v0.0.86

### Added

- Added `HeyGenTransport`. This is an integration for HeyGen Interactive Avatar. A video service that handles audio streaming and requests HeyGen to generate avatar video responses. (see https://www.heygen.com/). When used, the Pipecat bot joins the same virtual room as the HeyGen Avatar and the user.

- Added support to `TwilioFrameSerializer` for `region` and `edge` settings.

- Added support for using universal `LLMContext` with:

  - `LLMLogObserver`
  - `GatedLLMContextAggregator` (formerly `GatedOpenAILLMContextAggregator`)
  - `LangchainProcessor`
  - `Mem0MemoryService`

- Added `StrandsAgentProcessor` which allows you to use the Strands Agents framework to build your voice agents.
  See https://strandsagents.com

- Added `ElevenLabsSTTService` for speech-to-text transcription.

- Added a peer connection monitor to the `SmallWebRTCConnection` that automatically disconnects if the connection fails to establish within the timeout (1 minute by default).

- Added memory cleanup improvements to reduce memory peaks.

- Added `on_before_process_frame`, `on_after_process_frame`, `on_before_push_frame` and `on_after_push_frame`. These are synchronous events that get called before and after a frame is processed or pushed. Note that these events are synchrnous so they should ideally perform lightweight tasks in order to not block the pipeline. See `examples/foundational/45-before-and-after-events.py`.

- Added `on_before_leave` synchronous event to `DailyTransport`.

- Added `on_before_disconnect` synchronous event to `LiveKitTransport`.

- It is now possible to register synchronous event handlers. By default, all event handlers are executed in a separate task. However, in some cases we want to guarantee order of execution, for example, executing something before disconnecting a transport.

  ```python
  self._register_event_handler("on_event_name", sync=True)
  ```

- Added support for global location in `GoogleVertexLLMService`. The service now supports both regional locations (e.g., "us-east4") and the "global" location for Vertex AI endpoints. When using "global" location, the service will use `aiplatform.googleapis.com` as the API host instead of the regional format.

- Added `on_pipeline_finished` event to `PipelineTask`. This event will get fired when the pipeline is done running. This can be the result of a `StopFrame`, `CancelFrame` or `EndFrame`.

  ```python
  @task.event_handler("on_pipeline_finished")
  async def on_pipeline_finished(task: PipelineTask, frame: Frame):
      ...
  ```

- Added support for new RTVI `send-text` event, along with the ability to toggle the audio response off (skip tts) while handling the new context.

### Changed

- Updated `aiortc` to 1.13.0.

- Updated `sentry` to 2.38.0.

- `BaseOutputTransport` methods `write_audio_frame` and `write_video_frame` now return a boolean to indicate if the transport implementation was able to write the given frame or not.

- Updated Silero VAD model to v6.

- Updated `livekit` to 1.0.13.

- `torch` and `torchaudio` are no longer required for running Smart Turn locally. This avoids gigabytes of dependencies being installed.

- Updated `websockets` dependency to support version 15.0. Removed deprecated usage of `ConnectionClosed.code` and `ConnectionClosed.reason` attributes in `AWSTranscribeSTTService` for compatibility.

- Refactored `pyproject.toml` to reduce websockets dependency repetition using self-referencing extras. All websockets-dependent services now reference a shared `websockets-base` extra.

### Deprecated

- `GladiaSTTService`'s `confidence` arg is deprecated. `confidence` is no longer needed to determine which transcription or translation frames to emit.

- `PipelineTask` events `on_pipeline_stopped`, `on_pipeline_ended` and `on_pipeline_cancelled` are now deprecated. Use `on_pipeline_finished` instead.

- Support for the RTVI `append-to-context` event, in lieu of the new `send-text` event and making way for future events like `send-image`.

### Fixed

- Fixed an issue where the pipeline could freeze if a task cancellation never completed because a third-party library swallowed asyncio.CancelledError. We now apply a timeout to task cancellations to prevent these freezes. If the timeout is reached, the system logs warnings and leaves dangling tasks behind, which can help diagnose where cancellation is being blocked.

- Fixed an `AudioBufferProcessor` issues that was causing user audio to be missing in stereo recordings causing bot and user overlaps.

- Fixed a `BaseOutputTransport` issue that could produce large saved `AudioBufferProcessor` files when using an audio mixer.

- Fixed a `PipelineRunner` issue on Windows where setting up SIGINT and SIGTERM was raising an exception.

- Fixed an issue where multiple handlers for an event would not run in parallel.

- Fixed `DailyTransport.sip_call_transfer()` to automatically use the session ID from the `on_dialin_connected` event, when not explicitly provided. Now supports cold transfers (from incoming dial-in calls) by automatically tracking session IDs from connection events.

- Fixed a memory leak in `SmallWebRTCTransport`. In `aiortc`, when you receive a `MediaStreamTrack` (audio or video), frames are produced asynchronously. If the code never consumes these frames, they are queued in memory, causing a memory leak.

- Fixed an issue in `AsyncAITTSService`, where `TTSTextFrames` were not being pushed.

- Fixed an issue that would cause `push_interruption_task_frame_and_wait()` to not wait if a previous interruption had already happened.

- Fixed a couple of bugs in `ServiceSwitcher`:

  - Using multiple `ServiceSwitcher`s in a pipeline would result in an error.
  - `ServiceSwitcherFrame`s (such as `ManuallySwitchServiceFrame`s) were having an effect too early, essentially "jumping the queue" in terms of pipeline frame ordering.

- Fixed a self-cancellation deadlock in `UserIdleProcessor` when returning `False` from an idle callback. The task now terminates naturally instead of attempting to cancel itself.

- Fixed an issue in `AudioBufferProcessor` where a recording is not created when a bot speaks and user input is blocked.

- Fixed a `FastAPIWebsocketTransport` and `SmallWebRTCTransport` issue where `on_client_disconnected` would be triggered when the bot ends the conversation. That is, `on_client_disconnected` should only be triggered when the remote client actually disconnects.

- Fixed an issue in `HeyGenVideoService` where the `BotStartedSpeakingFrame` was blocked from moving through the Pipeline.