@ikonai/sdk 1.0.22 → 1.0.24

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@ikonai/sdk",
-  "version": "1.0.22",
+  "version": "1.0.24",
   "type": "module",
   "main": "./index.js",
   "types": "./index.d.ts",

package/webrtc/index.d.ts

@@ -1,3 +1,3 @@
 export { WebRTCConnection, type WebRTCConnectionConfig, type WebRTCConnectionState } from './webrtc-connection';
 export { WebRTCSignaling, type WebRTCSignalingConfig } from './webrtc-signaling';
-export { WEBRTC_AUDIO_CODEC, WEBRTC_AUDIO_SAMPLE_RATE, WEBRTC_AUDIO_CHANNELS, WEBRTC_AUDIO_BITRATE, WEBRTC_AUDIO_JITTER_BUFFER_TARGET_MS, WEBRTC_VIDEO_CODEC, WEBRTC_VIDEO_MAX_BITRATE, WEBRTC_VIDEO_MAX_FRAMERATE, WEBRTC_BUNDLE_POLICY, WEBRTC_RTCP_MUX_POLICY, WEBRTC_ICE_SERVERS, WEBRTC_DATA_CHANNEL_LABEL, WEBRTC_DATA_CHANNEL_ORDERED, WEBRTC_DATA_CHANNEL_MAX_RETRANSMITS, } from './webrtc-config';
+export { WEBRTC_AUDIO_CODEC, WEBRTC_AUDIO_SAMPLE_RATE, WEBRTC_AUDIO_CHANNELS, WEBRTC_AUDIO_BITRATE, WEBRTC_AUDIO_JITTER_BUFFER_TARGET_MS, WEBRTC_VIDEO_CODEC, WEBRTC_VIDEO_MAX_BITRATE, WEBRTC_VIDEO_MAX_FRAMERATE, WEBRTC_MAX_AUDIO_TRACKS, WEBRTC_MAX_VIDEO_TRACKS, WEBRTC_BUNDLE_POLICY, WEBRTC_RTCP_MUX_POLICY, WEBRTC_ICE_SERVERS, WEBRTC_DATA_CHANNEL_LABEL, WEBRTC_DATA_CHANNEL_ORDERED, WEBRTC_DATA_CHANNEL_MAX_RETRANSMITS, } from './webrtc-config';

package/webrtc/webrtc-config.d.ts

@@ -3,12 +3,15 @@ export declare const WEBRTC_AUDIO_SAMPLE_RATE = 48000;
 export declare const WEBRTC_AUDIO_CHANNELS = 2;
 export declare const WEBRTC_AUDIO_BITRATE = 64000;
 export declare const WEBRTC_AUDIO_JITTER_BUFFER_TARGET_MS = 150;
-export declare const WEBRTC_VIDEO_CODEC = "H264";
+export declare const WEBRTC_VIDEO_CODEC = "VP8";
 export declare const WEBRTC_VIDEO_MAX_BITRATE = 5000000;
 export declare const WEBRTC_VIDEO_MAX_FRAMERATE = 30;
+export declare const WEBRTC_VIDEO_PLAYOUT_DELAY_HINT_S = 0.05;
 export declare const WEBRTC_BUNDLE_POLICY: RTCBundlePolicy;
 export declare const WEBRTC_RTCP_MUX_POLICY: RTCRtcpMuxPolicy;
 export declare const WEBRTC_ICE_SERVERS: RTCIceServer[];
+export declare const WEBRTC_MAX_AUDIO_TRACKS = 16;
+export declare const WEBRTC_MAX_VIDEO_TRACKS = 16;
 export declare const WEBRTC_DATA_CHANNEL_LABEL = "ikon-data";
 export declare const WEBRTC_DATA_CHANNEL_ORDERED = false;
 export declare const WEBRTC_DATA_CHANNEL_MAX_RETRANSMITS = 0;

package/webrtc/webrtc-connection.d.ts

@@ -8,7 +8,7 @@ export interface WebRTCConnectionConfig {
     onDataChannelMessage?: (data: ArrayBuffer) => void;
     onDataChannelOpen?: () => void;
     onDataChannelClose?: () => void;
-    onAudioTrack?: (track: MediaStreamTrack, stream: MediaStream) => void;
+    onAudioTrack?: (trackIndex: number, track: MediaStreamTrack, stream: MediaStream) => void;
     onVideoTrack?: (trackIndex: number, track: MediaStreamTrack, stream: MediaStream) => void;
 }
 export declare class WebRTCConnection {
@@ -18,7 +18,7 @@ export declare class WebRTCConnection {
     private readonly config;
     private hasRemoteDescription;
     private pendingCandidates;
-    private videoTransceivers;
+    private audioTrackCount;
     private videoTrackCount;
     constructor(config: WebRTCConnectionConfig);
     get state(): WebRTCConnectionState;
@@ -29,7 +29,7 @@ export declare class WebRTCConnection {
     setRemoteDescription(sdp: string, type: 'answer' | 'offer'): Promise<void>;
     addIceCandidate(candidate: RTCIceCandidateInit): Promise<void>;
     addTrack(track: MediaStreamTrack, stream: MediaStream): RTCRtpSender;
-    replaceTrack(kind: 'audio' | 'video', track: MediaStreamTrack | null, videoTrackIndex?: number): Promise<RTCRtpSender | null>;
+    replaceTrack(kind: 'audio' | 'video', track: MediaStreamTrack | null, trackIndex?: number): Promise<RTCRtpSender | null>;
     removeTrack(sender: RTCRtpSender): void;
     sendDataChannelMessage(data: ArrayBuffer): void;
     close(): void;

package/webrtc/webrtc-signaling.d.ts

@@ -1,4 +1,4 @@
-import { ProtocolMessage, Opcode } from '../../../../shared/protocol/src/index.ts';
+import { ProtocolMessage, Opcode, WebRTCTrackMap } from '../../../../shared/protocol/src/index.ts';
 import { WebRTCConnectionState } from './webrtc-connection';
 export interface WebRTCSignalingConfig {
     sessionId: number;
@@ -11,15 +11,17 @@ export interface WebRTCSignalingConfig {
     onDataChannelMessage?: (data: ArrayBuffer) => void;
     onDataChannelOpen?: () => void;
     onDataChannelClose?: () => void;
-    onAudioTrack?: (track: MediaStreamTrack, stream: MediaStream) => void;
+    onAudioTrack?: (trackIndex: number, track: MediaStreamTrack, stream: MediaStream) => void;
     onVideoTrack?: (trackIndex: number, track: MediaStreamTrack, stream: MediaStream) => void;
+    onTrackMap?: (trackMap: WebRTCTrackMap) => void;
 }
 export declare class WebRTCSignaling {
     private readonly config;
     private connection;
     private sendMessage;
     private isConnecting;
-    private statsInterval;
+    private audioTrackCount;
+    private videoTrackCount;
     constructor(config: WebRTCSignalingConfig);
     get state(): WebRTCConnectionState;
     get peerConnection(): RTCPeerConnection | null;
@@ -40,12 +42,10 @@ export declare class WebRTCSignaling {
     stopScreenCapture(): Promise<void>;
     stopCapture(): Promise<void>;
     private applyEncodingParams;
-    private startStatsLogging;
-    private stopStatsLogging;
-    private logSenderStats;
     private sendOffer;
     private handleAnswer;
     private handleLocalIceCandidate;
     private handleRemoteIceCandidate;
     private handleReady;
+    private handleTrackMap;
 }