@dotdo/do 0.1.0

package/dist/ai/voice/tts.d.ts

@@ -0,0 +1,173 @@
+/**
+ * Voice - Text-to-Speech and Speech-to-Text
+ *
+ * Voice synthesis (TTS) and speech recognition (STT) capabilities.
+ *
+ * Features:
+ * - Text-to-speech with multiple voices
+ * - Speech-to-text transcription
+ * - Real-time streaming transcription
+ * - Word-level timestamps
+ * - Speaker diarization
+ *
+ * @module ai/voice/tts
+ */
+import type { VoiceSynthesisOptions, VoiceSynthesisResult, SpeechRecognitionOptions, TranscriptionResult, VoiceSynthesisProvider } from '../../types/ai';
+/**
+ * Synthesize speech from text
+ *
+ * @param text - Text to synthesize
+ * @param options - Synthesis options including voice
+ * @returns Audio buffer and metadata
+ *
+ * @example
+ * ```typescript
+ * // Using ElevenLabs
+ * const result = await synthesize('Hello, how are you?', {
+ *   provider: 'elevenlabs',
+ *   voice: 'rachel',
+ *   format: 'mp3'
+ * })
+ *
+ * // Using OpenAI
+ * const result = await synthesize('Welcome to our app', {
+ *   provider: 'openai',
+ *   voice: 'alloy',
+ *   speed: 1.0
+ * })
+ *
+ * // Play the audio
+ * const blob = new Blob([result.audio], { type: `audio/${result.format}` })
+ * const url = URL.createObjectURL(blob)
+ * ```
+ */
+export declare function synthesize(text: string, options: VoiceSynthesisOptions): Promise<VoiceSynthesisResult>;
+/**
+ * Synthesize speech with streaming output
+ *
+ * @param text - Text to synthesize
+ * @param options - Synthesis options
+ * @returns Async iterable of audio chunks
+ *
+ * @example
+ * ```typescript
+ * for await (const chunk of synthesizeStream('Long text...', {
+ *   provider: 'elevenlabs',
+ *   voice: 'rachel'
+ * })) {
+ *   // Process audio chunk
+ *   audioContext.decodeAudioData(chunk)
+ * }
+ * ```
+ */
+export declare function synthesizeStream(text: string, options: VoiceSynthesisOptions): AsyncIterable<ArrayBuffer>;
+/**
+ * Transcribe audio to text
+ *
+ * @param audio - Audio buffer to transcribe
+ * @param options - Transcription options
+ * @returns Transcription result with text and metadata
+ *
+ * @example
+ * ```typescript
+ * // Basic transcription
+ * const result = await transcribe(audioBuffer, {
+ *   provider: 'deepgram',
+ *   language: 'en'
+ * })
+ * console.log(result.text)
+ *
+ * // With word timestamps
+ * const result = await transcribe(audioBuffer, {
+ *   provider: 'deepgram',
+ *   punctuate: true
+ * })
+ * result.words?.forEach(w => {
+ *   console.log(`${w.word} (${w.start}s - ${w.end}s)`)
+ * })
+ *
+ * // With speaker diarization
+ * const result = await transcribe(meetingAudio, {
+ *   provider: 'deepgram',
+ *   diarize: true
+ * })
+ * result.speakers?.forEach(s => {
+ *   console.log(`Speaker ${s.speaker}: ${s.text}`)
+ * })
+ * ```
+ */
+export declare function transcribe(audio: ArrayBuffer, options?: SpeechRecognitionOptions): Promise<TranscriptionResult>;
+/**
+ * Transcribe audio stream in real-time
+ *
+ * @param audioStream - Readable stream of audio data
+ * @param options - Transcription options
+ * @returns Async iterable of partial transcription results
+ *
+ * @example
+ * ```typescript
+ * // From microphone
+ * const mediaStream = await navigator.mediaDevices.getUserMedia({ audio: true })
+ * const audioStream = getAudioStream(mediaStream)
+ *
+ * for await (const partial of transcribeStream(audioStream, {
+ *   provider: 'deepgram',
+ *   interimResults: true
+ * })) {
+ *   console.log('Partial:', partial.text)
+ * }
+ * ```
+ */
+export declare function transcribeStream(audioStream: ReadableStream, options?: SpeechRecognitionOptions): AsyncIterable<TranscriptionResult>;
+/**
+ * Transcribe audio from URL
+ *
+ * @param url - URL of audio file
+ * @param options - Transcription options
+ * @returns Transcription result
+ *
+ * @example
+ * ```typescript
+ * const result = await transcribeUrl(
+ *   'https://example.com/podcast.mp3',
+ *   { provider: 'assemblyai', diarize: true }
+ * )
+ * ```
+ */
+export declare function transcribeUrl(url: string, options?: SpeechRecognitionOptions): Promise<TranscriptionResult>;
+/**
+ * List available voices for a provider
+ *
+ * @param provider - Voice synthesis provider
+ * @returns Array of available voices
+ *
+ * @example
+ * ```typescript
+ * const voices = await listVoices('elevenlabs')
+ * voices.forEach(v => console.log(`${v.name}: ${v.description}`))
+ * ```
+ */
+export declare function listVoices(provider: VoiceSynthesisProvider): Promise<Array<{
+    id: string;
+    name: string;
+    description?: string;
+}>>;
+/**
+ * Clone a voice from audio sample (ElevenLabs)
+ *
+ * @param name - Name for the cloned voice
+ * @param audioSamples - Audio samples of the voice
+ * @param description - Optional description
+ * @returns Voice ID of the cloned voice
+ *
+ * @example
+ * ```typescript
+ * const voiceId = await cloneVoice(
+ *   'My Custom Voice',
+ *   [sample1, sample2, sample3],
+ *   'A warm, friendly voice'
+ * )
+ * ```
+ */
+export declare function cloneVoice(name: string, audioSamples: ArrayBuffer[], description?: string): Promise<string>;
+//# sourceMappingURL=tts.d.ts.map

