firefox-main/dom/media/webrtc/jsapi
Name Description Size Coverage
DefaultCodecPreferences.cpp 5423 96 %
DefaultCodecPreferences.h 5452 96 %
MediaTransportHandler.cpp 90838 85 %
MediaTransportHandler.h 11493 92 %
MediaTransportHandlerIPC.cpp dummy 17393 74 %
MediaTransportHandlerIPC.h 4019 100 %
MediaTransportParent.cpp 11402 81 %
moz.build 2532 -
PacketDumper.cpp static 5192 99 %
PacketDumper.h 1782 100 %
PeerConnectionCtx.cpp 21873 80 %
PeerConnectionCtx.h Refcounted class containing state shared across all PeerConnections and all Call instances. Managed by PeerConnectionCtx, and kept around while there are registered peer connections. 6664 96 %
PeerConnectionImpl.cpp 178792 87 %
PeerConnectionImpl.h do/while prevents res from conflicting with locals 34958 94 %
RemoteTrackSource.cpp 3521 73 %
RemoteTrackSource.h 1882 89 %
RTCDataChannel.cpp virtual 32397 80 %
RTCDataChannel.h 6111 82 %
RTCDataChannelDeclarations.h 1102 -
RTCDtlsTransport.cpp 3830 82 %
RTCDtlsTransport.h 1791 71 %
RTCDTMFSender.cpp 6969 94 %
RTCDTMFSender.h 2294 89 %
RTCEncodedAudioFrame.cpp static 7198 0 %
RTCEncodedAudioFrame.h 3100 0 %
RTCEncodedFrameBase.cpp 6957 0 %
RTCEncodedFrameBase.h 3833 0 %
RTCEncodedVideoFrame.cpp static 8091 0 %
RTCEncodedVideoFrame.h 3502 0 %
RTCError.cpp static 1428 80 %
RTCError.h 1474 17 %
RTCIceCandidate.cpp static 7768 54 %
RTCIceCandidate.h 3845 100 %
RTCIceCandidatePair.cpp 1206 67 %
RTCIceCandidatePair.h 1435 80 %
RTCIceTransport.cpp 2529 96 %
RTCIceTransport.h 2122 56 %
RTCRtpReceiver.cpp 52545 88 %
RTCRtpReceiver.h 8869 81 %
RTCRtpScriptTransform.cpp 3476 21 %
RTCRtpScriptTransform.h 1771 17 %
RTCRtpScriptTransformer.cpp 16059 0 %
RTCRtpScriptTransformer.h 7497 0 %
RTCRtpSender.cpp 92404 76 %
RTCRtpSender.h 11310 98 %
RTCRtpTransceiver.cpp 53650 89 %
RTCRtpTransceiver.h This is what ties all the various pieces that make up a transceiver together. This includes: MediaStreamTrack for rendering and capture MediaTransportHandler for RTP transmission/reception Audio/VideoConduit for feeding RTP/RTCP into webrtc.org for decoding, and feeding audio/video frames into webrtc.org for encoding into RTP/RTCP. 11032 97 %
RTCSctpTransport.cpp 2086 88 %
RTCSctpTransport.h 2104 76 %
RTCStatsIdGenerator.cpp 3748 96 %
RTCStatsIdGenerator.h 970 100 %
RTCStatsReport.cpp static 9042 90 %
RTCStatsReport.h Keeps the state needed to convert RTCStatsTimestamps. 7877 88 %
RTCTransformEventRunnable.cpp 2957 0 %
RTCTransformEventRunnable.h 1203 0 %
WebrtcGlobalChild.h 1327 100 %
WebrtcGlobalInformation.cpp 26639 83 %
WebrtcGlobalInformation.h 3362 100 %
WebrtcGlobalParent.h 1845 100 %
WebrtcGlobalStatsHistory.cpp 9054 87 %
WebrtcGlobalStatsHistory.h 3117 100 %