v0.0.57

### Added

- Added new `AudioContextWordTTSService`. This is a TTS base class for TTS services that handling multiple separate audio requests.

- Added new frames `EmulateUserStartedSpeakingFrame` and `EmulateUserStoppedSpeakingFrame` which can be used to emulated VAD behavior without VAD being present or not being triggered.

- Added a new `audio_in_stream_on_start` field to `TransportParams`.

- Added a new method `start_audio_in_streaming` in the `BaseInputTransport`.

  - This method should be used to start receiving the input audio in case the field `audio_in_stream_on_start` is set to `false`.

- Added support for the `RTVIProcessor` to handle buffered audio in `base64` format, converting it into InputAudioRawFrame for transport.

- Added support for the `RTVIProcessor` to trigger `start_audio_in_streaming` only after the `client-ready` message.

- Added new `MUTE_UNTIL_FIRST_BOT_COMPLETE` strategy to `STTMuteStrategy`. This strategy starts muted and remains muted until the first bot speech completes, ensuring the bot's first response cannot be interrupted. This complements the existing `FIRST_SPEECH` strategy which only mutes during the first detected bot speech.

- Added support for Google Cloud Speech-to-Text V2 through `GoogleSTTService`.

- Added `RimeTTSService`, a new `WordTTSService`. Updated the foundational example `07q-interruptible-rime.py` to use `RimeTTSService`.

- Added support for Groq's Whisper API through the new `GroqSTTService` and OpenAI's Whisper API through the new `OpenAISTTService`. Introduced a new base class `BaseWhisperSTTService` to handle common Whisper API functionality.

- Added `PerplexityLLMService` for Perplexity NIM API integration, with an OpenAI-compatible interface. Also, added foundational example `14n-function-calling-perplexity.py`.

- Added `DailyTransport.update_remote_participants()`. This allows you to update remote participant's settings, like their permissions or which of their devices are enabled. Requires that the local participant have participant admin permission.

### Changed

- We don't consider a colon `:` and end of sentence any more.

- Updated `DailyTransport` to respect the `audio_in_stream_on_start` field, ensuring it only starts receiving the audio input if it is enabled.

- Updated `FastAPIWebsocketOutputTransport` to send `TransportMessageFrame` and `TransportMessageUrgentFrame` to the serializer.

- Updated `WebsocketServerOutputTransport` to send `TransportMessageFrame` and `TransportMessageUrgentFrame` to the serializer.

- Enhanced `STTMuteConfig` to validate strategy combinations, preventing `MUTE_UNTIL_FIRST_BOT_COMPLETE` and `FIRST_SPEECH` from being used together as they handle first bot speech differently.

- Updated foundational example `07n-interruptible-google.py` to use all Google services.

- `RimeHttpTTSService` now uses the `mistv2` model by default.

- Improved error handling in `AzureTTSService` to properly detect and log synthesis cancellation errors.

- Enhanced `WhisperSTTService` with full language support and improved model documentation.

- Updated foundation example `14f-function-calling-groq.py` to use `GroqSTTService` for transcription.

- Updated `GroqLLMService` to use `llama-3.3-70b-versatile` as the default model.

- `RTVIObserver` doesn't handle `LLMSearchResponseFrame` frames anymore. For now, to handle those frames you need to create a `GoogleRTVIObserver` instead.

### Deprecated

- `STTMuteFilter` constructor's `stt_service` parameter is now deprecated and will be removed in a future version. The filter now manages mute state internally instead of querying the STT service.

- `RTVI.observer()` is now deprecated, instantiate an `RTVIObserver` directly instead.

- All RTVI frame processors (e.g. `RTVISpeakingProcessor`, `RTVIBotLLMProcessor`) are now deprecated, instantiate an `RTVIObserver` instead.

### Fixed

- Fixed a `FalImageGenService` issue that was causing the event loop to be blocked while loading the downloadded image.

- Fixed a `CartesiaTTSService` service issue that would cause audio overlapping in some cases.

- Fixed a websocket-based service issue (e.g. `CartesiaTTSService`) that was preventing a reconnection after the server disconnected cleanly, which was causing an inifite loop instead.

- Fixed a `BaseOutputTransport` issue that was causing upstream frames to no be pushed upstream.

- Fixed multiple issue where user transcriptions where not being handled properly. It was possible for short utterances to not trigger VAD which would cause user transcriptions to be ignored. It was also possible for one or more transcriptions to be generated after VAD in which case they would also be ignored.

- Fixed an issue that was causing `BotStoppedSpeakingFrame` to be generated too late. This could then cause issues unblocking `STTMuteFilter` later than desired.

- Fixed an issue that was causing `AudioBufferProcessor` to not record synchronized audio.

- Fixed an `RTVI` issue that was causing `bot-tts-text` messages to be sent before being processed by the output transport.

- Fixed an issue[#1192] in 11labs where we are trying to reconnect/disconnect the websocket connection even when the connection is already closed.

- Fixed an issue where `has_regular_messages` condition was always true in `GoogleLLMContext` due to `Part` having `function_call` & `function_response` with `None` values.

### Other

- Added new `instant-voice` example. This example showcases how to enable instant voice communication as soon as a user connects.

- Added new `local-input-select-stt` example. This examples allows you to play with local audio inputs by slecting them through a nice text interface.