package/dist/ai/voice/tts.d.ts.map

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package/dist/ai/voice/tts.js

@@ -0,0 +1,252 @@
+/**
+ * Voice - Text-to-Speech and Speech-to-Text
+ *
+ * Voice synthesis (TTS) and speech recognition (STT) capabilities.
+ *
+ * Features:
+ * - Text-to-speech with multiple voices
+ * - Speech-to-text transcription
+ * - Real-time streaming transcription
+ * - Word-level timestamps
+ * - Speaker diarization
+ *
+ * @module ai/voice/tts
+ */
+/**
+ * Default TTS models by provider
+ */
+const DEFAULT_TTS_MODELS = {
+    elevenlabs: 'eleven_multilingual_v2',
+    playht: 'PlayHT2.0',
+    azure: 'en-US-JennyNeural',
+    google: 'en-US-Standard-A',
+    openai: 'tts-1',
+    deepgram: 'aura-asteria-en',
+    cartesia: 'sonic-english',
+};
+/**
+ * Default STT models by provider
+ */
+const DEFAULT_STT_MODELS = {
+    deepgram: 'nova-2',
+    assemblyai: 'default',
+    whisper: 'whisper-1',
+    google: 'default',
+    azure: 'default',
+};
+/**
+ * Synthesize speech from text
+ *
+ * @param text - Text to synthesize
+ * @param options - Synthesis options including voice
+ * @returns Audio buffer and metadata
+ *
+ * @example
+ * ```typescript
+ * // Using ElevenLabs
+ * const result = await synthesize('Hello, how are you?', {
+ *   provider: 'elevenlabs',
+ *   voice: 'rachel',
+ *   format: 'mp3'
+ * })
+ *
+ * // Using OpenAI
+ * const result = await synthesize('Welcome to our app', {
+ *   provider: 'openai',
+ *   voice: 'alloy',
+ *   speed: 1.0
+ * })
+ *
+ * // Play the audio
+ * const blob = new Blob([result.audio], { type: `audio/${result.format}` })
+ * const url = URL.createObjectURL(blob)
+ * ```
+ */
+export async function synthesize(text, options) {
+    // TODO: Implement text-to-speech
+    // 1. Select provider and model
+    // 2. Build provider-specific request
+    // 3. Make gateway request
+    // 4. Return audio buffer
+    throw new Error('Not implemented');
+}
+/**
+ * Synthesize speech with streaming output
+ *
+ * @param text - Text to synthesize
+ * @param options - Synthesis options
+ * @returns Async iterable of audio chunks
+ *
+ * @example
+ * ```typescript
+ * for await (const chunk of synthesizeStream('Long text...', {
+ *   provider: 'elevenlabs',
+ *   voice: 'rachel'
+ * })) {
+ *   // Process audio chunk
+ *   audioContext.decodeAudioData(chunk)
+ * }
+ * ```
+ */
+export async function* synthesizeStream(text, options) {
+    // TODO: Implement streaming TTS
+    // Some providers support streaming audio output
+    throw new Error('Not implemented');
+}
+/**
+ * Transcribe audio to text
+ *
+ * @param audio - Audio buffer to transcribe
+ * @param options - Transcription options
+ * @returns Transcription result with text and metadata
+ *
+ * @example
+ * ```typescript
+ * // Basic transcription
+ * const result = await transcribe(audioBuffer, {
+ *   provider: 'deepgram',
+ *   language: 'en'
+ * })
+ * console.log(result.text)
+ *
+ * // With word timestamps
+ * const result = await transcribe(audioBuffer, {
+ *   provider: 'deepgram',
+ *   punctuate: true
+ * })
+ * result.words?.forEach(w => {
+ *   console.log(`${w.word} (${w.start}s - ${w.end}s)`)
+ * })
+ *
+ * // With speaker diarization
+ * const result = await transcribe(meetingAudio, {
+ *   provider: 'deepgram',
+ *   diarize: true
+ * })
+ * result.speakers?.forEach(s => {
+ *   console.log(`Speaker ${s.speaker}: ${s.text}`)
