@vanira/sdk 0.0.41 → 0.0.43

package/dist/runtime/reactNativeRuntime.d.ts

@@ -0,0 +1,18 @@
+import { VaniraRuntime, RuntimeCapabilities } from './types';
+import { WebRTCClientConfig } from '../types';
+import { WebRTCClient } from '../core/WebRTCClient';
+import { VaniraAI } from '../core/VaniraAI';
+
+export declare const reactNativeCapabilities: RuntimeCapabilities;
+export declare const reactNativeRuntime: VaniraRuntime;
+/**
+ * Create a WebRTCClient pre-wired for React Native.
+ *
+ * The callId is auto-generated with an 'rn_' prefix for tracing.
+ * The /calls/create request body will include `{ type: 'web', runtime: 'react-native' }`.
+ */
+export declare function createReactNativeClient(config: Omit<WebRTCClientConfig, 'audioAdapter' | 'mediaAdapter' | 'peerAdapter' | 'dataChannelAdapter'>): WebRTCClient;
+/**
+ * Create a VaniraAI instance pre-wired for React Native.
+ */
+export declare function createReactNativeAI(config: Omit<WebRTCClientConfig, 'audioAdapter' | 'mediaAdapter' | 'peerAdapter' | 'dataChannelAdapter'>): VaniraAI;

package/dist/runtime/types.d.ts

@@ -0,0 +1,96 @@
+import { AudioAdapter, MediaAdapter } from '../adapters/interfaces';
+import { PeerConnectionAdapter, DataChannelAdapter } from '../adapters/PeerConnectionAdapter';
+
+/**
+ * Declarative flags describing what the current platform supports.
+ * Use these instead of hardcoded platform checks throughout the codebase.
+ *
+ * The backend also receives the runtime name and uses it to skip
+ * `playedStream` waiting for platforms that don't fire `onended`.
+ */
+export interface RuntimeCapabilities {
+    /**
+     * Whether HTMLAudioElement.onended fires after a TTS stream finishes.
+     * Browser: true — fires naturally after the stream ends.
+     * RN: false — react-native-webrtc routes audio natively, no onended event.
+     * Flutter: false — flutter_webrtc routes audio natively.
+     */
+    supportsAudioEndedEvent: boolean;
+    /**
+     * Whether getDisplayMedia (screen capture) is available.
+     * Browser: true (desktop), false on mobile browsers.
+     * RN: false — no screen capture API.
+     * Flutter: false — flutter_webrtc does not expose getDisplayMedia.
+     */
+    supportsScreenShare: boolean;
+    /**
+     * Whether document, window, and DOM APIs are present.
+     * Browser: true.  RN: false.  Flutter: false.
+     */
+    supportsDom: boolean;
+    /**
+     * Whether new Audio() / HTMLAudioElement is constructable.
+     * Browser: true.  RN: false.  Flutter: false.
+     */
+    supportsHtmlAudio: boolean;
+    /**
+     * Whether customElements.define() is available for widget registration.
+     * Browser: true.  RN: false.  Flutter: false.
+     */
+    supportsCustomElements: boolean;
+    /**
+     * Whether localStorage / sessionStorage are available.
+     * Browser: true.  RN: false (AsyncStorage used instead).  Flutter: false.
+     */
+    supportsLocalStorage: boolean;
+    /**
+     * Whether BroadcastChannel (cross-tab communication) is available.
+     * Browser: true.  RN: false.  Flutter: false.
+     */
+    supportsBroadcastChannel: boolean;
+}
+/**
+ * A VaniraRuntime encapsulates everything the WebRTCClient needs to know
+ * about the current platform:
+ *
+ *   - Which adapters to use for media, audio, peer connection, data channel
+ *   - What metadata to send to the backend (type, runtime)
+ *   - How to generate call IDs
+ *   - What features are available (capabilities)
+ *
+ * The WebRTCClient accepts a runtime via config.runtime and extracts all
+ * adapter + metadata from it.  Individual adapter fields on config are still
+ * supported for backward compatibility.
+ */
+export interface VaniraRuntime {
+    /** Human-readable runtime identifier */
+    readonly name: 'browser' | 'react-native' | 'flutter' | string;
+    /**
+     * Sent as `type` in the /calls/create request body.
+     * Must be `web` for all WebRTC flows — the API only accepts `web` | `sip`.
+     * Use `runtimeName` for browser vs react-native vs flutter.
+     */
+    readonly callType: 'web';
+    /**
+     * Sent as `runtime` in the /calls/create request body.
+     * Replaces the old callId prefix convention.
+     * Values: 'browser' | 'react-native' | 'flutter'
+     */
+    readonly runtimeName: string;
+    /**
+     * Prefix used when auto-generating a callId (no prefix conflicts with
+     * runtime field, but kept for logging / tracing convenience).
+     * 'web_' | 'rn_' | 'flutter_'
+     */
+    readonly callIdPrefix: string;
+    /** Runtime capability flags */
+    readonly capabilities: RuntimeCapabilities;
+    /** Audio output adapter (remote TTS playback) */
+    readonly audio: AudioAdapter;
+    /** Media input adapter (microphone access) */
+    readonly media: MediaAdapter;
+    /** RTCPeerConnection factory adapter */
+    readonly peer: PeerConnectionAdapter;
+    /** DataChannel event binding + message normalisation adapter */
+    readonly dataChannel: DataChannelAdapter;
+}

