pypi pipecat-ai 0.0.57
v0.0.57
on Python PyPI

latest releases: 1.1.0, 1.0.0, 0.0.108...
14 months ago

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.

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