dv-pipecat-ai 0.0.82.dev857__py3-none-any.whl → 0.0.85.dev837__py3-none-any.whl
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- {dv_pipecat_ai-0.0.82.dev857.dist-info → dv_pipecat_ai-0.0.85.dev837.dist-info}/METADATA +98 -130
- {dv_pipecat_ai-0.0.82.dev857.dist-info → dv_pipecat_ai-0.0.85.dev837.dist-info}/RECORD +192 -140
- pipecat/adapters/base_llm_adapter.py +38 -1
- pipecat/adapters/services/anthropic_adapter.py +9 -14
- pipecat/adapters/services/aws_nova_sonic_adapter.py +120 -5
- pipecat/adapters/services/bedrock_adapter.py +236 -13
- pipecat/adapters/services/gemini_adapter.py +12 -8
- pipecat/adapters/services/open_ai_adapter.py +19 -7
- pipecat/adapters/services/open_ai_realtime_adapter.py +5 -0
- pipecat/audio/dtmf/dtmf-0.wav +0 -0
- pipecat/audio/dtmf/dtmf-1.wav +0 -0
- pipecat/audio/dtmf/dtmf-2.wav +0 -0
- pipecat/audio/dtmf/dtmf-3.wav +0 -0
- pipecat/audio/dtmf/dtmf-4.wav +0 -0
- pipecat/audio/dtmf/dtmf-5.wav +0 -0
- pipecat/audio/dtmf/dtmf-6.wav +0 -0
- pipecat/audio/dtmf/dtmf-7.wav +0 -0
- pipecat/audio/dtmf/dtmf-8.wav +0 -0
- pipecat/audio/dtmf/dtmf-9.wav +0 -0
- pipecat/audio/dtmf/dtmf-pound.wav +0 -0
- pipecat/audio/dtmf/dtmf-star.wav +0 -0
- pipecat/audio/filters/krisp_viva_filter.py +193 -0
- pipecat/audio/filters/noisereduce_filter.py +15 -0
- pipecat/audio/turn/base_turn_analyzer.py +9 -1
- pipecat/audio/turn/smart_turn/base_smart_turn.py +14 -8
- pipecat/audio/turn/smart_turn/data/__init__.py +0 -0
- pipecat/audio/turn/smart_turn/data/smart-turn-v3.0.onnx +0 -0
- pipecat/audio/turn/smart_turn/http_smart_turn.py +6 -2
- pipecat/audio/turn/smart_turn/local_smart_turn.py +1 -1
- pipecat/audio/turn/smart_turn/local_smart_turn_v2.py +1 -1
- pipecat/audio/turn/smart_turn/local_smart_turn_v3.py +124 -0
- pipecat/audio/vad/data/README.md +10 -0
- pipecat/audio/vad/data/silero_vad_v2.onnx +0 -0
- pipecat/audio/vad/silero.py +9 -3
- pipecat/audio/vad/vad_analyzer.py +13 -1
- pipecat/extensions/voicemail/voicemail_detector.py +5 -5
- pipecat/frames/frames.py +277 -86
- pipecat/observers/loggers/debug_log_observer.py +3 -3
- pipecat/observers/loggers/llm_log_observer.py +7 -3
- pipecat/observers/loggers/user_bot_latency_log_observer.py +22 -10
- pipecat/pipeline/runner.py +18 -6
- pipecat/pipeline/service_switcher.py +64 -36
- pipecat/pipeline/task.py +125 -79
- pipecat/pipeline/tts_switcher.py +30 -0
- pipecat/processors/aggregators/dtmf_aggregator.py +2 -3
- pipecat/processors/aggregators/{gated_openai_llm_context.py → gated_llm_context.py} +9 -9
- pipecat/processors/aggregators/gated_open_ai_llm_context.py +12 -0
- pipecat/processors/aggregators/llm_context.py +40 -2
- pipecat/processors/aggregators/llm_response.py +32 -15
- pipecat/processors/aggregators/llm_response_universal.py +19 -15
- pipecat/processors/aggregators/user_response.py +6 -6
- pipecat/processors/aggregators/vision_image_frame.py +24 -2
- pipecat/processors/audio/audio_buffer_processor.py +43 -8
- pipecat/processors/dtmf_aggregator.py +174 -77
- pipecat/processors/filters/stt_mute_filter.py +17 -0
- pipecat/processors/frame_processor.py +110 -24
- pipecat/processors/frameworks/langchain.py +8 -2
- pipecat/processors/frameworks/rtvi.py +210 -68
- pipecat/processors/frameworks/strands_agents.py +170 -0
- pipecat/processors/logger.py +2 -2
- pipecat/processors/transcript_processor.py +26 -5
- pipecat/processors/user_idle_processor.py +35 -11
- pipecat/runner/daily.py +59 -20
- pipecat/runner/run.py +395 -93
- pipecat/runner/types.py +6 -4
- pipecat/runner/utils.py +51 -10
- pipecat/serializers/__init__.py +5 -1
- pipecat/serializers/asterisk.py +16 -2
- pipecat/serializers/convox.py +41 -4
- pipecat/serializers/custom.py +257 -0
- pipecat/serializers/exotel.py +5 -5
- pipecat/serializers/livekit.py +20 -0
- pipecat/serializers/plivo.py +5 -5
- pipecat/serializers/protobuf.py +6 -5
- pipecat/serializers/telnyx.py +2 -2
- pipecat/serializers/twilio.py +43 -23
- pipecat/serializers/vi.py +324 -0
- pipecat/services/ai_service.py +2 -6
- pipecat/services/anthropic/llm.py +2 -25
- pipecat/services/assemblyai/models.py +6 -0
- pipecat/services/assemblyai/stt.py +13 -5
- pipecat/services/asyncai/tts.py +5 -3
- pipecat/services/aws/__init__.py +1 -0
- pipecat/services/aws/llm.py +147 -105
- pipecat/services/aws/nova_sonic/__init__.py +0 -0
- pipecat/services/aws/nova_sonic/context.py +436 -0
