@elevenlabs/client 0.15.2 → 1.0.0-rc.0

package/dist/utils/WebRTCConnection.js

@@ -0,0 +1,494 @@
+import { BaseConnection, parseFormat, } from "./BaseConnection";
+import { sourceInfo } from "../sourceInfo";
+import { isValidSocketEvent } from "./events";
+import { Room, RoomEvent, Track, ConnectionState, createLocalAudioTrack, } from "livekit-client";
+import { constructOverrides, CONVERSATION_INITIATION_CLIENT_DATA_TYPE, } from "./overrides";
+import { arrayBufferToBase64 } from "./audio";
+import { loadRawAudioProcessor } from "./rawAudioProcessor.generated";
+const DEFAULT_LIVEKIT_WS_URL = "wss://livekit.rtc.elevenlabs.io";
+const HTTPS_API_ORIGIN = "https://api.elevenlabs.io";
+// Convert WSS origin to HTTPS for API calls
+function convertWssToHttps(origin) {
+    return origin.replace(/^wss:\/\//, "https://");
+}
+export class WebRTCConnection extends BaseConnection {
+    conversationId;
+    inputFormat;
+    outputFormat;
+    room;
+    isConnected = false;
+    audioEventId = 1;
+    audioCaptureContext = null;
+    audioElements = [];
+    outputDeviceId = null;
+    outputAnalyser = null;
+    outputFrequencyData = null;
+    // InputController state
+    _isMuted = false;
+    // InputController interface exposed as a property
+    input = {
+        close: async () => {
+            // Close only microphone tracks, not the entire connection
+            if (this.isConnected) {
+                try {
+                    this.room.localParticipant.audioTrackPublications.forEach(publication => {
+                        if (publication.track) {
+                            publication.track.stop();
+                        }
+                    });
+                }
+                catch (error) {
+                    console.warn("Error stopping local tracks:", error);
+                }
+            }
+        },
+        setDevice: async (config) => {
+            // WebRTC only supports changing inputDeviceId
+            // sampleRate, format, and preferHeadphonesForIosDevices are not supported
+            if (config?.sampleRate !== undefined ||
+                config?.format !== undefined ||
+                config?.preferHeadphonesForIosDevices !== undefined) {
+                throw new Error("WebRTC input device does not support sampleRate, format, or preferHeadphonesForIosDevices options");
+            }
+            const inputDeviceId = config?.inputDeviceId;
+            if (!inputDeviceId) {
+                // No device ID specified - this is a no-op for WebRTC
+                // The default device is already being used
+                return;
+            }
+            await this.setAudioInputDevice(inputDeviceId);
+        },
+        setMuted: async (isMuted) => {
+            if (!this.isConnected || !this.room.localParticipant) {
+                console.warn("Cannot set microphone muted: room not connected or no local participant");
+                return;
+            }
+            // Get the microphone track publication
+            const micTrackPublication = this.room.localParticipant.getTrackPublication(Track.Source.Microphone);
+            if (micTrackPublication?.track) {
+                try {
+                    // Use LiveKit's built-in track muting
+                    if (isMuted) {
+                        await micTrackPublication.track.mute();
+                    }
+                    else {
+                        await micTrackPublication.track.unmute();
+                    }
+                }
+                catch (_error) {
+                    // If track muting fails, fall back to participant-level control
+                    await this.room.localParticipant.setMicrophoneEnabled(!isMuted);
+                }
+            }
+            else {
+                // No track found, use participant-level control directly
+                await this.room.localParticipant.setMicrophoneEnabled(!isMuted);
+            }
+            this._isMuted = isMuted;
+        },
+        isMuted: () => this._isMuted,
+        getAnalyser: () => undefined, // WebRTC doesn't provide input analyser
+    };
+    // OutputController interface exposed as a property
+    output = {
+        close: async () => {
