@webex/web-client-media-engine 3.20.4 → 3.22.0
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/dist/cjs/index.js +1395 -1343
- package/dist/cjs/index.js.map +1 -1
- package/dist/esm/index.js +1395 -1343
- package/dist/esm/index.js.map +1 -1
- package/dist/types/index.d.ts +19 -12
- package/package.json +2 -2
package/dist/types/index.d.ts
CHANGED
|
@@ -1,5 +1,5 @@
|
|
|
1
|
-
import { LocalStream, ConnectionState, media } from '@webex/webrtc-core';
|
|
2
|
-
export { AudioDeviceConstraints, ConnectionState, LocalCameraStream, LocalDisplayStream, LocalMicrophoneStream, LocalStream, LocalStreamEventNames, LocalSystemAudioStream, MediaStreamTrackKind, PeerConnection, RemoteMediaState, RemoteStream, RemoteStreamEventNames, StreamEventNames, VideoContentHint, VideoDeviceConstraints, Logger as WebRtcCoreLogger, WebrtcCoreError, WebrtcCoreErrorType, createCameraStream, createDisplayStream, createDisplayStreamWithAudio, createMicrophoneStream, getAudioInputDevices, getAudioOutputDevices, getDevices, getVideoInputDevices, setOnDeviceChangeHandler } from '@webex/webrtc-core';
|
|
1
|
+
import { LocalStream, ConnectionState, ConnectionType, media } from '@webex/webrtc-core';
|
|
2
|
+
export { AudioDeviceConstraints, ConnectionState, ConnectionType, LocalCameraStream, LocalDisplayStream, LocalMicrophoneStream, LocalStream, LocalStreamEventNames, LocalSystemAudioStream, MediaStreamTrackKind, PeerConnection, RemoteMediaState, RemoteStream, RemoteStreamEventNames, StreamEventNames, VideoContentHint, VideoDeviceConstraints, Logger as WebRtcCoreLogger, WebrtcCoreError, WebrtcCoreErrorType, createCameraStream, createDisplayStream, createDisplayStreamWithAudio, createMicrophoneStream, getAudioInputDevices, getAudioOutputDevices, getDevices, getVideoInputDevices, setOnDeviceChangeHandler } from '@webex/webrtc-core';
|
|
3
3
|
import { StreamId, StreamState, MediaContent, NamedMediaGroup, MediaType, Policy, PolicySpecificInfo, CodecInfo, StreamRequest as StreamRequest$1 } from '@webex/json-multistream';
|
|
4
4
|
export { ActiveSpeakerInfo, CodecInfo, H264Codec, Logger as JMPLogger, MediaContent, MediaFamily, MediaType, NamedMediaGroup, Policy, PolicySpecificInfo, ReceiverSelectedInfo, StreamState, getMediaContent, getMediaFamily, getMediaType } from '@webex/json-multistream';
|
|
5
5
|
import { LogData } from '@webex/rtcstats';
|
|
@@ -15,7 +15,7 @@ declare enum MediaCodecMimeType {
|
|
|
15
15
|
AV1 = "video/AV1",
|
|
16
16
|
OPUS = "audio/opus"
|
|
17
17
|
}
|
|
18
|
-
|
|
18
|
+
type EncodingParams = {
|
|
19
19
|
maxPayloadBitsPerSecond?: number;
|
|
20
20
|
maxFs?: number;
|
|
21
21
|
maxWidth?: number;
|
|
@@ -58,9 +58,9 @@ declare enum OveruseState {
|
|
|
58
58
|
OVERUSED = 1
|
|
59
59
|
}
|
|
60
60
|
|
|
61
|
-
|
|
61
|
+
type OveruseUpdateCallback = (state: OveruseState) => void;
|
|
62
62
|
|
|
63
|
-
|
|
63
|
+
type ReceiveSlotId = StreamId;
|
|
64
64
|
declare function compareReceiveSlotIds(id1: ReceiveSlotId, id2: ReceiveSlotId): boolean;
|
|
65
65
|
declare enum ReceiveSlotEvents {
|
|
66
66
|
MediaStarted = "media-started",
|
|
@@ -110,7 +110,7 @@ declare class EgressSdpMunger {
|
|
|
110
110
|
deleteCodecParameters(parameters: string[]): void;
|
|
111
111
|
}
|
|
112
112
|
|
|
113
|
-
|
|
113
|
+
type StatsMap = Map<string, any>;
|
|
114
114
|
|
|
115
115
|
declare abstract class Transceiver {
|
|
116
116
|
protected _rtcRtpTransceiver: RTCRtpTransceiver;
|
|
@@ -225,35 +225,39 @@ interface TransceiverStats {
|
|
|
225
225
|
};
|
|
226
226
|
}
|
|
227
227
|
|
|
228
|
-
|
|
228
|
+
type BundlePolicy = 'max-bundle' | 'max-compat';
|
|
229
229
|
|
|
230
|
-
|
|
230
|
+
type MetricsCallback = (logData: LogData) => void;
|
|
231
231
|
|
|
232
232
|
declare enum MultistreamConnectionEventNames {
|
|
233
233
|
VideoSourceCountUpdate = "video-source-count-update",
|
|
234
234
|
AudioSourceCountUpdate = "audio-source-count-update",