- pipecat/services/aws/nova_sonic/frames.py +25 -0
- pipecat/services/aws/nova_sonic/llm.py +1265 -0
- pipecat/services/aws/stt.py +3 -3
- pipecat/services/aws_nova_sonic/__init__.py +19 -1
- pipecat/services/aws_nova_sonic/aws.py +11 -1151
- pipecat/services/aws_nova_sonic/context.py +8 -354
- pipecat/services/aws_nova_sonic/frames.py +13 -17
- pipecat/services/azure/llm.py +51 -1
- pipecat/services/azure/realtime/__init__.py +0 -0
- pipecat/services/azure/realtime/llm.py +65 -0
- pipecat/services/azure/stt.py +15 -0
- pipecat/services/cartesia/stt.py +77 -70
- pipecat/services/cartesia/tts.py +80 -13
- pipecat/services/deepgram/__init__.py +1 -0
- pipecat/services/deepgram/flux/__init__.py +0 -0
- pipecat/services/deepgram/flux/stt.py +640 -0
- pipecat/services/elevenlabs/__init__.py +4 -1
- pipecat/services/elevenlabs/stt.py +339 -0
- pipecat/services/elevenlabs/tts.py +87 -46
- pipecat/services/fish/tts.py +5 -2
- pipecat/services/gemini_multimodal_live/events.py +38 -524
- pipecat/services/gemini_multimodal_live/file_api.py +23 -173
- pipecat/services/gemini_multimodal_live/gemini.py +41 -1403
- pipecat/services/gladia/stt.py +56 -72
- pipecat/services/google/__init__.py +1 -0
- pipecat/services/google/gemini_live/__init__.py +3 -0
- pipecat/services/google/gemini_live/file_api.py +189 -0
- pipecat/services/google/gemini_live/llm.py +1582 -0
- pipecat/services/google/gemini_live/llm_vertex.py +184 -0
- pipecat/services/google/llm.py +15 -11
- pipecat/services/google/llm_openai.py +3 -3
- pipecat/services/google/llm_vertex.py +86 -16
- pipecat/services/google/stt.py +4 -0
- pipecat/services/google/tts.py +7 -3
- pipecat/services/heygen/api.py +2 -0
- pipecat/services/heygen/client.py +8 -4
- pipecat/services/heygen/video.py +2 -0
- pipecat/services/hume/__init__.py +5 -0
- pipecat/services/hume/tts.py +220 -0
- pipecat/services/inworld/tts.py +6 -6
- pipecat/services/llm_service.py +15 -5
- pipecat/services/lmnt/tts.py +4 -2
- pipecat/services/mcp_service.py +4 -2
- pipecat/services/mem0/memory.py +6 -5
- pipecat/services/mistral/llm.py +29 -8
- pipecat/services/moondream/vision.py +42 -16
- pipecat/services/neuphonic/tts.py +5 -2
- pipecat/services/openai/__init__.py +1 -0
- pipecat/services/openai/base_llm.py +27 -20
- pipecat/services/openai/realtime/__init__.py +0 -0
- pipecat/services/openai/realtime/context.py +272 -0
- pipecat/services/openai/realtime/events.py +1106 -0
- pipecat/services/openai/realtime/frames.py +37 -0
- pipecat/services/openai/realtime/llm.py +829 -0
- pipecat/services/openai/tts.py +49 -10
- pipecat/services/openai_realtime/__init__.py +27 -0
- pipecat/services/openai_realtime/azure.py +21 -0
- pipecat/services/openai_realtime/context.py +21 -0
- pipecat/services/openai_realtime/events.py +21 -0
- pipecat/services/openai_realtime/frames.py +21 -0
- pipecat/services/openai_realtime_beta/azure.py +16 -0
- pipecat/services/openai_realtime_beta/openai.py +17 -5
- pipecat/services/piper/tts.py +7 -9
- pipecat/services/playht/tts.py +34 -4
- pipecat/services/rime/tts.py +12 -12
- pipecat/services/riva/stt.py +3 -1
- pipecat/services/salesforce/__init__.py +9 -0
- pipecat/services/salesforce/llm.py +700 -0
- pipecat/services/sarvam/__init__.py +7 -0
- pipecat/services/sarvam/stt.py +540 -0
- pipecat/services/sarvam/tts.py +97 -13
- pipecat/services/simli/video.py +2 -2
- pipecat/services/speechmatics/stt.py +22 -10
- pipecat/services/stt_service.py +47 -0
- pipecat/services/tavus/video.py +2 -2
- pipecat/services/tts_service.py +75 -22
- pipecat/services/vision_service.py +7 -6
- pipecat/services/vistaar/llm.py +51 -9
- pipecat/tests/utils.py +4 -4
- pipecat/transcriptions/language.py +41 -1
- pipecat/transports/base_input.py +13 -34
- pipecat/transports/base_output.py +140 -104
- pipecat/transports/daily/transport.py +199 -26
- pipecat/transports/heygen/__init__.py +0 -0
- pipecat/transports/heygen/transport.py +381 -0
- pipecat/transports/livekit/transport.py +228 -63
- pipecat/transports/local/audio.py +6 -1
- pipecat/transports/local/tk.py +11 -2
- pipecat/transports/network/fastapi_websocket.py +1 -1
- pipecat/transports/smallwebrtc/connection.py +103 -19
- pipecat/transports/smallwebrtc/request_handler.py +246 -0
- pipecat/transports/smallwebrtc/transport.py +65 -23
- pipecat/transports/tavus/transport.py +23 -12