+            // No-op for WebRTC - LiveKit handles cleanup
+            // Audio elements are cleaned up when the connection closes
+        },
+        setDevice: async (config) => {
+            // WebRTC only supports changing outputDeviceId
+            // sampleRate and format are not supported
+            if (config?.sampleRate !== undefined || config?.format !== undefined) {
+                throw new Error("WebRTC output device does not support sampleRate or format options");
+            }
+            const outputDeviceId = config?.outputDeviceId;
+            if (!outputDeviceId) {
+                // No device ID specified - this is a no-op for WebRTC
+                // The default device is already being used
+                return;
+            }
+            await this.setAudioOutputDevice(outputDeviceId);
+        },
+        setVolume: (volume) => {
+            this.setAudioVolume(volume);
+        },
+        interrupt: (_resetDuration) => {
+            // No-op for WebRTC - LiveKit handles audio playback and interruption
+            // Audio interruption is managed by the server/agent
+        },
+        getAnalyser: () => this.outputAnalyser ?? undefined,
+    };
+    constructor(room, conversationId, inputFormat, outputFormat, config = {}) {
+        super(config);
+        this.room = room;
+        this.conversationId = conversationId;
+        this.inputFormat = inputFormat;
+        this.outputFormat = outputFormat;
+        this.setupRoomEventListeners();
+    }
+    static async create(config) {
+        let conversationToken;
+        // Handle different authentication scenarios
+        if ("conversationToken" in config && config.conversationToken) {
+            // Direct token provided
+            conversationToken = config.conversationToken;
+        }
+        else if ("agentId" in config && config.agentId) {
+            // Agent ID provided - fetch token from API
+            try {
+                const { name: source, version } = sourceInfo;
+                const configOrigin = config.origin ?? HTTPS_API_ORIGIN;
+                const origin = convertWssToHttps(configOrigin); //origin is wss, not https
+                const url = `${origin}/v1/convai/conversation/token?agent_id=${config.agentId}&source=${source}&version=${version}`;
+                const response = await fetch(url);
+                if (!response.ok) {
+                    throw new Error(`ElevenLabs API returned ${response.status} ${response.statusText}`);
+                }
+                const data = await response.json();
+                conversationToken = data.token;
+                if (!conversationToken) {
+                    throw new Error("No conversation token received from API");
+                }
+            }
+            catch (error) {
+                let msg = error instanceof Error ? error.message : String(error);
+                if (error instanceof Error && error.message.includes("401")) {
+                    msg =
+                        "Your agent has authentication enabled, but no signed URL or conversation token was provided.";
+                }
+                throw new Error(`Failed to fetch conversation token for agent ${config.agentId}: ${msg}`);
+            }
+        }
+        else {
+            throw new Error("Either conversationToken or agentId is required for WebRTC connection");
+        }
+        const room = new Room();
+        try {
+            // Create connection instance first to set up event listeners
+            const conversationId = `room_${Date.now()}`;
+            const inputFormat = parseFormat("pcm_48000");
+            const outputFormat = parseFormat("pcm_48000");
+            const connection = new WebRTCConnection(room, conversationId, inputFormat, outputFormat, config);
+            // Use configurable LiveKit URL or default if not provided
+            const livekitUrl = config.livekitUrl || DEFAULT_LIVEKIT_WS_URL;
+            // Enable microphone on SignalConnected (before room.connect resolves).
+            // The server may wait for the client to publish audio before fully
+            // establishing the subscriber peer connection, matching the behaviour
+            // of @livekit/components-react's useLiveKitRoom hook.
