chromium/src/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.idl

// Copyright 2017 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

// https://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
[Exposed=Window]
interface RTCRtpSender {
    readonly attribute MediaStreamTrack? track;
    readonly attribute RTCDtlsTransport? transport;
    readonly attribute RTCDtlsTransport? rtcpTransport;
    [CallWith=ScriptState, HighEntropy, Measure] static RTCRtpCapabilities? getCapabilities(DOMString kind);
    [CallWith=ScriptState] Promise<void> setParameters(RTCRtpSendParameters parameters);
    RTCRtpSendParameters getParameters();
    [Measure, CallWith=ScriptState] Promise<void> replaceTrack(MediaStreamTrack? withTrack);
    [Measure] readonly attribute RTCDTMFSender? dtmf;
    [Measure, RaisesException] void setStreams(MediaStream... streams);
    [CallWith=ScriptState] Promise<RTCStatsReport> getStats();
    [Measure,
     CallWith=ScriptState,
     RaisesException] RTCInsertableStreams createEncodedStreams();
    [DeprecateAs=V8RTCRtpSender_CreateEncodedAudioStreams_Method,
     RuntimeEnabled=RTCInsertableStreams,
     CallWith=ScriptState,
     RaisesException] RTCInsertableStreams createEncodedAudioStreams();
    [DeprecateAs=V8RTCRtpSender_CreateEncodedVideoStreams_Method,
     RuntimeEnabled=RTCInsertableStreams,
     CallWith=ScriptState,
     RaisesException] RTCInsertableStreams createEncodedVideoStreams();
};