- pipecat/transports/websocket/client.py +41 -5
- pipecat/transports/websocket/fastapi.py +21 -11
- pipecat/transports/websocket/server.py +14 -7
- pipecat/transports/whatsapp/api.py +8 -0
- pipecat/transports/whatsapp/client.py +47 -0
- pipecat/utils/base_object.py +54 -22
- pipecat/utils/redis.py +58 -0
- pipecat/utils/string.py +13 -1
- pipecat/utils/tracing/service_decorators.py +21 -21
- pipecat/serializers/genesys.py +0 -95
- pipecat/services/google/test-google-chirp.py +0 -45
- pipecat/services/openai.py +0 -698
- {dv_pipecat_ai-0.0.82.dev857.dist-info → dv_pipecat_ai-0.0.85.dev837.dist-info}/WHEEL +0 -0
- {dv_pipecat_ai-0.0.82.dev857.dist-info → dv_pipecat_ai-0.0.85.dev837.dist-info}/licenses/LICENSE +0 -0
- {dv_pipecat_ai-0.0.82.dev857.dist-info → dv_pipecat_ai-0.0.85.dev837.dist-info}/top_level.txt +0 -0
- /pipecat/services/{aws_nova_sonic → aws/nova_sonic}/ready.wav +0 -0
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output_sample_rate: Audio output sample rate in Hz.
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output_channel_count: Number of output audio channels.
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max_tokens: Maximum number of tokens to generate.
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top_p: Nucleus sampling parameter.
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temperature: Sampling temperature for text generation.
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"""
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# Audio input
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input_sample_rate: Optional[int] = Field(default=16000)
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# Audio output
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output_sample_rate: Optional[int] = Field(default=24000)
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# Inference
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temperature: Optional[float] = Field(default=0.7)
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class AWSNovaSonicLLMService(LLMService):
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"""AWS Nova Sonic speech-to-speech LLM service.
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Provides bidirectional audio streaming, real-time transcription, text generation,
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and function calling capabilities using AWS Nova Sonic model.
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"""
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# Override the default adapter to use the AWSNovaSonicLLMAdapter one
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adapter_class = AWSNovaSonicLLMAdapter
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def __init__(
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self,
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*,
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secret_access_key: str,
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access_key_id: str,
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session_token: Optional[str] = None,
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region: str,
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model: str = "amazon.nova-sonic-v1:0",
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voice_id: str = "matthew", # matthew, tiffany, amy
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params: Optional[Params] = None,
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system_instruction: Optional[str] = None,
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tools: Optional[ToolsSchema] = None,
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send_transcription_frames: bool = True,
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**kwargs,
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):
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"""Initializes the AWS Nova Sonic LLM service.
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Args:
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secret_access_key: AWS secret access key for authentication.
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access_key_id: AWS access key ID for authentication.
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session_token: AWS session token for authentication.