+            const micEnabled = config.textOnly
+                ? Promise.resolve()
+                : new Promise((resolve, reject) => {
+                    room.once(RoomEvent.SignalConnected, () => {
+                        room.localParticipant
+                            .setMicrophoneEnabled(true)
+                            .then(() => resolve())
+                            .catch(reject);
+                    });
+                });
+            // Connect to the LiveKit room
+            await room.connect(livekitUrl, conversationToken);
+            // Wait for the Connected event to ensure isConnected is true
+            await new Promise(resolve => {
+                if (connection.isConnected) {
+                    resolve();
+                }
+                else {
+                    const onConnected = () => {
+                        room.off(RoomEvent.Connected, onConnected);
+                        resolve();
+                    };
+                    room.on(RoomEvent.Connected, onConnected);
+                }
+            });
+            // Ensure the microphone was successfully enabled
+            await micEnabled;
+            if (room.name) {
+                connection.conversationId =
+                    room.name.match(/(conv_[a-zA-Z0-9]+)/)?.[0] || room.name;
+            }
+            const overridesEvent = constructOverrides(config);
+            connection.debug({
+                type: CONVERSATION_INITIATION_CLIENT_DATA_TYPE,
+                message: overridesEvent,
+            });
+            await connection.sendMessage(overridesEvent);
+            return connection;
+        }
+        catch (error) {
+            await room.disconnect();
+            throw error;
+        }
+    }
+    setupRoomEventListeners() {
+        this.room.on(RoomEvent.Connected, async () => {
+            this.isConnected = true;
+        });
+        this.room.on(RoomEvent.Disconnected, reason => {
+            this.isConnected = false;
+            this.disconnect({
+                reason: "agent",
+                context: new CloseEvent("close", { reason: reason?.toString() }),
+            });
+        });
+        this.room.on(RoomEvent.ConnectionStateChanged, state => {
+            if (state === ConnectionState.Disconnected) {
+                this.isConnected = false;
+                this.disconnect({
+                    reason: "error",
+                    message: `LiveKit connection state changed to ${state}`,
+                    context: new Event("connection_state_changed"),
+                });
+            }
+        });
+        // Handle incoming data messages
+        this.room.on(RoomEvent.DataReceived, (payload, _participant) => {
+            try {
+                const message = JSON.parse(new TextDecoder().decode(payload));
+                // Filter out audio messages for WebRTC - they're handled via audio tracks
+                if (message.type === "audio") {
+                    return;
+                }
+                if (isValidSocketEvent(message)) {
+                    this.handleMessage(message);
+                }
+                else {
+                    console.warn("Invalid socket event received:", message);
+                }
+            }
+            catch (error) {
+                console.warn("Failed to parse incoming data message:", error);
+                console.warn("Raw payload:", new TextDecoder().decode(payload));
+            }
+        });
+        this.room.on(RoomEvent.TrackSubscribed, async (track, _publication, participant) => {
+            if (track.kind === Track.Kind.Audio &&
+                participant.identity.includes("agent")) {
+                // Play the audio track
+                const remoteAudioTrack = track;
+                const audioElement = remoteAudioTrack.attach();
+                audioElement.autoplay = true;
+                audioElement.controls = false;
+                // Set output device if one was previously selected
+                if (this.outputDeviceId && audioElement.setSinkId) {
+                    try {
+                        await audioElement.setSinkId(this.outputDeviceId);
+                    }
+                    catch (error) {
+                        console.warn("Failed to set output device for new audio element:", error);
+                    }
+                }
+                // Add to DOM (hidden) to ensure it plays
+                audioElement.style.display = "none";
+                document.body.appendChild(audioElement);
+                // Store reference for volume control
+                this.audioElements.push(audioElement);
+                // Apply current volume if it exists (for when volume was set before audio track arrived)
+                if (this.audioElements.length === 1) {
+                    // First audio element - trigger a callback to sync with current volume
+                    this.onDebug?.({ type: "audio_element_ready" });
+                }
+                // Set up audio capture for onAudio callback