|
|
235
235
|
ActiveSpeakerNotification = "active-speaker-notification",
|
|
236
|
-
|
|
236
|
+
PeerConnectionStateUpdate = "peer-connection-state-update",
|
|
237
|
+
IceConnectionStateUpdate = "ice-connection-state-update",
|
|
237
238
|
IceGatheringStateUpdate = "ice-gathering-state-update",
|
|
238
239
|
NegotiationNeeded = "negotiation-needed",
|
|
239
240
|
CreateOfferOnSuccess = "createofferonsuccess",
|
|
240
241
|
CreateAnswerOnSuccess = "createansweronsuccess",
|
|
241
242
|
SetLocalDescriptionOnSuccess = "setlocaldescriptiononsuccess",
|
|
242
|
-
SetRemoteDescriptionOnSuccess = "setremotedescriptiononsuccess"
|
|
243
|
+
SetRemoteDescriptionOnSuccess = "setremotedescriptiononsuccess",
|
|
244
|
+
IceCandidateError = "icecandidateerror"
|
|
243
245
|
}
|
|
244
246
|
interface MultistreamConnectionEvents extends EventMap {
|
|
245
247
|
[MultistreamConnectionEventNames.ActiveSpeakerNotification]: (csis: number[]) => void;
|
|
246
248
|
[MultistreamConnectionEventNames.VideoSourceCountUpdate]: (numTotalSources: number, numLiveSources: number, mediaContent: MediaContent) => void;
|
|
247
249
|
[MultistreamConnectionEventNames.AudioSourceCountUpdate]: (numTotalSources: number, numLiveSources: number, mediaContent: MediaContent) => void;
|
|
248
|
-
[MultistreamConnectionEventNames.
|
|
250
|
+
[MultistreamConnectionEventNames.PeerConnectionStateUpdate]: (state: RTCPeerConnectionState) => void;
|
|
251
|
+
[MultistreamConnectionEventNames.IceConnectionStateUpdate]: (state: RTCIceConnectionState) => void;
|
|
249
252
|
[MultistreamConnectionEventNames.IceGatheringStateUpdate]: (state: RTCIceGatheringState) => void;
|
|
250
253
|
[MultistreamConnectionEventNames.NegotiationNeeded]: () => void;
|
|
251
254
|
[MultistreamConnectionEventNames.CreateAnswerOnSuccess]: (answer: RTCSessionDescriptionInit) => void;
|
|
252
255
|
[MultistreamConnectionEventNames.CreateOfferOnSuccess]: (offer: RTCSessionDescriptionInit) => void;
|
|
253
256
|
[MultistreamConnectionEventNames.SetLocalDescriptionOnSuccess]: (description: RTCSessionDescriptionInit) => void;
|
|
254
257
|
[MultistreamConnectionEventNames.SetRemoteDescriptionOnSuccess]: (description: RTCSessionDescriptionInit) => void;
|
|
258
|
+
[MultistreamConnectionEventNames.IceCandidateError]: (error: RTCPeerConnectionIceErrorEvent) => void;
|
|
255
259
|
}
|
|
256
|
-
|
|
260
|
+
type MultistreamConnectionOptions = {
|
|
257
261
|
disableSimulcast: boolean;
|
|
258
262
|
bundlePolicy: BundlePolicy;
|
|
259
263
|
iceServers: RTCIceServer[] | undefined;
|
|
@@ -281,6 +285,9 @@ declare class MultistreamConnection extends EventEmitter<MultistreamConnectionEv
|
|
|
281
285
|
private initializePeerConnection;
|
|
282
286
|
private propagatePeerConnectionEvents;
|
|
283
287
|
getConnectionState(): ConnectionState;
|
|
288
|
+
getPeerConnectionState(): RTCPeerConnectionState;
|
|
289
|
+
getIceConnectionState(): RTCIceConnectionState;
|
|
290
|
+
getCurrentConnectionType(): Promise<ConnectionType>;
|
|
284
291
|
getIceGatheringState(): RTCIceGatheringState;
|
|
285
292
|
private getVideoEncodingOptions;
|
|
286
293
|
private createSendTransceiver;
|
package/package.json
CHANGED
|
@@ -1,6 +1,6 @@
|
|
|
1
1
|
{
|
|
2
2
|
"name": "@webex/web-client-media-engine",
|
|
3
|
-
"version": "3.
|
|
3
|
+
"version": "3.22.0",
|
|
4
4
|
"description": "Web Client Media Engine is common web code for interacting with the multistream media server.",
|
|
5
5
|
"source": "src/index.ts",
|
|
6
6
|
"main": "dist/cjs/index.js",
|
|
@@ -60,7 +60,7 @@
|
|
|
60
60
|
"@webex/ts-events": "^1.0.1",
|
|
61
61
|
"@webex/ts-sdp": "1.6.0",
|
|
62
62
|
"@webex/web-capabilities": "^1.3.0",
|
|
63
|
-
"@webex/webrtc-core": "2.
|
|
63
|
+
"@webex/webrtc-core": "2.10.0",
|
|
64
64
|
"@webex/web-media-effects": "^2.15.6",
|
|
65
65
|
"async": "^3.2.4",
|
|
66
66
|
"js-logger": "^1.6.1",
|