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region: AWS region where the service is hosted.
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model: Model identifier. Defaults to "amazon.nova-sonic-v1:0".
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voice_id: Voice ID for speech synthesis. Options: matthew, tiffany, amy.
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params: Model parameters for audio configuration and inference.
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system_instruction: System-level instruction for the model.
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tools: Available tools/functions for the model to use.
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send_transcription_frames: Whether to emit transcription frames.
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**kwargs: Additional arguments passed to the parent LLMService.
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"""
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super().__init__(**kwargs)
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self._secret_access_key = secret_access_key
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self._access_key_id = access_key_id
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self._session_token = session_token
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self._region = region
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self._model = model
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self._client: Optional[BedrockRuntimeClient] = None
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self._voice_id = voice_id
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self._params = params or Params()
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self._system_instruction = system_instruction
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self._tools = tools
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self._send_transcription_frames = send_transcription_frames
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self._context: Optional[AWSNovaSonicLLMContext] = None
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self._stream: Optional[
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DuplexEventStream[
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InvokeModelWithBidirectionalStreamInput,
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InvokeModelWithBidirectionalStreamOutput,
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InvokeModelWithBidirectionalStreamOperationOutput,
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]
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] = None
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self._receive_task: Optional[asyncio.Task] = None
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self._prompt_name: Optional[str] = None
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self._input_audio_content_name: Optional[str] = None
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self._content_being_received: Optional[CurrentContent] = None
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self._assistant_is_responding = False
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self._ready_to_send_context = False
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self._handling_bot_stopped_speaking = False
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self._triggering_assistant_response = False
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self._assistant_response_trigger_audio: Optional[bytes] = (
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None # Not cleared on _disconnect()
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)
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self._disconnecting = False
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self._connected_time: Optional[float] = None
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self._wants_connection = False
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#
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# standard AIService frame handling
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#
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async def start(self, frame: StartFrame):
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"""Start the service and initiate connection to AWS Nova Sonic.
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Args:
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|
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frame: The start frame triggering service initialization.
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|
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"""
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|
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await super().start(frame)
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|
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self._wants_connection = True
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|
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await self._start_connecting()
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|
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async def stop(self, frame: EndFrame):
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|
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"""Stop the service and close connections.
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Args:
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|
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frame: The end frame triggering service shutdown.
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|
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"""
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|
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await super().stop(frame)
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|
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self._wants_connection = False
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|
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await self._disconnect()
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|
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|
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async def cancel(self, frame: CancelFrame):
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|
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"""Cancel the service and close connections.
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|
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|
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Args:
|
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|
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frame: The cancel frame triggering service cancellation.
|
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|
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"""
|
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|
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await super().cancel(frame)
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|
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self._wants_connection = False
|
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|
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await self._disconnect()
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|
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|
-
#
|
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|
-
# conversation resetting
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|
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#
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|
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|
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295
|
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async def reset_conversation(self):
|
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|
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"""Reset the conversation state while preserving context.
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297
|
-
|
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298
|
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Handles bot stopped speaking event, disconnects from the service,
|
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|
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and reconnects with the preserved context.
|
|
300
|
-
"""
|
|
301
|
-
logger.debug("Resetting conversation")
|
|
302
|
-
await self._handle_bot_stopped_speaking(delay_to_catch_trailing_assistant_text=False)
|
|
303
|
-
|
|
304
|
-
# Carry over previous context through disconnect
|
|
305
|
-
context = self._context
|
|
306
|
-
await self._disconnect()
|
|
307
|
-
self._context = context
|
|
308
|
-
|
|
309
|
-
await self._start_connecting()
|
|
310
|
-
|
|
311
|
-
#
|
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312
|
-
# frame processing
|
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313
|
-
#
|
|
314
|
-
|
|
315
|
-
async def process_frame(self, frame: Frame, direction: FrameDirection):
|
|
316
|
-
"""Process incoming frames and handle service-specific logic.
|
|
317
|
-
|
|
318
|
-
Args:
|
|
319
|
-
frame: The frame to process.
|
|
320
|
-
direction: The direction the frame is traveling.
|
|
321
|
-
"""
|
|
322
|
-
await super().process_frame(frame, direction)
|
|
323
|
-
|
|
324
|
-
if isinstance(frame, OpenAILLMContextFrame):
|
|
325
|
-
await self._handle_context(frame.context)
|
|
326
|
-
elif isinstance(frame, LLMContextFrame):
|
|
327
|
-
raise NotImplementedError(
|
|
328
|
-
"Universal LLMContext is not yet supported for AWS Nova Sonic."