+                await this.setupAudioCapture(remoteAudioTrack);
+            }
+        });
+        this.room.on(RoomEvent.ActiveSpeakersChanged, async (speakers) => {
+            if (speakers.length > 0) {
+                this.updateMode(speakers[0].identity.startsWith("agent") ? "speaking" : "listening");
+            }
+            else {
+                this.updateMode("listening");
+            }
+        });
+        this.room.on(RoomEvent.ParticipantDisconnected, (participant) => {
+            if (participant.identity?.startsWith("agent")) {
+                this.disconnect({
+                    reason: "agent",
+                    context: new CloseEvent("close", { reason: "agent disconnected" }),
+                });
+            }
+        });
+    }
+    close() {
+        if (this.isConnected) {
+            try {
+                // Explicitly stop all local tracks before disconnecting to ensure microphone is released
+                this.room.localParticipant.audioTrackPublications.forEach(publication => {
+                    if (publication.track) {
+                        publication.track.stop();
+                    }
+                });
+            }
+            catch (error) {
+                console.warn("Error stopping local tracks:", error);
+            }
+            // Clean up audio capture context (non-blocking)
+            if (this.audioCaptureContext) {
+                this.audioCaptureContext.close().catch(error => {
+                    console.warn("Error closing audio capture context:", error);
+                });
+                this.audioCaptureContext = null;
+            }
+            // Clean up audio elements
+            this.audioElements.forEach(element => {
+                if (element.parentNode) {
+                    element.parentNode.removeChild(element);
+                }
+            });
+            this.audioElements = [];
+            this.room.disconnect();
+        }
+    }
+    async sendMessage(message) {
+        if (!this.isConnected || !this.room.localParticipant) {
+            console.warn("Cannot send message: room not connected or no local participant");
+            return;
+        }
+        // In WebRTC mode, audio is sent via published tracks, not data messages
+        if ("user_audio_chunk" in message) {
+            // Ignore audio data messages - audio flows through WebRTC tracks
+            return;
+        }
+        try {
+            const encoder = new TextEncoder();
+            const data = encoder.encode(JSON.stringify(message));
+            await this.room.localParticipant.publishData(data, { reliable: true });
+        }
+        catch (error) {
+            this.debug({
+                type: "send_message_error",
+                message: {
+                    message,
+                    error,
+                },
+            });
+            console.error("Failed to send message via WebRTC:", error);
+        }
+    }
+    // Get the room instance for advanced usage
+    getRoom() {
+        return this.room;
+    }
+    async setupAudioCapture(track) {
+        try {
+            // Create audio context for processing
+            const audioContext = new AudioContext();
+            this.audioCaptureContext = audioContext;
+            // Create analyser for frequency data
+            this.outputAnalyser = audioContext.createAnalyser();
+            this.outputAnalyser.fftSize = 2048;
+            this.outputAnalyser.smoothingTimeConstant = 0.8;
+            // Create MediaStream from the track
+            const mediaStream = new MediaStream([track.mediaStreamTrack]);
+            // Create audio source from the stream
+            const source = audioContext.createMediaStreamSource(mediaStream);
+            // Connect source to analyser
+            source.connect(this.outputAnalyser);
+            await loadRawAudioProcessor(audioContext.audioWorklet);
+            const worklet = new AudioWorkletNode(audioContext, "rawAudioProcessor");
+            // Connect analyser to worklet for processing
+            this.outputAnalyser.connect(worklet);
+            // Configure the processor for the output format
+            worklet.port.postMessage({
+                type: "setFormat",
+                format: this.outputFormat.format,
+                sampleRate: this.outputFormat.sampleRate,
+            });
+            // Handle processed audio data
+            worklet.port.onmessage = (event) => {
+                const [audioData, maxVolume] = event.data;
+                // Only send audio if there's significant volume (not just silence)
+                const volumeThreshold = 0.01;
+                if (maxVolume > volumeThreshold) {
+                    // Convert to base64
+                    const base64Audio = arrayBufferToBase64(audioData.buffer);
+                    // Use sequential event ID for proper feedback tracking
+                    const eventId = this.audioEventId++;
