pipecat-ai 0.0.66

v0.0.66

on Python PyPI

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:

  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 TTSSpeakFrames.

• 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.