|
|
329
|
-
)
|
|
330
|
-
elif isinstance(frame, InputAudioRawFrame):
|
|
331
|
-
await self._handle_input_audio_frame(frame)
|
|
332
|
-
elif isinstance(frame, BotStoppedSpeakingFrame):
|
|
333
|
-
await self._handle_bot_stopped_speaking(delay_to_catch_trailing_assistant_text=True)
|
|
334
|
-
elif isinstance(frame, AWSNovaSonicFunctionCallResultFrame):
|
|
335
|
-
await self._handle_function_call_result(frame)
|
|
336
|
-
|
|
337
|
-
await self.push_frame(frame, direction)
|
|
338
|
-
|
|
339
|
-
async def _handle_context(self, context: OpenAILLMContext):
|
|
340
|
-
if not self._context:
|
|
341
|
-
# We got our initial context - try to finish connecting
|
|
342
|
-
self._context = AWSNovaSonicLLMContext.upgrade_to_nova_sonic(
|
|
343
|
-
context, self._system_instruction
|
|
344
|
-
)
|
|
345
|
-
await self._finish_connecting_if_context_available()
|
|
346
|
-
|
|
347
|
-
async def _handle_input_audio_frame(self, frame: InputAudioRawFrame):
|
|
348
|
-
# Wait until we're done sending the assistant response trigger audio before sending audio
|
|
349
|
-
# from the user's mic
|
|
350
|
-
if self._triggering_assistant_response:
|
|
351
|
-
return
|
|
352
|
-
|
|
353
|
-
await self._send_user_audio_event(frame.audio)
|
|
354
|
-
|
|
355
|
-
async def _handle_bot_stopped_speaking(self, delay_to_catch_trailing_assistant_text: bool):
|
|
356
|
-
# Protect against back-to-back BotStoppedSpeaking calls, which I've observed
|
|
357
|
-
if self._handling_bot_stopped_speaking:
|
|
358
|
-
return
|
|
359
|
-
self._handling_bot_stopped_speaking = True
|
|
360
|
-
|
|
361
|
-
async def finalize_assistant_response():
|
|
362
|
-
if self._assistant_is_responding:
|
|
363
|
-
# Consider the assistant finished with their response (possibly after a short delay,
|
|
364
|
-
# to allow for any trailing FINAL assistant text block to come in that need to make
|
|
365
|
-
# it into context).
|
|
366
|
-
#
|
|
367
|
-
# TODO: ideally we could base this solely on the LLM output events, but I couldn't
|
|
368
|
-
# figure out a reliable way to determine when we've gotten our last FINAL text block
|
|
369
|
-
# after the LLM is done talking.
|
|
370
|
-
#
|
|
371
|
-
# First I looked at stopReason, but it doesn't seem like the last FINAL text block
|
|
372
|
-
# is reliably marked END_TURN (sometimes the *first* one is, but not the last...
|
|
373
|
-
# bug?)
|
|
374
|
-
#
|
|
375
|
-
# Then I considered schemes where we tally or match up SPECULATIVE text blocks with
|
|
376
|
-
# FINAL text blocks to know how many or which FINAL blocks to expect, but user
|
|
377
|
-
# interruptions throw a wrench in these schemes: depending on the exact timing of
|
|
378
|
-
# the interruption, we should or shouldn't expect some FINAL blocks.
|
|
379
|
-
if delay_to_catch_trailing_assistant_text:
|
|
380
|
-
# This delay length is a balancing act between "catching" trailing assistant
|
|
381
|
-
# text that is quite delayed but not waiting so long that user text comes in
|
|
382
|
-
# first and results in a bit of context message order scrambling.
|
|
383
|
-
await asyncio.sleep(1.25)
|
|
384
|
-
self._assistant_is_responding = False
|
|
385
|
-
await self._report_assistant_response_ended()
|
|
386
|
-
|
|
387
|
-
self._handling_bot_stopped_speaking = False
|
|
388
|
-
|
|
389
|
-
# Finalize the assistant response, either now or after a delay
|
|
390
|
-
if delay_to_catch_trailing_assistant_text:
|
|
391
|
-
self.create_task(finalize_assistant_response())
|
|
392
|
-
else:
|
|
393
|
-
await finalize_assistant_response()
|
|
394
|
-
|
|
395
|
-
async def _handle_function_call_result(self, frame: AWSNovaSonicFunctionCallResultFrame):
|
|
396
|
-
result = frame.result_frame
|
|
397
|
-
await self._send_tool_result(tool_call_id=result.tool_call_id, result=result.result)
|
|
398
|
-
|
|
399
|
-
#
|
|
400
|
-
# LLM communication: lifecycle
|
|
401
|
-
#
|
|
402
|
-
|
|
403
|
-
async def _start_connecting(self):
|
|
404
|
-
try:
|
|
405
|
-
logger.info("Connecting...")
