v0.0.50

### Added

- Added `GeminiMultimodalLiveLLMService`. This is an integration for Google's Gemini Multimodal Live API, supporting:

  - Real-time audio and video input processing
  - Streaming text responses with TTS
  - Audio transcription for both user and bot speech
  - Function calling
  - System instructions and context management
  - Dynamic parameter updates (temperature, top_p, etc.)

- Added `AudioTranscriber` utility class for handling audio transcription with Gemini models.

- Added new context classes for Gemini:

  - `GeminiMultimodalLiveContext`
  - `GeminiMultimodalLiveUserContextAggregator`
  - `GeminiMultimodalLiveAssistantContextAggregator`
  - `GeminiMultimodalLiveContextAggregatorPair`

- Added new foundational examples for `GeminiMultimodalLiveLLMService`:

  - `26-gemini-multimodal-live.py`
  - `26a-gemini-multimodal-live-transcription.py`
  - `26b-gemini-multimodal-live-video.py`
  - `26c-gemini-multimodal-live-video.py`

- Added `SimliVideoService`. This is an integration for Simli AI avatars.
  (see https://www.simli.com)

- Added NVIDIA Riva's `FastPitchTTSService` and `ParakeetSTTService`.
  (see https://www.nvidia.com/en-us/ai-data-science/products/riva/)

- Added `IdentityFilter`. This is the simplest frame filter that lets through all incoming frames.

- New `STTMuteStrategy` called `FUNCTION_CALL` which mutes the STT service during LLM function calls.

- `DeepgramSTTService` now exposes two event handlers `on_speech_started` and `on_utterance_end` that could be used to implement interruptions. See new example `examples/foundational/07c-interruptible-deepgram-vad.py`.

- Added `GroqLLMService`, `GrokLLMService`, and `NimLLMService` for Groq, Grok, and NVIDIA NIM API integration, with an OpenAI-compatible interface.

- New examples demonstrating function calling with Groq, Grok, Azure OpenAI, Fireworks, and NVIDIA NIM: `14f-function-calling-groq.py`, `14g-function-calling-grok.py`, `14h-function-calling-azure.py`, `14i-function-calling-fireworks.py`, and `14j-function-calling-nvidia.py`.

- In order to obtain the audio stored by the `AudioBufferProcessor` you can now also register an `on_audio_data` event handler. The `on_audio_data` handler will be called every time `buffer_size` (a new constructor argument) is reached. If `buffer_size` is 0 (default) you need to manually get the audio as before using `AudioBufferProcessor.merge_audio_buffers()`.

```
@audiobuffer.event_handler("on_audio_data")
async def on_audio_data(processor, audio, sample_rate, num_channels):
    await save_audio(audio, sample_rate, num_channels)
```

- Added a new RTVI message called `disconnect-bot`, which when handled pushes an `EndFrame` to trigger the pipeline to stop.

### Changed

- `STTMuteFilter` now supports multiple simultaneous muting strategies.

- `XTTSService` language now defaults to `Language.EN`.

- `SoundfileMixer` doesn't resample input files anymore to avoid startup delays. The sample rate of the provided sound files now need to match the sample rate of the output transport.

- Input frames (audio, image and transport messages) are now system frames. This means they are processed immediately by all processors instead of being queued internally.

- Expanded the transcriptions.language module to support a superset of languages.

- Updated STT and TTS services with language options that match the supported languages for each service.

- Updated the `AzureLLMService` to use the `OpenAILLMService`. Updated the `api_version` to `2024-09-01-preview`.

- Updated the `FireworksLLMService` to use the `OpenAILLMService`. Updated the default model to `accounts/fireworks/models/firefunction-v2`.

- Updated the `simple-chatbot` example to include a Javascript and React client example, using RTVI JS and React.

### Removed

- Removed `AppFrame`. This was used as a special user custom frame, but there's actually no use case for that.

### Fixed

- Fixed a `ParallelPipeline` issue that would cause system frames to be queued.

- Fixed `FastAPIWebsocketTransport` so it can work with binary data (e.g. using the protobuf serializer).

- Fixed an issue in `CartesiaTTSService` that could cause previous audio to be received after an interruption.

- Fixed Cartesia, ElevenLabs, LMNT and PlayHT TTS websocket reconnection. Before, if an error occurred no reconnection was happening.

- Fixed a `BaseOutputTransport` issue that was causing audio to be discarded after an `EndFrame` was received.

- Fixed an issue in `WebsocketServerTransport` and `FastAPIWebsocketTransport` that would cause a busy loop when using audio mixer.

- Fixed a `DailyTransport` and `LiveKitTransport` issue where connections were being closed in the input transport prematurely. This was causing frames queued inside the pipeline being discarded.

- Fixed an issue in `DailyTransport` that would cause some internal callbacks to not be executed.

- Fixed an issue where other frames were being processed while a `CancelFrame` was being pushed down the pipeline.

- `AudioBufferProcessor` now handles interruptions properly.

- Fixed a `WebsocketServerTransport` issue that would prevent interruptions with `TwilioSerializer` from working.

- `DailyTransport.capture_participant_video` now allows capturing user's screen share by simply passing `video_source="screenVideo"`.

- Fixed Google Gemini message handling to properly convert appended messages to Gemini's required format.

- Fixed an issue with `FireworksLLMService` where chat completions were failing by removing the `stream_options` from the chat completion options.