+                    // Trigger the onAudio callback by simulating an audio event
+                    this.handleMessage({
+                        type: "audio",
+                        audio_event: {
+                            audio_base_64: base64Audio,
+                            event_id: eventId,
+                        },
+                    });
+                }
+            };
+            // Connect the audio processing chain
+            source.connect(worklet);
+        }
+        catch (error) {
+            console.warn("Failed to set up audio capture:", error);
+        }
+    }
+    setAudioVolume(volume) {
+        this.audioElements.forEach(element => {
+            element.volume = volume;
+        });
+    }
+    async setAudioOutputDevice(deviceId) {
+        if (!("setSinkId" in HTMLAudioElement.prototype)) {
+            throw new Error("setSinkId is not supported in this browser");
+        }
+        // Set output device for all existing audio elements
+        const promises = this.audioElements.map(async (element) => {
+            try {
+                await element.setSinkId(deviceId);
+            }
+            catch (error) {
+                console.error("Failed to set sink ID for audio element:", error);
+                throw error;
+            }
+        });
+        await Promise.all(promises);
+        // Store the device ID for future audio elements
+        this.outputDeviceId = deviceId;
+    }
+    async setAudioInputDevice(deviceId) {
+        if (!this.isConnected || !this.room.localParticipant) {
+            throw new Error("Cannot change input device: room not connected or no local participant");
+        }
+        try {
+            // Get the current microphone track publication
+            const currentMicTrackPublication = this.room.localParticipant.getTrackPublication(Track.Source.Microphone);
+            // Stop the current microphone track if it exists
+            if (currentMicTrackPublication?.track) {
+                await currentMicTrackPublication.track.stop();
+                await this.room.localParticipant.unpublishTrack(currentMicTrackPublication.track);
+            }
+            // Create constraints for the new input device
+            const audioConstraints = {
+                deviceId: { exact: deviceId },
+                echoCancellation: true,
+                noiseSuppression: true,
+                autoGainControl: true,
+                channelCount: { ideal: 1 },
+            };
+            // Create new audio track with the specified device
+            const audioTrack = await createLocalAudioTrack(audioConstraints);
+            // Publish the new microphone track
+            await this.room.localParticipant.publishTrack(audioTrack, {
+                name: "microphone",
+                source: Track.Source.Microphone,
+            });
+        }
+        catch (error) {
+            console.error("Failed to change input device:", error);
+            // Try to re-enable default microphone on failure
+            try {
+                await this.room.localParticipant.setMicrophoneEnabled(true);
+            }
+            catch (recoveryError) {
+                console.error("Failed to recover microphone after device switch error:", recoveryError);
+            }
+            throw error;
+        }
+    }
+    getOutputByteFrequencyData() {
+        if (!this.outputAnalyser)
+            return null;
+        this.outputFrequencyData ??= new Uint8Array(this.outputAnalyser.frequencyBinCount);
+        this.outputAnalyser.getByteFrequencyData(this.outputFrequencyData);
+        return this.outputFrequencyData;
+    }
+}
+//# sourceMappingURL=WebRTCConnection.js.map

package/dist/utils/WebRTCConnection.js.map

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package/dist/utils/WebSocketConnection.d.ts

@@ -1,13 +1,18 @@
 import { BaseConnection, type SessionConfig, type FormatConfig } from "./BaseConnection";
-import { type OutgoingSocketEvent } from "./events";
-export declare class WebSocketConnection extends BaseConnection {
+import { type OutgoingSocketEvent, type IncomingSocketEvent } from "./events";
+import type { OutputEventTarget, OutputListener } from "./output";
+export declare class WebSocketConnection extends BaseConnection implements OutputEventTarget {
     private readonly socket;
     readonly conversationId: string;
     readonly inputFormat: FormatConfig;
     readonly outputFormat: FormatConfig;
+    private outputListeners;
     private constructor();
     static create(config: SessionConfig): Promise<WebSocketConnection>;
     close(): void;
     sendMessage(message: OutgoingSocketEvent): void;
-    setMicMuted(isMuted: boolean): Promise<void>;
+    addListener(listener: OutputListener): void;
+    removeListener(listener: OutputListener): void;
+    protected handleMessage(parsedEvent: IncomingSocketEvent): void;
 }
+//# sourceMappingURL=WebSocketConnection.d.ts.map

package/dist/utils/WebSocketConnection.d.ts.map

@@ -0,0 +1 @@
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