|
|
406
|
-
|
|
407
|
-
if self._client:
|
|
408
|
-
# Here we assume that if we have a client we are connected or connecting
|
|
409
|
-
return
|
|
410
|
-
|
|
411
|
-
# Set IDs for the connection
|
|
412
|
-
self._prompt_name = str(uuid.uuid4())
|
|
413
|
-
self._input_audio_content_name = str(uuid.uuid4())
|
|
414
|
-
|
|
415
|
-
# Create the client
|
|
416
|
-
self._client = self._create_client()
|
|
417
|
-
|
|
418
|
-
# Start the bidirectional stream
|
|
419
|
-
self._stream = await self._client.invoke_model_with_bidirectional_stream(
|
|
420
|
-
InvokeModelWithBidirectionalStreamOperationInput(model_id=self._model)
|
|
421
|
-
)
|
|
422
|
-
|
|
423
|
-
# Send session start event
|
|
424
|
-
await self._send_session_start_event()
|
|
425
|
-
|
|
426
|
-
# Finish connecting
|
|
427
|
-
self._ready_to_send_context = True
|
|
428
|
-
await self._finish_connecting_if_context_available()
|
|
429
|
-
except Exception as e:
|
|
430
|
-
logger.error(f"{self} initialization error: {e}")
|
|
431
|
-
self._disconnect()
|
|
432
|
-
|
|
433
|
-
async def _finish_connecting_if_context_available(self):
|
|
434
|
-
# We can only finish connecting once we've gotten our initial context and we're ready to
|
|
435
|
-
# send it
|
|
436
|
-
if not (self._context and self._ready_to_send_context):
|
|
437
|
-
return
|
|
438
|
-
|
|
439
|
-
logger.info("Finishing connecting (setting up session)...")
|
|
440
|
-
|
|
441
|
-
# Read context
|
|
442
|
-
history = self._context.get_messages_for_initializing_history()
|
|
443
|
-
|
|
444
|
-
# Send prompt start event, specifying tools.
|
|
445
|
-
# Tools from context take priority over self._tools.
|
|
446
|
-
tools = (
|
|
447
|
-
self._context.tools
|
|
448
|
-
if self._context.tools
|
|
449
|
-
else self.get_llm_adapter().from_standard_tools(self._tools)
|
|
450
|
-
)
|
|
451
|
-
logger.debug(f"Using tools: {tools}")
|
|
452
|
-
await self._send_prompt_start_event(tools)
|
|
453
|
-
|
|
454
|
-
# Send system instruction.
|
|
455
|
-
# Instruction from context takes priority over self._system_instruction.
|
|
456
|
-
# (NOTE: this prioritizing occurred automatically behind the scenes: the context was
|
|
457
|
-
# initialized with self._system_instruction and then updated itself from its messages when
|
|
458
|
-
# get_messages_for_initializing_history() was called).
|
|
459
|
-
logger.debug(f"Using system instruction: {history.system_instruction}")
|
|
460
|
-
if history.system_instruction:
|
|
461
|
-
await self._send_text_event(text=history.system_instruction, role=Role.SYSTEM)
|
|
462
|
-
|
|
463
|
-
# Send conversation history
|
|
464
|
-
for message in history.messages:
|
|
465
|
-
await self._send_text_event(text=message.text, role=message.role)
|
|
466
|
-
|
|
467
|
-
# Start audio input
|
|
468
|
-
await self._send_audio_input_start_event()
|
|
469
|
-
|
|
470
|
-
# Start receiving events
|
|
471
|
-
self._receive_task = self.create_task(self._receive_task_handler())
|
|
472
|
-
|
|
473
|
-
# Record finished connecting time (must be done before sending assistant response trigger)
|
|
474
|
-
self._connected_time = time.time()
|
|
475
|
-
|
|
476
|
-
logger.info("Finished connecting")
|
|
477
|
-
|
|
478
|
-
# If we need to, send assistant response trigger (depends on self._connected_time)
|
|
479
|
-
if self._triggering_assistant_response:
|
|
480
|
-
await self._send_assistant_response_trigger()
|
|
481
|
-
|
|
482
|
-
async def _disconnect(self):
|
|
483
|
-
try:
|
|
484
|
-
logger.info("Disconnecting...")