+ * })
+ * ```
+ */
+export async function transcribe(audio, options) {
+    // TODO: Implement speech-to-text
+    // 1. Select provider and model
+    // 2. Build provider-specific request
+    // 3. Make gateway request
+    // 4. Normalize response
+    throw new Error('Not implemented');
+}
+/**
+ * Transcribe audio stream in real-time
+ *
+ * @param audioStream - Readable stream of audio data
+ * @param options - Transcription options
+ * @returns Async iterable of partial transcription results
+ *
+ * @example
+ * ```typescript
+ * // From microphone
+ * const mediaStream = await navigator.mediaDevices.getUserMedia({ audio: true })
+ * const audioStream = getAudioStream(mediaStream)
+ *
+ * for await (const partial of transcribeStream(audioStream, {
+ *   provider: 'deepgram',
+ *   interimResults: true
+ * })) {
+ *   console.log('Partial:', partial.text)
+ * }
+ * ```
+ */
+export async function* transcribeStream(audioStream, options) {
+    // TODO: Implement real-time streaming transcription
+    // 1. Open WebSocket to provider
+    // 2. Stream audio chunks
+    // 3. Yield partial results
+    throw new Error('Not implemented');
+}
+/**
+ * Transcribe audio from URL
+ *
+ * @param url - URL of audio file
+ * @param options - Transcription options
+ * @returns Transcription result
+ *
+ * @example
+ * ```typescript
+ * const result = await transcribeUrl(
+ *   'https://example.com/podcast.mp3',
+ *   { provider: 'assemblyai', diarize: true }
+ * )
+ * ```
+ */
+export async function transcribeUrl(url, options) {
+    // TODO: Implement URL-based transcription
+    // Some providers accept URLs directly
+    throw new Error('Not implemented');
+}
+/**
+ * List available voices for a provider
+ *
+ * @param provider - Voice synthesis provider
+ * @returns Array of available voices
+ *
+ * @example
+ * ```typescript
+ * const voices = await listVoices('elevenlabs')
+ * voices.forEach(v => console.log(`${v.name}: ${v.description}`))
+ * ```
+ */
+export async function listVoices(provider) {
+    // TODO: Implement voice listing
+    // Provider-specific API calls
+    throw new Error('Not implemented');
+}
+/**
+ * Clone a voice from audio sample (ElevenLabs)
+ *
+ * @param name - Name for the cloned voice
+ * @param audioSamples - Audio samples of the voice
+ * @param description - Optional description
+ * @returns Voice ID of the cloned voice
+ *
+ * @example
+ * ```typescript
+ * const voiceId = await cloneVoice(
+ *   'My Custom Voice',
+ *   [sample1, sample2, sample3],
+ *   'A warm, friendly voice'
+ * )
+ * ```
+ */
+export async function cloneVoice(name, audioSamples, description) {
+    // TODO: Implement voice cloning (ElevenLabs specific)
+    throw new Error('Not implemented');
+}
+/**
+ * Format TTS request for provider
+ *
+ * @internal
+ */
+function formatTTSRequest(text, provider, options) {
+    // TODO: Handle provider-specific request formats
+    throw new Error('Not implemented');
+}
+/**
+ * Format STT request for provider
+ *
+ * @internal
+ */
+function formatSTTRequest(audio, provider, options) {
+    // TODO: Handle provider-specific request formats
+    throw new Error('Not implemented');
+}
+/**
+ * Parse STT response from provider
+ *
+ * @internal
+ */
+function parseSTTResponse(response, provider) {
+    // TODO: Normalize provider responses
+    throw new Error('Not implemented');
+}
+//# sourceMappingURL=tts.js.map