package/dist/types.d.ts

@@ -46,19 +46,51 @@ export interface WebRTCClientConfig {
     }) => void;
     onConnected?: () => void;
     onDisconnected?: () => void;
-    onError?: (error: any) => void;
+    onError?: (error: unknown) => void;
     onTranscription?: (text: string, isFinal: boolean) => void;
     onLocalStream?: (stream: MediaStream) => void;
     onRemoteTrack?: (track: MediaStreamTrack, stream: MediaStream) => void;
-    onClientToolCall?: (toolCall: any) => void;
+    onClientToolCall?: (toolCall: unknown) => void;
+    /**
+     * STUN/TURN servers to use for the RTCPeerConnection.
+     * If omitted, the SDK falls back to its own built-in STUN + metered.ca TURN defaults.
+     * Pass your own fetched servers (e.g. from WebRTCClient.fetchIceServers()) for
+     * tighter control over relay selection.
+     */
     iceServers?: RTCIceServer[];
     token?: string;
+    runtime?: import('./runtime/types').VaniraRuntime;
+    /**
+     * Audio output adapter. Controls how incoming remote audio is played.
+     * Default (browser): HTMLAudioElement-backed player.
+     * React Native: no-op stub (react-native-webrtc routes audio natively).
+     */
+    audioAdapter?: import('./adapters/interfaces').AudioAdapter;
+    /**
+     * Media input adapter. Controls how the microphone is accessed.
+     * Default (browser): navigator.mediaDevices.getUserMedia.
+     * React Native: react-native-webrtc mediaDevices.getUserMedia polyfill.
+     */
+    mediaAdapter?: import('./adapters/interfaces').MediaAdapter;
+    /**
+     * RTCPeerConnection factory adapter.
+     * Default (browser): BrowserPeerAdapter (wraps new RTCPeerConnection()).
+     * React Native: RNPeerAdapter (wraps react-native-webrtc global).
+     */
+    peerAdapter?: import('./adapters/PeerConnectionAdapter').PeerConnectionAdapter;
+    /**
+     * DataChannel event binding + message normalisation adapter.
+     * Default (browser): BrowserDataChannelAdapter (handles string/ArrayBuffer/Blob).
+     * React Native: RNDataChannelAdapter (string-only, no binary normalisation).
+     */
+    dataChannelAdapter?: import('./adapters/PeerConnectionAdapter').DataChannelAdapter;
 }
 export interface ControlEvent {
     event: string;
     text?: string;
     isFinal?: boolean;
     name?: string;
-    tool_call?: any;
-    data?: any;
+    tool_call?: unknown;
+    data?: unknown;
+    [key: string]: unknown;
 }

package/dist/ui/presets/WidgetPresetRenderer.d.ts

@@ -27,5 +27,7 @@ export declare class WidgetPresetRenderer {
     private renderCamera;
     private renderUpload;
     private renderNavigate;
+    private renderHighlight;
+    private renderTypeText;
 }
 export {};