|
|
485
|
-
|
|
486
|
-
# NOTE: see explanation of HACK, below
|
|
487
|
-
self._disconnecting = True
|
|
488
|
-
|
|
489
|
-
# Clean up client
|
|
490
|
-
if self._client:
|
|
491
|
-
await self._send_session_end_events()
|
|
492
|
-
self._client = None
|
|
493
|
-
|
|
494
|
-
# Clean up stream
|
|
495
|
-
if self._stream:
|
|
496
|
-
await self._stream.input_stream.close()
|
|
497
|
-
self._stream = None
|
|
498
|
-
|
|
499
|
-
# NOTE: see explanation of HACK, below
|
|
500
|
-
await asyncio.sleep(1)
|
|
501
|
-
|
|
502
|
-
# Clean up receive task
|
|
503
|
-
# HACK: we should ideally be able to cancel the receive task before stopping the input
|
|
504
|
-
# stream, above (meaning we wouldn't need self._disconnecting). But for some reason if
|
|
505
|
-
# we don't close the input stream and wait a second first, we're getting an error a lot
|
|
506
|
-
# like this one: https://github.com/awslabs/amazon-transcribe-streaming-sdk/issues/61.
|
|
507
|
-
if self._receive_task:
|
|
508
|
-
await self.cancel_task(self._receive_task, timeout=1.0)
|
|
509
|
-
self._receive_task = None
|
|
510
|
-
|
|
511
|
-
# Reset remaining connection-specific state
|
|
512
|
-
self._prompt_name = None
|
|
513
|
-
self._input_audio_content_name = None
|
|
514
|
-
self._content_being_received = None
|
|
515
|
-
self._assistant_is_responding = False
|
|
516
|
-
self._ready_to_send_context = False
|
|
517
|
-
self._handling_bot_stopped_speaking = False
|
|
518
|
-
self._triggering_assistant_response = False
|
|
519
|
-
self._disconnecting = False
|
|
520
|
-
self._connected_time = None
|
|
521
|
-
|
|
522
|
-
logger.info("Finished disconnecting")
|
|
523
|
-
except Exception as e:
|
|
524
|
-
logger.error(f"{self} error disconnecting: {e}")
|
|
525
|
-
|
|
526
|
-
def _create_client(self) -> BedrockRuntimeClient:
|
|
527
|
-
config = Config(
|
|
528
|
-
endpoint_uri=f"https://bedrock-runtime.{self._region}.amazonaws.com",
|
|
529
|
-
region=self._region,
|
|
530
|
-
aws_credentials_identity_resolver=StaticCredentialsResolver(
|
|
531
|
-
credentials=AWSCredentialsIdentity(
|
|
532
|
-
access_key_id=self._access_key_id,
|
|
533
|
-
secret_access_key=self._secret_access_key,
|
|
534
|
-
session_token=self._session_token,
|
|
535
|
-
)
|
|
536
|
-
),
|
|
537
|
-
http_auth_scheme_resolver=HTTPAuthSchemeResolver(),
|
|
538
|
-
http_auth_schemes={"aws.auth#sigv4": SigV4AuthScheme()},
|
|
539
|
-
)
|
|
540
|
-
return BedrockRuntimeClient(config=config)
|
|
541
|
-
|
|
542
|
-
#
|
|
543
|
-
# LLM communication: input events (pipecat -> LLM)
|
|
544
|
-
#
|
|
545
|
-
|
|
546
|
-
async def _send_session_start_event(self):
|
|
547
|
-
session_start = f"""
|
|
548
|
-
{{
|
|
549
|
-
"event": {{
|
|
550
|
-
"sessionStart": {{
|
|
551
|
-
"inferenceConfiguration": {{
|
|
552
|
-
"maxTokens": {self._params.max_tokens},
|
|
553
|
-
"topP": {self._params.top_p},
|
|
554
|
-
"temperature": {self._params.temperature}
|
|
555
|
-
}}
|
|
556
|
-
}}
|
|
557
|
-
}}
|
|
558
|
-
}}
|
|
559
|
-
"""
|
|
560
|
-
await self._send_client_event(session_start)
|
|
561
|
-
|
|
562
|
-
async def _send_prompt_start_event(self, tools: List[Any]):
|
|
563
|
-
if not self._prompt_name:
|
|
564
|
-
return
|
|
565
|
-
|
|
566
|
-
tools_config = (
|
|
567
|
-
f""",
|
|
568
|
-
"toolUseOutputConfiguration": {{
|
|
569
|
-
"mediaType": "application/json"
|
|
570
|
-
}},
|
|
571
|
-
"toolConfiguration": {{
|
|
572
|
-
"tools": {json.dumps(tools)}
|
|
573
|
-
}}
|
|
574
|
-
"""
|
|
575
|
-
if tools
|
|
576
|
-
else ""
|
|
577
|
-
)
|
|
578
|
-
|
|
579
|
-
prompt_start = f'''
|
|
580
|
-
{{
|
|