package/dist/ai/voice/tts.js.map

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package/dist/ai/voice/webrtc.d.ts

@@ -0,0 +1,228 @@
+/**
+ * WebRTC Session Handling
+ *
+ * Manages browser-based voice calls using WebRTC.
+ * Supports multiple providers (LiveKit, Daily, Twilio).
+ *
+ * @module ai/voice/webrtc
+ */
+import type { WebRTCConfig, RealtimeTranscriptionConfig, VoiceAIProvider } from '../../types/voice-ai';
+import { VoiceAgentManager } from './agents';
+import { VoiceSessionManager } from './sessions';
+/**
+ * Options for creating a WebRTC session
+ */
+export interface CreateWebRTCSessionOptions {
+    /** Voice agent ID */
+    agentId: string;
+    /** Participant identity */
+    identity: string;
+    /** Display name */
+    displayName?: string;
+    /** Audio-only mode (no video) */
+    audioOnly?: boolean;
+    /** Auto-connect when joining */
+    autoConnect?: boolean;
+    /** Session metadata */
+    metadata?: Record<string, unknown>;
+    /** Real-time transcription config */
+    transcription?: RealtimeTranscriptionConfig;
+}
+/**
+ * Result of creating a WebRTC session
+ */
+export interface WebRTCSessionResult {
+    /** Session ID (for tracking) */
+    sessionId: string;
+    /** Room name */
+    roomName: string;
+    /** Room URL (for client connection) */
+    roomUrl: string;
+    /** Access token (for authentication) */
+    token: string;
+    /** WebRTC config */
+    config: WebRTCConfig;
+}
+/**
+ * WebRTC room information
+ */
+export interface WebRTCRoom {
+    /** Room name */
+    name: string;
+    /** Room URL */
+    url: string;
+    /** Session ID */
+    sessionId: string;
+    /** Agent ID */
+    agentId: string;
+    /** Provider */
+    provider: VoiceAIProvider;
+    /** Current participants */
+    participants: WebRTCParticipant[];
+    /** Created timestamp */
+    createdAt: number;
+    /** Room status */
+    status: 'active' | 'closed';
+}
+/**
+ * WebRTC participant
+ */
+export interface WebRTCParticipant {
+    /** Participant identity */
+    identity: string;
+    /** Display name */
+    displayName?: string;
+    /** Is AI agent */
+    isAgent: boolean;
+    /** Joined timestamp */
+    joinedAt: number;
+    /** Connection state */
+    connectionState: 'connecting' | 'connected' | 'disconnected';
+}
+/**
+ * WebRTC event types
+ */
+export type WebRTCEventType = 'room_created' | 'participant_joined' | 'participant_left' | 'track_subscribed' | 'track_unsubscribed' | 'room_closed';
+/**
+ * WebRTC session manager
+ *
+ * Creates and manages WebRTC sessions for browser-based voice calls.
+ * Works with LiveKit, Daily, or other WebRTC providers.
+ *
+ * @example
+ * ```typescript
+ * const manager = new WebRTCSessionManager(storage, agentManager, sessionManager)
+ *
+ * // Create a session for a user
+ * const { sessionId, roomUrl, token } = await manager.create({
+ *   agentId: 'vagent_support',
+ *   identity: 'user_123',
+ *   displayName: 'John Doe',
+ *   audioOnly: true,
+ * })
+ *
+ * // Client uses token to connect
+ * // const room = await LivekitClient.connect(roomUrl, token)
+ *
+ * // Get room info
+ * const room = await manager.getRoom(sessionId)
+ * ```
+ */
+export declare class WebRTCSessionManager {
+    private storage;
+    private agentManager;
+    private sessionManager;
+    /**
+     * Create a WebRTC session manager
+     *
+     * @param storage - DO storage interface
+     * @param agentManager - Voice agent manager
+     * @param sessionManager - Voice session manager
