and intersrc. These act as wormholes for data in the same process and can be used to forward data from one pipeline to another. The implementation makes use of gstreamer-utils' StreamProducer, and supports dynamically adding and removing consumers, before and after producers, and changing producer names while PLAYING, both on the sink and the src. This initial implementation comes with a small demo, and a few tests. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1257>
2023-08-10webrtcsrc: add support for navigation eventsAndoni Morales Alastruey
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>
2023-08-10net/webrtc/signaller: advertise running producers in Listener modeLoïc Le Page
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>
2023-07-06Update docs after GStreamer update on the CISebastian Dröge
2023-07-06webrtcsink: Add LiveKit WebRTC sink and signallerOlivier Crête
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-06-16webrtcsink: bring in signalling code from whipsink as a signallerMathieu Duponchelle
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-06-14togglerecord: Add support for non-live inputsVivia Nikolaidou
Live input + is-live=false: While not recording, drop input When recording is started, offset to collapse the gap Live input + is-live=true: While not recording, drop input Don't modify the offset Non-live input + is-live=false: While not recording, block input Don't modify the offset Non-live input + is-live=true: While not recording, block input When recording is started, offset to current running time Co-authored-by: Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1206>
2023-06-13fallbackswitch: add 'stop-on-eos' propertyGuillaume Desmottes
Fix the following use case: - main input of fallbackswitch is finite (a media file) - fallback input is infinite (videotestsrc) - main input is done and send eos, which is propagated downstream - fallbackswitch switches to fallback, sending STREAM_START which reset EOS downstream (aggregator does that) - fallback input keeps pushing buffers forever. Solve it by adding a 'stop-on-eos' property so fallbackswitch stops pushing property once the main input is eos. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1242>
2023-06-07Use MPL as license specifier for plugins only requiring GStreamer < 1.20Sebastian Dröge
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>
2023-06-06webrtcsink: encoded streams: address last review commentsMathieu Duponchelle
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-05-25webrtcsink: expose consumer-pipeline-created signalMathieu Duponchelle
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>
2023-05-15fallbackswitch: document the pad priority orderingGuillaume Desmottes
I just wasted lots of time trying to figure out why my higher priority pad wasn't used... Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1208>
2023-05-10transcriberbin: Add "language-code" propertySeungha Yang
Proxy the child transcriber element's property so that transcriberbin can apply the property with required state management Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1205>
2023-05-09net/webrtc: src: add signal "request-encoded-filter"François Laignel
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-04-13ffv1dec: Drop rank for nowArun Raghavan
We'll keep the rank lower than avdec_ffv1, at least until we're comfortable with support for the entire range of possible inputs working well. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1174>
2023-04-13webrtc: extract a BaseWebRTCSinkMathieu Duponchelle
For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-11tttocea608: specify raw 608 fieldMathieu Duponchelle
The element can only output field=0 raw 608 data. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1166>
2023-04-10fix typosGuillaume Desmottes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
2023-04-10webrtc: signaller iface: fix session-ended vs end-session confusionMathieu Duponchelle
Session ending is bidirectional: the signaller can tell the sink that a session was ended, and the sink can tell the signaller to end a session. As such, two signals are needed, before this patch the second case was not working as in essence the sink was telling itself that a session was ended, and obviously failing to even find it when trying to end it again. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
2023-04-07transcriberbin: Allow video with ANY caps featuresSeungha Yang
transcriberbin does not read/write video buffers actually. Allow ANY caps features in order to avoid unnecessary GPU upload/download Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1165>
2023-04-07webrtcsink: update docs for property and signal changesMatthew Waters
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06transcriberbin: Add support for runtime translation-languages updateSeungha Yang
Allows updating translation-languages at runtime Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1162>
2023-04-05closedcaption: add cea608tocea708 elementMatthew Waters
Implement an element that can take an input 608 caption stream and generate a valid 708 caption stream by parsing the 608 data and generating the equivalent DTVCCPackets and Service blocks. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1112>
2023-03-29transcriberbin: add support for translationsMathieu Duponchelle
With this, if the transcriber element in use supports "translation_src_" request source pads, the user can now specify what languages to translate to and how to map them to 608 channels (only CC1 and CC3 are supported). For instance, translation-languages="languages, CC3=transcript, CC1=fr" will cause the original transcript to be muxed into the CC3 channel, and the French translation to be muxed into the CC1 channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1149>
2023-03-28chore(webrtcsink): fix max-bitrate blurb and nickDavid Revay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1150>
2023-03-27webrtcsink: Add ice-transport-policy optionVivia Nikolaidou
Can be used to force relay ICE candidates, ensuring TURN server is used. Proxy to the corresponding setting in webrtcbin, Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1143>
2023-03-16net/aws/transcriber: fix translate lookaheadFrançois Laignel
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-16net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPadFrançois Laignel
... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-14net/aws/transcriber: translate: optional experimental translation tokenizationFrançois Laignel
This commit adds an optional experimental translation tokenization feature. It can be activated using the `translation_src_%u` pads property `tokenization-method`. For the moment, the feature is deactivated by default. The Translate ws accepts '<span></span>' tags in the input and adds matching tags in the output. When an 'id' is also provided as an attribute of the 'span', the matching output tag also uses this 'id'. In the context of close captions, the 'id's are of little use. However, we can take advantage of the spans in the output to identify translation chunks, which more or less reflect the rythm of the input transcript. This commit adds simples spans (no 'id') to the input Transcript Items and parses the resulting spans in the translated output, assigning the timestamps and durations sequentially from the input Transcript Items. Edge cases such as absence of spans, nested spans were observed and are handled here. Similarly, mismatches between the number of input and output items are taken care of by some sort of reconcialiation. Note that this is still experimental and requires further testings. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-14net/aws/transcriber: add translation request src padsFrançois Laignel
This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-09net/webrtc: implement AWS KVS signallerMathieu Duponchelle
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>
2023-03-01net/aws: use aws-sdk-transcribestreamingFrançois Laignel
Switch from manual webservice client impl to `aws-sdk-transcribestreaming`. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-20 23:32:46 +0300