v0.0.66

### Added

- Added two new input parameters to `RimeTTSService`: `pause_between_brackets` and `phonemize_between_brackets`.

- Added support for cross-platform local smart turn detection. You can use `LocalSmartTurnAnalyzer` for on-device inference using Torch.

- `BaseOutputTransport` now allows multiple destinations if the transport implementation supports it (e.g. Daily's custom tracks). With multiple destinations it is possible to send different audio or video tracks with a single transport simultaneously. To do that, you need to set the new `Frame.transport_destination` field with your desired transport destination (e.g. custom track name), tell the transport you want a new destination with `TransportParams.audio_out_destinations` or `TransportParams.video_out_destinations` and the transport should take care of the rest.

- Similar to the new `Frame.transport_destination`, there's a new `Frame.transport_source` field which is set by the `BaseInputTransport` if the incoming data comes from a non-default source (e.g. custom tracks).

- `TTSService` has a new `transport_destination` constructor parameter. This parameter will be used to update the `Frame.transport_destination` field for each generated `TTSAudioRawFrame`. This allows sending multiple bots' audio to multiple destinations in the same pipeline.

- Added `DailyTransportParams.camera_out_enabled` and `DailyTransportParams.microphone_out_enabled` which allows you to enable/disable the main output camera or microphone tracks. This is useful if you only want to use custom tracks and not send the main tracks. Note that you still need `audio_out_enabled=True` or `video_out_enabled`.

- Added `DailyTransport.capture_participant_audio()` which allows you to capture an audio source (e.g. "microphone", "screenAudio" or a custom track name) from a remote participant.

- Added `DailyTransport.update_publishing()` which allows you to update the call video and audio publishing settings (e.g. audio and video quality).

- Added `RTVIObserverParams` which allows you to configure what RTVI messages are sent to the clients.

- Added a `context_window_compression` InputParam to `GeminiMultimodalLiveLLMService` which allows you to enable a sliding context window for the session as well as set the token limit of the sliding window.

- Updated `SmallWebRTCConnection` to support `ice_servers` with credentials.

- Added `VADUserStartedSpeakingFrame` and `VADUserStoppedSpeakingFrame`, indicating when the VAD detected the user to start and stop speaking. These events are helpful when using smart turn detection, as the user's stop time can differ from when their turn ends (signified by UserStoppedSpeakingFrame).

- Added `TranslationFrame`, a new frame type that contains a translated transcription.

- Added `TransportParams.audio_in_passthrough`. If set (the default), incoming audio will be pushed downstream.

- Added `MCPClient`; a way to connect to MCP servers and use the MCP servers' tools.

- Added `Mem0 OSS`, along with Mem0 cloud support now the OSS version is also available.

### Changed

- `TransportParams.audio_mixer` now supports a string and also a dictionary to provide a mixer per destination. For example:

```python
  audio_out_mixer={
      "track-1": SoundfileMixer(...),
      "track-2": SoundfileMixer(...),
      "track-N": SoundfileMixer(...),
  },
```

- The `STTMuteFilter` now mutes `InterimTranscriptionFrame` and `TranscriptionFrame` which allows the `STTMuteFilter` to be used in conjunction with transports that generate transcripts, e.g. `DailyTransport`.

- Function calls now receive a single parameter `FunctionCallParams` instead of `(function_name, tool_call_id, args, llm, context, result_callback)` which is now deprecated.

- Changed the user aggregator timeout for late transcriptions from 1.0s to 0.5s (`LLMUserAggregatorParams.aggregation_timeout`). Sometimes, the STT services might give us more than one transcription which could come after the user stopped speaking. We still want to include these additional transcriptions with the first one because it's part of the user turn. This is what this timeout is helpful with.

- Short utterances not detected by VAD while the bot is speaking are now ignored. This reduces the amount of bot interruptions significantly providing a more natural conversation experience.

- Updated `GladiaSTTService` to output a `TranslationFrame` when specifying a `translation` and `translation_config`.

- STT services now passthrough audio frames by default. This allows you to add audio recording without worrying about what's wrong in your pipeline when it doesn't work the first time.

- Input transports now always push audio downstream unless disabled with `TransportParams.audio_in_passthrough`. After many Pipecat releases, we realized this is the common use case. There are use cases where the input transport already provides STT and you also don't want recordings, in which case there's no need to push audio to the rest of the pipeline, but this is not a very common case.

- Added `RivaSegmentedSTTService`, which allows Riva offline/batch models, such as to be "canary-1b-asr" used in Pipecat.

### Deprecated

- Function calls with parameters `(function_name, tool_call_id, args, llm, context, result_callback)` are deprectated, use a single `FunctionCallParams` parameter instead.

- `TransportParams.camera_*` parameters are now deprecated, use `TransportParams.video_*` instead.

- `TransportParams.vad_enabled` parameter is now deprecated, use `TransportParams.audio_in_enabled` and `TransportParams.vad_analyzer` instead.

- `TransportParams.vad_audio_passthrough` parameter is now deprecated, use `TransportParams.audio_in_passthrough` instead.

- `ParakeetSTTService` is now deprecated, use `RivaSTTService` instead, which uses the model "parakeet-ctc-1.1b-asr" by default.

- `FastPitchTTSService` is now deprecated, use `RivaTTSService` instead, which uses the model "magpie-tts-multilingual" by default.

### Fixed

- Fixed an issue with `SimliVideoService` where the bot was continuously outputting audio, which prevents the `BotStoppedSpeakingFrame` from being emitted.

- Fixed an issue where `OpenAIRealtimeBetaLLMService` would add two assistant messages to the context.

- Fixed an issue with `GeminiMultimodalLiveLLMService` where the context contained tokens instead of words.

- Fixed an issue with HTTP Smart Turn handling, where the service returns a 500 error. Previously, this would cause an unhandled exception. Now, a 500 error is treated as an incomplete response.

- Fixed a TTS services issue that could cause assistant output not to be aggregated to the context when also using `TTSSpeakFrame`s.

- Fixed an issue where the `SmartTurnMetricsData` was reporting 0ms for inference and processing time when using the `FalSmartTurnAnalyzer`.

### Other

- Added `examples/daily-custom-tracks` to show how to send and receive Daily custom tracks.

- Added `examples/daily-multi-translation` to showcase how to send multiple simulataneous translations with the same transport.

- Added 04 foundational examples for client/server transports. Also, renamed `29-livekit-audio-chat.py` to `04b-transports-livekit.py`.

- Added foundational example `13c-gladia-translation.py` showing how to use `TranscriptionFrame` and `TranslationFrame`.