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Add WebRTC and WebRTC encoded transform (#809)
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‎feature-group-definitions/webrtc-encoded-transform.dist.yml‎

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# Generated from: webrtc-encoded-transform.yml
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# Do not edit this file by hand. Edit the source file instead!
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name: WebRTC encoded transform
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description: The WebRTC encoded transform API allows you to modify audio and video streams in WebRTC connections. For example, it can be used for visual effects or custom codecs.
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spec: https://w3c.github.io/webrtc-encoded-transform/
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status:
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baseline: false
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support:
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firefox: "117"
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firefox_android: "117"
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safari: "15.4"
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safari_ios: "15.4"
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compat_features:
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- api.DedicatedWorkerGlobalScope.rtctransform_event
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- api.RTCEncodedAudioFrame
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- api.RTCEncodedAudioFrame.data
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- api.RTCEncodedAudioFrame.getMetadata
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- api.RTCEncodedAudioFrame.timestamp
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- api.RTCEncodedVideoFrame
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- api.RTCEncodedVideoFrame.data
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- api.RTCEncodedVideoFrame.getMetadata
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- api.RTCEncodedVideoFrame.timestamp
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- api.RTCEncodedVideoFrame.type
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- api.RTCRtpReceiver.transform
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- api.RTCRtpScriptTransform
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- api.RTCRtpScriptTransform.RTCRtpScriptTransform
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- api.RTCRtpScriptTransformer
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- api.RTCRtpScriptTransformer.generateKeyFrame
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- api.RTCRtpScriptTransformer.options
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- api.RTCRtpScriptTransformer.readable
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- api.RTCRtpScriptTransformer.sendKeyFrameRequest
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- api.RTCRtpScriptTransformer.writable
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- api.RTCRtpSender.transform
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- api.RTCTransformEvent
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- api.RTCTransformEvent.transformer

‎feature-group-definitions/webrtc-encoded-transform.yml‎

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name: WebRTC encoded transform
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description: The WebRTC encoded transform API allows you to modify audio and video streams in WebRTC connections. For example, it can be used for visual effects or custom codecs.
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spec: https://w3c.github.io/webrtc-encoded-transform/
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compat_features:
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- api.DedicatedWorkerGlobalScope.rtctransform_event
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- api.RTCEncodedAudioFrame
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- api.RTCEncodedAudioFrame.data
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- api.RTCEncodedAudioFrame.getMetadata
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- api.RTCEncodedAudioFrame.timestamp
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- api.RTCEncodedVideoFrame
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- api.RTCEncodedVideoFrame.data
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- api.RTCEncodedVideoFrame.getMetadata
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- api.RTCEncodedVideoFrame.timestamp
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- api.RTCEncodedVideoFrame.type
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- api.RTCRtpReceiver.transform
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- api.RTCRtpScriptTransform
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- api.RTCRtpScriptTransform.RTCRtpScriptTransform
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- api.RTCRtpScriptTransformer
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- api.RTCRtpScriptTransformer.generateKeyFrame
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- api.RTCRtpScriptTransformer.options
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- api.RTCRtpScriptTransformer.readable
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- api.RTCRtpScriptTransformer.sendKeyFrameRequest
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- api.RTCRtpScriptTransformer.writable
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- api.RTCRtpSender.transform
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- api.RTCTransformEvent
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- api.RTCTransformEvent.transformer

‎feature-group-definitions/webrtc.dist.yml‎

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# Generated from: webrtc.yml
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# Do not edit this file by hand. Edit the source file instead!
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name: WebRTC
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description: The WebRTC API establishes real-time communication channels directly between browsers. It is commonly used in video conferencing applications.
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spec: https://w3c.github.io/webrtc-pc/
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caniuse: rtcpeerconnection
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status:
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baseline: high
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baseline_low_date: 2020-01-15
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baseline_high_date: 2022-07-15
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support:
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chrome: "56"
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chrome_android: "56"
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edge: "79"
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firefox: "44"
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firefox_android: "44"
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safari: "11"
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safari_ios: "11"
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# The WebRTC API is huge and has been evolving for over a decade. These are
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# central interfaces that result in the right computed status (matching caniuse)
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# but this is <1% complete:
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# https://github.com/web-platform-dx/web-features/issues/819
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compat_features:
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- api.RTCPeerConnection
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- api.RTCDataChannel

‎feature-group-definitions/webrtc.yml‎

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name: WebRTC
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description: The WebRTC API establishes real-time communication channels directly between browsers. It is commonly used in video conferencing applications.
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spec: https://w3c.github.io/webrtc-pc/
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caniuse: rtcpeerconnection
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# The WebRTC API is huge and has been evolving for over a decade. These are
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# central interfaces that result in the right computed status (matching caniuse)
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# but this is <1% complete:
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# https://github.com/web-platform-dx/web-features/issues/819
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compat_features:
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- api.RTCPeerConnection
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- api.RTCDataChannel

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