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Classes

Concrete class implementations — providers, orchestrators, and core SDK components.

Concrete class implementations — providers, orchestrators, and core SDK components.

  • AgentStateMachine — Orchestrator state machine that derives a high-level AgentState from three underlying sub-machines: capture, playback, and processing.
  • AnthropicLLM — Anthropic LLM provider for Claude models.
  • AssemblyAISTT — AssemblyAI real-time STT provider using a raw WebSocket connection.
  • AudioBufferQueue — Bounded FIFO queue that buffers audio chunks between pipeline stages.
  • AudioCapture — Manages microphone audio capture using the Web Audio API.
  • AudioHeaderCache — Caches the audio container header from a stream for re-injection on reconnect.
  • AudioPlayer — Manages audio playback using the Web Audio API with support for both complete and streaming playback modes.
  • AudioRouter — Manages audio buffer queues and header cache wiring for the pipeline.
  • AzureSTT — Microsoft Azure Speech real-time STT provider using a raw WebSocket.
  • AzureTTS — Microsoft Azure Speech TTS provider using native fetch.
  • BrowserAudioOutput — Browser audio output provider that plays audio through the Web Audio API.
  • BufferInput — Server-side audio input provider that accepts pushed audio buffers.
  • CartesiaTTS — Cartesia TTS provider for low-latency real-time streaming text-to-speech via WebSocket.
  • CompositeVoice— The primary class of the CompositeVoice SDK, orchestrating a complete 5-role audio pipeline from input capture through speech recognition, language model…
  • ConversationManager — Manages multi-turn conversation history for the LLM pipeline.
  • DeepgramAgent — Deepgram Agent API provider — covers STT + LLM + TTS in a single WebSocket.
  • DeepgramFlux — Deepgram Flux (V2) real-time STT provider using native WebSocket.
  • DeepgramSTT — Deepgram real-time STT provider using native WebSocket (no SDK required).
  • DeepgramTTS — Deepgram TTS provider for real-time streaming text-to-speech via native WebSocket.
  • DiscordVoice — Duplex Discord voice-channel provider (‘input’ + ‘output’).
  • EagerLLMController — Manages the eager/speculative LLM pipeline.
  • ElevenLabsAgent — ElevenLabs Conversational AI agent — covers STT + LLM + TTS in a single WebSocket.
  • ElevenLabsSTT — ElevenLabs STT provider for real-time streaming speech-to-text via WebSocket.
  • ElevenLabsTTS — ElevenLabs TTS provider for real-time streaming text-to-speech via WebSocket.
  • EventEmitter — A type-safe event emitter with support for wildcard listeners and both synchronous and asynchronous event dispatch.
  • FallbackSTT — A meta STT provider that chains multiple live STT providers with automatic failover.
  • FishAudioTTS — Fish Audio TTS provider using native fetch with msgpack-encoded requests.
  • GeminiLiveAgent — Gemini Live speech-to-speech agent — covers STT + LLM + TTS in a single WebSocket.
  • GeminiLLM — Google Gemini LLM provider.
  • GladiaSTT — Gladia real-time STT provider using the v2 live API.
  • GoogleMeetInput — Browser audio input provider that joins a Google Meet conference via the Meet Media API (Developer Preview) and emits the meeting’s mixed audio.
  • GoogleSTT — Google Cloud STT provider using native fetch for batch (per-utterance) speech-to-text.
  • GoogleTTS — Google Cloud TTS provider using native fetch for text-to-speech synthesis.
  • GroqLLM — Groq LLM provider for ultra-fast inference.
  • GuardrailPipeline — Runs a Guardrail chain over LLM text on its way to TTS.
  • GuardrailStream — Stateful guardrail filter for one streamed LLM response.
  • LMNTTTS — LMNT TTS provider using native fetch for text-to-speech synthesis.
  • Logger — Structured logger with context-aware formatting and configurable levels.
  • MicrophoneInput — Browser audio input provider that captures audio from the microphone.
  • MiniMaxTTS — MiniMax TTS provider using native fetch for text-to-speech synthesis.
  • MistralLLM — Mistral LLM provider.
  • MurfTTS — Murf AI TTS provider using native fetch for text-to-speech synthesis.
  • NativeSTT — Native browser STT provider backed by the Web Speech API (SpeechRecognition).
  • NativeTTS — Native browser TTS provider using the Web Speech API (SpeechSynthesis).
  • NullOutput — No-op audio output provider that discards all audio.
  • OpenAICompatibleLLM — Base LLM provider for any service that speaks the OpenAI chat completions format.
  • OpenAILLM — OpenAI LLM provider for GPT models.
  • OpenAIRealtimeAgent — OpenAI Realtime speech-to-speech agent — covers STT + LLM + TTS in a single WebSocket.
  • OpenAIRealtimeSTT — OpenAI real-time STT provider using the Realtime API transcription intent over a raw WebSocket connection.
  • OpenAITTS — OpenAI TTS provider using native fetch for text-to-speech synthesis.
  • PollyTTS — Amazon Polly TTS provider using native fetch with SigV4-signed requests.
  • ProviderEventAdapter — Optional adapter that bridges provider callbacks to a typed EventEmitter.
  • RecoveryOrchestrator — Pipeline-level error recovery orchestrator that coordinates recovery across providers with exponential backoff.
  • RevAISTT — Rev AI streaming STT provider using a raw WebSocket connection.
  • RimeTTS — Rime TTS provider using native fetch for text-to-speech synthesis.
  • SileroVAD — Local Silero voice-activity-detection engine.
  • SmallestTTS — Smallest.ai Waves TTS provider using native fetch for text-to-speech synthesis.
  • SonioxSTT — Soniox real-time STT provider using a raw WebSocket connection.
  • SpeechifyTTS — Speechify TTS provider using native fetch for text-to-speech synthesis.
  • SpeechmaticsSTT — Speechmatics real-time STT provider using a raw WebSocket connection.
  • TeamsCall — Duplex provider that joins a Microsoft Teams meeting via Azure Communication Services and bridges its audio to the pipeline.
  • TranscribeSTT — Amazon Transcribe real-time STT provider using the streaming WebSocket API.
  • TTSBackpressure — A counting semaphore that provides backpressure between the LLM and TTS pipeline stages.
  • TurnMetricsCollector — Collects per-turn timing marks and emits a TurnMetricsSummary when each turn finishes.
  • TurnTakingController — Manages pause/resume of audio capture during TTS playback.
  • TwilioMediaStream — Duplex input/output provider for Twilio Media Streams phone calls.
  • VADProcessor — Converts pipeline audio into VAD frames and tracks speech segments.
  • VonageAudioSocket — Duplex audio provider for the Vonage Voice API WebSocket endpoint.
  • WebLLMLLM — WebLLM in-browser LLM provider.
  • WebRTCInput — Browser audio input provider that extracts PCM from a WebRTC track.
  • WebRTCOutput — Browser audio output provider that renders TTS audio into a WebRTC track.
  • WebSocketManager — Managed WebSocket connection with automatic reconnection and exponential backoff.
  • ZoomRtmsInput — Zoom Realtime Media Streams (RTMS) input provider — streams live meeting audio into the CompositeVoice pipeline.

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