pypi pipecat-ai 0.0.59
v0.0.59
on Python PyPI

latest releases: 1.2.1, 1.2.0, 1.1.0...
14 months ago

Added

  • When registering a function call it is now possible to indicate if you want the function call to be cancelled if there's a user interruption via cancel_on_interruption (defaults to False). This is now possible because function calls are executed concurrently.

  • Added support for detecting idle pipelines. By default, if no activity has been detected during 5 minutes, the PipelineTask will be automatically cancelled. It is possible to override this behavior by passing cancel_on_idle_timeout=False. It is also possible to change the default timeout with idle_timeout_secs or the frames that prevent the pipeline from being idle with idle_timeout_frames. Finally, an on_idle_timeout event handler will be triggered if the idle timeout is reached (whether the pipeline task is cancelled or not).

  • Added FalSTTService, which provides STT for Fal's Wizper API.

  • Added a reconnect_on_error parameter to websocket-based TTS services as well as a on_connection_error event handler. The reconnect_on_error indicates whether the TTS service should reconnect on error. The on_connection_error will always get called if there's any error no matter the value of reconnect_on_error. This allows, for example, to fallback to a different TTS provider if something goes wrong with the current one.

  • Added new SkipTagsAggregator that extends BaseTextAggregator to aggregate text and skips end of sentence matching if aggregated text is between start/end tags.

  • Added new PatternPairAggregator that extends BaseTextAggregator to identify content between matching pattern pairs in streamed text. This allows for detection and processing of structured content like XML-style tags that may span across multiple text chunks or sentence boundaries.

  • Added new BaseTextAggregator. Text aggregators are used by the TTS service to aggregate LLM tokens and decide when the aggregated text should be pushed to the TTS service. They also allow for the text to be manipulated while it's being aggregated. A text aggregator can be passed via text_aggregator to the TTS service.

  • Added new sample_rate constructor parameter to TavusVideoService to allow changing the output sample rate.

  • Added new NeuphonicTTSService.
    (see https://neuphonic.com)

  • Added new UltravoxSTTService.
    (see https://github.com/fixie-ai/ultravox)

  • Added on_frame_reached_upstream and on_frame_reached_downstream event handlers to PipelineTask. Those events will be called when a frame reaches the beginning or end of the pipeline respectively. Note that by default, the event handlers will not be called unless a filter is set with PipelineTask.set_reached_upstream_filter() or PipelineTask.set_reached_downstream_filter().

  • Added support for Chirp voices in GoogleTTSService.

  • Added a flush_audio() method to FishTTSService and LmntTTSService.

  • Added a set_language convenience method for GoogleSTTService, allowing you to set a single language. This is in addition to the set_languages method which allows you to set a list of languages.

  • Added on_user_turn_audio_data and on_bot_turn_audio_data to AudioBufferProcessor. This gives the ability to grab the audio of only that turn for both the user and the bot.

  • Added new base class BaseObject which is now the base class of FrameProcessor, PipelineRunner, PipelineTask and BaseTransport. The new BaseObject adds supports for event handlers.

  • Added support for a unified format for specifying function calling across all LLM services. 

  weather_function = FunctionSchema(
      name="get_current_weather",
      description="Get the current weather",
      properties={
          "location": {
              "type": "string",
              "description": "The city and state, e.g. San Francisco, CA",
          },
          "format": {
              "type": "string",
              "enum": ["celsius", "fahrenheit"],
              "description": "The temperature unit to use. Infer this from the user's location.",
          },
      },
      required=["location"],
  )
  tools = ToolsSchema(standard_tools=[weather_function])

  • Added speech_threshold parameter to GladiaSTTService.

  • Allow passing user (user_kwargs) and assistant (assistant_kwargs) context aggregator parameters when using create_context_aggregator(). The values are passed as a mapping that will then be converted to arguments.

  • Added speed as an InputParam for both ElevenLabsTTSService and ElevenLabsHttpTTSService.

  • Added new LLMFullResponseAggregator to aggregate full LLM completions. At every completion the on_completion event handler is triggered.

  • Added a new frame, RTVIServerMessageFrame, and RTVI message RTVIServerMessage which provides a generic mechanism for sending custom messages from server to client. The RTVIServerMessageFrame is processed by the RTVIObserver and will be delivered to the client's onServerMessage callback or ServerMessage event.

  • Added GoogleLLMOpenAIBetaService for Google LLM integration with an OpenAI-compatible interface. Added foundational example 14o-function-calling-gemini-openai-format.py.

  • Added AzureRealtimeBetaLLMService to support Azure's OpeanAI Realtime API. Added foundational example 19a-azure-realtime-beta.py.

  • Introduced GoogleVertexLLMService, a new class for integrating with Vertex AI Gemini models. Added foundational example 14p-function-calling-gemini-vertex-ai.py.

  • Added support in OpenAIRealtimeBetaLLMService for a slate of new features:

    • The 'gpt-4o-transcribe' input audio transcription model, along with new language and prompt options specific to that model.

    • The input_audio_noise_reduction session property. 

      session_properties = SessionProperties(
  # ...
  input_audio_noise_reduction=InputAudioNoiseReduction(
    type="near_field" # also supported: "far_field"
  )
  # ...
)

    • The 'semantic_vad' turn_detection session property value, a more sophisticated model for detecting when the user has stopped speaking.

