Add API to request key frames #125
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This PR is based on w3c/webrtc-extensions#37, but has some differences. |
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Bot error should be fixed by #123 |
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Fixes #124 |
…t the first enqueued key frame since generateKeyFrame was called Also make the possible error checks synchronously instead of within the in parallel steps.
…oposal https://bugs.webkit.org/show_bug.cgi?id=234429 Reviewed by Eric Carlson. Source/WebCore: Update implementation according w3c/webrtc-encoded-transform#125. This means adding a specific method to send a FIR, a specific method to generate a key frame on sender side. This also means improving the error handling and the promise resolution timing. RID support is not yet available until we can properly pipe that information down to encoders. Test: http/wpt/webrtc/audiovideo-script-transform.html * Modules/mediastream/RTCRtpScriptTransformer.cpp: * Modules/mediastream/RTCRtpScriptTransformer.h: * Modules/mediastream/RTCRtpScriptTransformer.idl: LayoutTests: * http/wpt/webrtc/audio-video-transform.js: Added. * http/wpt/webrtc/audiovideo-script-transform-expected.txt: Added. * http/wpt/webrtc/audiovideo-script-transform.html: Added. * http/wpt/webrtc/context-transform.js: Canonical link: https://commits.webkit.org/245414@main git-svn-id: https://svn.webkit.org/repository/webkit/trunk@287258 268f45cc-cd09-0410-ab3c-d52691b4dbfc
…oposal https://bugs.webkit.org/show_bug.cgi?id=234429 Reviewed by Eric Carlson. Source/WebCore: Update implementation according w3c/webrtc-encoded-transform#125. This means adding a specific method to send a FIR, a specific method to generate a key frame on sender side. This also means improving the error handling and the promise resolution timing. RID support is not yet available until we can properly pipe that information down to encoders. Test: http/wpt/webrtc/audiovideo-script-transform.html * Modules/mediastream/RTCRtpScriptTransformer.cpp: * Modules/mediastream/RTCRtpScriptTransformer.h: * Modules/mediastream/RTCRtpScriptTransformer.idl: LayoutTests: * http/wpt/webrtc/audio-video-transform.js: Added. * http/wpt/webrtc/audiovideo-script-transform-expected.txt: Added. * http/wpt/webrtc/audiovideo-script-transform.html: Added. * http/wpt/webrtc/context-transform.js: git-svn-id: http://svn.webkit.org/repository/webkit/trunk@287258 268f45cc-cd09-0410-ab3c-d52691b4dbfc
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@youennf @alvestrand Do we understand how we can ensure key synchronization when a new participant joins the conference? In this situation, the sender may desire to generate a key frame and then apply a new encryption key (and keyId) to that keyframe. The updated key needs to be applied to the new keyframe, not to a previous or subsequent frame (which would lead to having the new participant being unable to decrypt and decode video until another keyframe update). If a keyframe is encrypted using the old encryption key, then a newly joined receiver will not be able to decrypt it. There are a number of ways in which the sender can generate a key frame and then ensure that a key update is applied to that newly generated key frame. The current PR does not make it clear how this would be achieved. One way this might work is to have generateKeyFrame() return a timestamp which is then provided to setEncryptionKey(). It may also be possible to avoid a race condition without passing a timestamp. |
This includes 3 different APIs:
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