@vpalmisano/webrtcperf-js
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    Variable paramsConst

    params: {
        absCaptureTime: EnableValue;
        actions: Action[];
        drawWatermarkGrid: boolean;
        enableVideoStats: EnableValue;
        fakeScreenshare: null | FakeScreenshareParams;
        getDisplayMediaWaitTime: number;
        getUserMediaWaitTime: number;
        jitterBufferTarget:
            | null
            | number
            | { audio: null
            | number; video: null | number };
        peerConnectionDebug: EnableValue;
        playoutDelayHint: null | number;
        saveAudioTrackEnableEnd: number;
        saveAudioTrackEnableStart: number;
        saveRecvAudioTrack: EnableValue;
        saveRecvVideoTrack: EnableValue;
        saveSendAudioTrack: EnableValue;
        saveSendVideoTrack: EnableValue;
        saveVideoTrackEnableEnd: number;
        saveVideoTrackEnableStart: number;
        timestampInsertableStreams: boolean;
        timestampWatermarkAudio: EnableValue;
        timestampWatermarkVideo: EnableValue;
    } = ...

    Parameters for the webrtcperf tool.

    Type declaration

    • absCaptureTime: EnableValue

      If set, the RTCPeerConnection offer will be modified to include the abs-capture-time extension.

    • actions: Action[]

      List of actions to perform.

    • drawWatermarkGrid: boolean

      It set, a grid will be drawn on the video track.

    • enableVideoStats: EnableValue

      Enable video stats.

    • fakeScreenshare: null | FakeScreenshareParams

      It set, the fake screenshare will be created with the specified parameters.

    • getDisplayMediaWaitTime: number

      It set, the getDisplayMedia will wait for the specified time before returning.

    • getUserMediaWaitTime: number

      It set, the getUserMedia will wait for the specified time before returning.

    • jitterBufferTarget: null | number | { audio: null | number; video: null | number }

      If set, all the created RTCRtpReceivers will be configured with the specified jitter buffer target (in seconds). It can be configured for each track kind or bo

    • peerConnectionDebug: EnableValue

      It set, the peer connection will run with additional debug logs.

    • playoutDelayHint: null | number

      If set, all the created RTCRtpReceivers will be configured with the specified playout delay hint (in seconds).

    • saveAudioTrackEnableEnd: number

      The time in milliseconds after which the RTCPeerConnection sent audio track will be disabled.

    • saveAudioTrackEnableStart: number

      The time in milliseconds after which the RTCPeerConnection sent audio track will be enabled.

    • saveRecvAudioTrack: EnableValue

      If set, the RTCPeerConnection received audio tracks will be saved.

    • saveRecvVideoTrack: EnableValue

      It set, the RTCPeerConnection received video tracks will be saved.

    • saveSendAudioTrack: EnableValue

      It set, the RTCPeerConnection sent audio tracks will be saved.

    • saveSendVideoTrack: EnableValue

      It set, the RTCPeerConnection sent video tracks will be saved.

    • saveVideoTrackEnableEnd: number

      The time in milliseconds after which the RTCPeerConnection sent video track will be disabled.

    • saveVideoTrackEnableStart: number

      The time in milliseconds after which the RTCPeerConnection sent video track will be enabled.

    • timestampInsertableStreams: boolean
    • timestampWatermarkAudio: EnableValue

      It set, a watermark with the current timestamp will be added to the sent audio tracks. It will recognize the watermark on the received audio tracks and collect the delay into the audioEndToEndDelayStats object.

    • timestampWatermarkVideo: EnableValue

      It set, a watermark with the current timestamp will be added to the sent video tracks. It will recognize the watermark on the received video tracks and collect the delay into the videoEndToEndDelayStats object.

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