    • on_conversation_item_created and on_conversation_item_updated events to OpenAIRealtimeBetaLLMService. 

      @llm.event_handler("on_conversation_item_created")
async def on_conversation_item_created(llm, item_id, item):
  # ...

@llm.event_handler("on_conversation_item_updated")
async def on_conversation_item_updated(llm, item_id, item):
  # `item` may not always be available here
  # ...

    • The retrieve_conversation_item(item_id) method for introspecting a conversation item on the server. 

      item = await llm.retrieve_conversation_item(item_id)

Changed

  • Updated OpenAISTTService to use gpt-4o-transcribe as the default transcription model.

  • Updated OpenAITTSService to use gpt-4o-mini-tts as the default TTS model.

  • Function calls are now executed in tasks. This means that the pipeline will not be blocked while the function call is being executed.

  • ⚠️ PipelineTask will now be automatically cancelled if no bot activity is happening in the pipeline. There are a few settings to configure this behavior, see PipelineTask documentation for more details.

  • All event handlers are now executed in separate tasks in order to prevent blocking the pipeline. It is possible that event handlers take some time to execute in which case the pipeline would be blocked waiting for the event handler to complete.

  • Updated TranscriptProcessor to support text output from OpenAIRealtimeBetaLLMService.

  • OpenAIRealtimeBetaLLMService and GeminiMultimodalLiveLLMService now push a TTSTextFrame.

  • Updated the default mode for CartesiaTTSService and CartesiaHttpTTSService to sonic-2.

Deprecated

  • Passing a start_callback to LLMService.register_function() is now deprecated, simply move the code from the start callback to the function call.

  • TTSService parameter text_filter is now deprecated, use text_filters instead which is now a list. This allows passing multiple filters that will be executed in order.

Removed

  • Removed deprecated audio.resample_audio(), use create_default_resampler() instead.

  • Removed deprecatedstt_service parameter from STTMuteFilter.

  • Removed deprecated RTVI processors, use an RTVIObserver instead.

  • Removed deprecated AWSTTSService, use PollyTTSService instead.

  • Removed deprecated field tier from DailyTranscriptionSettings, use model instead.

  • Removed deprecated pipecat.vad package, use pipecat.audio.vad instead.

Fixed

  • Fixed an assistant aggregator issue that could cause assistant text to be split into multiple chunks during function calls.

  • Fixed an assistant aggregator issue that was causing assistant text to not be added to the context during function calls. This could lead to duplications.

  • Fixed a SegmentedSTTService issue that was causing audio to be sent prematurely to the STT service. Instead of analyzing the volume in this service we rely on VAD events which use both VAD and volume.

  • Fixed a GeminiMultimodalLiveLLMService issue that was causing messages to be duplicated in the context when pushing LLMMessagesAppendFrame frames.

  • Fixed an issue with SegmentedSTTService based services (e.g. GroqSTTService) that was not allow audio to pass-through downstream.

  • Fixed a CartesiaTTSService and RimeTTSService issue that would consider text between spelling out tags end of sentence.

  • Fixed a match_endofsentence issue that would result in floating point numbers to be considered an end of sentence.

  • Fixed a match_endofsentence issue that would result in emails to be considered an end of sentence.

  • Fixed an issue where the RTVI message disconnect-bot was pushing an EndFrame, resulting in the pipeline not shutting down. It now pushes an EndTaskFrame upstream to shutdown the pipeline.

  • Fixed an issue with the GoogleSTTService where stream timeouts during periods of inactivity were causing connection failures. The service now properly detects timeout errors and handles reconnection gracefully, ensuring continuous operation even after periods of silence or when using an STTMuteFilter.

  • Fixed an issue in RimeTTSService where the last line of text sent didn't result in an audio output being generated.

  • Fixed OpenAIRealtimeBetaLLMService by adding proper handling for:

    • The conversation.item.input_audio_transcription.delta server message, which was added server-side at some point and not handled client-side.
    • Errors reported by the response.done server message.

Other

  • Add foundational example 07w-interruptible-fal.py, showing FalSTTService.

  • Added a new Ultravox example examples/foundational/07u-interruptible-ultravox.py.

  • Added new Neuphonic examples examples/foundational/07v-interruptible-neuphonic.py and examples/foundational/07v-interruptible-neuphonic-http.py.

  • Added a new example examples/foundational/36-user-email-gathering.py to show how to gather user emails. The example uses's Cartesia's <spell></spell> tags and Rime spell() function to spell out the emails for confirmation.

  • Update the 34-audio-recording.py example to include an STT processor.

  • Added foundational example 35-voice-switching.py showing how to use the new PatternPairAggregator. This example shows how to encode information for the LLM to instruct TTS voice changes, but this can be used to encode any information into the LLM response, which you want to parse and use in other parts of your application.

  • Added a Pipecat Cloud deployment example to the examples directory.

  • Removed foundational examples 28b and 28c as the TranscriptProcessor no longer has an LLM depedency. Renamed foundational example 28a to 28-transcript-processor.py.

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