Update changelog for version 0.0.100
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committed by
Aleix Conchillo Flaqué
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59ed422052
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- Added Hathora service to support Hathora-hosted TTS and STT models (only non-streaming)
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- Enhanced interruption handling in `AsyncAITTSService` by supporting multi-context WebSocket sessions for more robust context management.
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- Corrected TTFB metric calculation in `AsyncAIHttpTTSService`.
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- Added `CambTTSService`, using Camb.ai's TTS integration with MARS models (mars-flash, mars-pro, mars-instruct) for high-quality text-to-speech synthesis.
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- Fixed an issue where the "bot-llm-text" RTVI event would not fire for realtime (speech-to-speech) services:
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- `AWSNovaSonicLLMService`
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- `GeminiLiveLLMService`
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- `OpenAIRealtimeLLMService`
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- `GrokRealtimeLLMService`
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The issue was that these services weren't pushing `LLMTextFrame`s. Now they do.
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- Fixed an issue where `on_user_turn_stop_timeout` could fire while a user is talking when using `ExternalUserTurnStrategies`.
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- Fixed an issue where user turn start strategies were not being reset after a user turn started, causing incorrect strategy behavior.
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- Added the `additional_headers` param to `WebsocketClientParams`, allowing `WebsocketClientTransport` to send custom headers on connect, for cases such as authentication.
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- Fixed `MinWordsUserTurnStartStrategy` to not aggregate transcriptions, preventing incorrect turn starts when words are spoken with pauses between them.
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- For consistency with other package names, we just deprecated `pipecat.turns.mute` (introduced in Pipecat 0.0.99) in favor of `pipecat.turns.user_mute`.
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- Fixed an issue where Grok Realtime would error out when running with SmallWebRTC transport.
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- Added `UserIdleController` for detecting user idle state, integrated into `LLMUserAggregator` and `UserTurnProcessor` via optional `user_idle_timeout` parameter. Emits `on_user_turn_idle` event for application-level handling. Deprecated `UserIdleProcessor` in favor of the new compositional approach.
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- Throttle `UserSpeakingFrame` to broadcast at most every 200ms instead of on every audio chunk, reducing frame processing overhead during user speech.
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- Fixed a `Mem0MemoryService` issue where passing `async_mode: true` was causing an error. See https://docs.mem0.ai/platform/features/async-mode-default-change.
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- Fixed `AWSNovaSonicLLMService.reset_conversation()`, which would previously error out. Now it successfully reconnects and "rehydrates" from the context object.
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- Fixed `AzureTTSService` transcript formatting issues:
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- Punctuation now appears without extra spaces (e.g., "Hello!" instead of "Hello !")
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- CJK languages (Chinese, Japanese, Korean) no longer have unwanted spaces between characters
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- Added `on_user_mute_started` and `on_user_mute_stopped` event handlers to `LLMUserAggregator` for tracking user mute state changes.
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- Fixed an issue where `UninterruptibleFrame` frames would not be preserved in some cases.
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- Fixed memory leak in `LiveKitTransport` when `video_in_enabled` is `False`.
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- Fixed an issue in `AIService` where unhandled exceptions in `start()`, `stop()`, or `cancel()` implementations would prevent `process_frame()` to continue and therefore `StartFrame`, `EndFrame`, or `CancelFrame` from being pushed downstream, causing the pipeline to not start or stop properly.
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- Moved `NVIDIATTSService` and `NVIDIASTTService` client initialization from constructor to `start()` for better error handling.
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- Optimized `NVIDIATTSService` to process incoming audio frames immediately.
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- Optimized `NVIDIASTTService` by removing unnecessary queue and task.
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- Fixed a `CambTTSService` issue where client was being initialized in the constructor which wouldn't allow for proper Pipeline error handling.
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