+     */
+    constructor(storage: DurableObjectStorage, agentManager: VoiceAgentManager, sessionManager: VoiceSessionManager);
+    /**
+     * Create a new WebRTC session
+     *
+     * Creates a room and generates an access token for the participant.
+     *
+     * @param options - Session options
+     * @returns Session details including token and room URL
+     */
+    create(options: CreateWebRTCSessionOptions): Promise<WebRTCSessionResult>;
+    /**
+     * Get room information
+     *
+     * @param sessionId - Voice session ID
+     * @returns Room info or null if not found
+     */
+    getRoom(sessionId: string): Promise<WebRTCRoom | null>;
+    /**
+     * Get room by name
+     *
+     * @param roomName - Room name
+     * @returns Room info or null if not found
+     */
+    getRoomByName(roomName: string): Promise<WebRTCRoom | null>;
+    /**
+     * Generate a new token for an existing room
+     *
+     * Useful when a participant needs to reconnect.
+     *
+     * @param sessionId - Voice session ID
+     * @param identity - Participant identity
+     * @returns New access token
+     */
+    generateToken(sessionId: string, identity: string): Promise<string>;
+    /**
+     * Update participant connection state
+     *
+     * Called from webhook when participant joins/leaves.
+     *
+     * @param roomName - Room name
+     * @param identity - Participant identity
+     * @param state - Connection state
+     */
+    updateParticipantState(roomName: string, identity: string, state: WebRTCParticipant['connectionState']): Promise<void>;
+    /**
+     * Add a participant to a room
+     *
+     * @param roomName - Room name
+     * @param participant - Participant info
+     */
+    addParticipant(roomName: string, participant: WebRTCParticipant): Promise<void>;
+    /**
+     * Remove a participant from a room
+     *
+     * @param roomName - Room name
+     * @param identity - Participant identity
+     */
+    removeParticipant(roomName: string, identity: string): Promise<void>;
+    /**
+     * Close a room
+     *
+     * @param roomName - Room name
+     */
+    closeRoom(roomName: string): Promise<void>;
+    /**
+     * List active rooms
+     *
+     * @param agentId - Optional filter by agent
+     * @returns List of active rooms
+     */
+    listActiveRooms(agentId?: string): Promise<WebRTCRoom[]>;
+    /**
+     * Clean up old/stale rooms
+     *
+     * Should be called periodically to clean up rooms that weren't
+     * properly closed.
+     *
+     * @param maxAge - Maximum age in milliseconds (default: 24 hours)
+     */
+    cleanup(maxAge?: number): Promise<number>;
+}
+/**
+ * WebRTC session singleton access
+ */
+export declare const WebRTCSession: {
+    /**
+     * Initialize with dependencies
+     */
+    init(storage: DurableObjectStorage, agentManager: VoiceAgentManager, sessionManager: VoiceSessionManager): void;
+    /**
+     * Create a WebRTC session
+     */
+    create(options: CreateWebRTCSessionOptions): Promise<WebRTCSessionResult>;
+    /**
+     * Get room info
+     */
+    getRoom(sessionId: string): Promise<WebRTCRoom | null>;
+    /**
+     * Generate new token
+     */
+    generateToken(sessionId: string, identity: string): Promise<string>;
+    /**
+     * List active rooms
+     */
+    listActiveRooms(agentId?: string): Promise<WebRTCRoom[]>;
+};
+//# sourceMappingURL=webrtc.d.ts.map

package/dist/ai/voice/webrtc.d.ts.map

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