- Introduce mechanisms in the pipeline to ensure that the end call process waits for the completion of the end speech before hanging up, improving user experience during call termination.
- Update the useVoicePreview hook to handle server-initiated call endings gracefully, distinguishing between normal and error disconnections.
- Adjust TTS stop frame timeout settings to optimize the timing of call terminations, ensuring timely responses without unnecessary delays.
- Refactor related components to support the new end call logic, enhancing overall workflow management and user interaction.
- Introduce a new WorkflowEngine class to manage workflow graphs, enabling dynamic node-based interactions.
- Update AssistantConfig to include a graph field for workflow definitions, allowing for flexible configuration.
- Modify pipeline execution to support workflow-driven dialogue, integrating node transitions and system prompts based on active nodes.
- Enhance frontend components to visualize active nodes and provide debugging capabilities, including highlighting the current node during interactions.
- Refactor existing components to accommodate new workflow functionalities and improve overall user experience.
- Introduce event handlers in PassthroughLLMAssistantAggregator for managing LLM text streaming, including start, delta, and end events.
- Implement a new method to finalize text streams, ensuring proper handling of interruptions.
- Update useVoicePreview to support new message types for LLM text streaming, allowing real-time updates to chat messages.
- Enhance message sorting logic to maintain order based on timestamps and sequence numbers, improving user experience during voice interactions.
- Refactor TextInputProcessor to handle immediate and silent text inputs, improving user experience during voice interactions.
- Introduce PassthroughLLMAssistantAggregator to manage LLM responses while preserving context for downstream TTS processing.
- Update event handling for text input and client readiness, ensuring timely updates to the conversation context.
- Modify run_pipeline to integrate new aggregators and streamline message handling, enhancing overall pipeline efficiency.
- Improve message ordering in useVoicePreview to ensure accurate display of chat messages based on timestamps.
- Rename `setSelectedDeviceId` to `selectDevice` in `DebugVoicePanel` and `VoiceSessionControls` for clarity and consistency.
- Update `useVoicePreview` hook to implement the `selectDevice` function, enabling dynamic microphone switching during voice sessions.
- Enhance device selection logic to support real-time audio track replacement without requiring session reconnection.
- Introduce new Xfyun ASR and TTS services, enabling integration with iFlytek's voice recognition and synthesis capabilities.
- Update AssistantConfig model to include interface types for STT and TTS.
- Enhance credential testing to validate Xfyun credentials.
- Modify service factory to create Xfyun services based on configuration.
- Update README with new configuration details for Xfyun integration.
- Add new frontend components for visualizing audio streams and managing user interactions.
- Add audio input selection to DebugVoicePanel, allowing users to choose their microphone device.
- Update useVoicePreview hook to manage available audio inputs and selected device state.
- Enhance device enumeration and selection handling to ensure a seamless user experience during voice interactions.
- Add a new Docker configuration for the UI in launch.json to facilitate development.
- Refactor pipeline.py to integrate a TranscriptProcessor for managing user and assistant transcripts, including event handlers for real-time updates and message handling.
- Update useVoicePreview.ts to establish a data channel for sending and receiving text messages, improving interaction flow.
- Modify AssistantPage.tsx to support displaying chat messages and sending user input, enhancing the user experience during voice interactions.
- Revise DebugTranscriptPanel to dynamically render chat messages with timestamps, improving the visual representation of conversation history.
- Update README to reflect the integration of the DebugVoicePanel with WebSocket support for voice interactions.
- Refactor voice_webrtc.py to improve error handling during WebRTC signaling and include assistant_id in the offer payload.
- Add useVoicePreview hook to manage microphone access and WebRTC connections for real-time voice previews.
- Modify AssistantPage to incorporate new visualizer options and pass assistantId to DebugVoicePanel, enhancing user experience during audio interactions.
- Update API model to include new fields for voice, speed, and language, supporting TTS and ASR configurations.
