blob: d8002e6511acf6c8d8e55915cdf702b7f07da21d (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
|
/* -*- Mode: IDL; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/.
*
* The origin of this IDL file is
* http://lists.w3.org/Archives/Public/public-webrtc/2014May/0067.html
*/
[Pref="media.peerconnection.enabled",
Exposed=Window]
interface RTCRtpReceiver {
readonly attribute MediaStreamTrack track;
readonly attribute RTCDtlsTransport? transport;
static RTCRtpCapabilities? getCapabilities(DOMString kind);
sequence<RTCRtpContributingSource> getContributingSources();
sequence<RTCRtpSynchronizationSource> getSynchronizationSources();
[NewObject]
Promise<RTCStatsReport> getStats();
// test-only: for testing getContributingSources
[ChromeOnly]
undefined mozInsertAudioLevelForContributingSource(unsigned long source,
DOMHighResTimeStamp timestamp,
unsigned long rtpTimestamp,
boolean hasLevel,
byte level);
};
//https://w3c.github.io/webrtc-extensions/#rtcrtpreceiver-jitterbuffertarget-rtcrtpreceiver-interface
partial interface RTCRtpReceiver {
[Throws]
attribute DOMHighResTimeStamp? jitterBufferTarget;
};
// https://w3c.github.io/webrtc-encoded-transform/#specification
partial interface RTCRtpReceiver {
[SetterThrows,
Pref="media.peerconnection.scripttransform.enabled"] attribute RTCRtpTransform? transform;
};
|