581
|
-
"event": {{
|
|
582
|
-
"promptStart": {{
|
|
583
|
-
"promptName": "{self._prompt_name}",
|
|
584
|
-
"textOutputConfiguration": {{
|
|
585
|
-
"mediaType": "text/plain"
|
|
586
|
-
}},
|
|
587
|
-
"audioOutputConfiguration": {{
|
|
588
|
-
"mediaType": "audio/lpcm",
|
|
589
|
-
"sampleRateHertz": {self._params.output_sample_rate},
|
|
590
|
-
"sampleSizeBits": {self._params.output_sample_size},
|
|
591
|
-
"channelCount": {self._params.output_channel_count},
|
|
592
|
-
"voiceId": "{self._voice_id}",
|
|
593
|
-
"encoding": "base64",
|
|
594
|
-
"audioType": "SPEECH"
|
|
595
|
-
}}{tools_config}
|
|
596
|
-
}}
|
|
597
|
-
}}
|
|
598
|
-
}}
|
|
599
|
-
'''
|
|
600
|
-
await self._send_client_event(prompt_start)
|
|
601
|
-
|
|
602
|
-
async def _send_audio_input_start_event(self):
|
|
603
|
-
if not self._prompt_name:
|
|
604
|
-
return
|
|
605
|
-
|
|
606
|
-
audio_content_start = f'''
|
|
607
|
-
{{
|
|
608
|
-
"event": {{
|
|
609
|
-
"contentStart": {{
|
|
610
|
-
"promptName": "{self._prompt_name}",
|
|
611
|
-
"contentName": "{self._input_audio_content_name}",
|
|
612
|
-
"type": "AUDIO",
|
|
613
|
-
"interactive": true,
|
|
614
|
-
"role": "USER",
|
|
615
|
-
"audioInputConfiguration": {{
|
|
616
|
-
"mediaType": "audio/lpcm",
|
|
617
|
-
"sampleRateHertz": {self._params.input_sample_rate},
|
|
618
|
-
"sampleSizeBits": {self._params.input_sample_size},
|
|
619
|
-
"channelCount": {self._params.input_channel_count},
|
|
620
|
-
"audioType": "SPEECH",
|
|
621
|
-
"encoding": "base64"
|
|
622
|
-
}}
|
|
623
|
-
}}
|
|
624
|
-
}}
|
|
625
|
-
}}
|
|
626
|
-
'''
|
|
627
|
-
await self._send_client_event(audio_content_start)
|
|
628
|
-
|
|
629
|
-
async def _send_text_event(self, text: str, role: Role):
|
|
630
|
-
if not self._stream or not self._prompt_name or not text:
|
|
631
|
-
return
|
|
632
|
-
|
|
633
|
-
content_name = str(uuid.uuid4())
|
|
634
|
-
|
|
635
|
-
text_content_start = f'''
|
|
636
|
-
{{
|
|
637
|
-
"event": {{
|
|
638
|
-
"contentStart": {{
|
|
639
|
-
"promptName": "{self._prompt_name}",
|
|
640
|
-
"contentName": "{content_name}",
|
|
641
|
-
"type": "TEXT",
|
|
642
|
-
"interactive": true,
|
|
643
|
-
"role": "{role.value}",
|
|
644
|
-
"textInputConfiguration": {{
|
|
645
|
-
"mediaType": "text/plain"
|
|
646
|
-
}}
|
|
647
|
-
}}
|
|
648
|
-
}}
|
|
649
|
-
}}
|
|
650
|
-
'''
|
|
651
|
-
await self._send_client_event(text_content_start)
|
|
652
|
-
|
|
653
|
-
escaped_text = json.dumps(text) # includes quotes
|
|
654
|
-
text_input = f'''
|
|
655
|
-
{{
|
|
656
|
-
"event": {{
|
|
657
|
-
"textInput": {{
|
|
658
|
-
"promptName": "{self._prompt_name}",
|
|
659
|
-
"contentName": "{content_name}",
|
|
660
|
-
"content": {escaped_text}
|
|
661
|
-
}}
|
|
662
|
-
}}
|
|
663
|
-
}}
|
|
664
|
-
'''
|
|
665
|
-
await self._send_client_event(text_input)
|
|
666
|
-
|
|
667
|
-
text_content_end = f'''
|
|
668
|
-
{{
|
|
669
|
-
"event": {{
|
|
670
|
-
"contentEnd": {{
|
|
671
|
-
"promptName": "{self._prompt_name}",
|
|
672
|
-
"contentName": "{content_name}"
|
|
673
|
-
}}
|
|
674
|
-
}}
|
|
675
|
-
}}
|
|
676
|
-
'''
|
|
677
|
-
await self._send_client_event(text_content_end)
|
|
678
|
-
|
|
679
|
-
async def _send_user_audio_event(self, audio: bytes):
|
|
680
|
-
if not self._stream:
|
|
681
|
-
return
|
|
682
|
-
|
|
683
|
-
blob = base64.b64encode(audio)
|
|
684
|
-
audio_event = f'''
|
|
685
|
-
{{
|
|
686
|
-
"event": {{
|
|
687
|
-
"audioInput": {{
|
|
688
|
-
"promptName": "{self._prompt_name}",
|
|
689
|
-
"contentName": "{self._input_audio_content_name}",
|
|
690
|
-
"content": "{blob.decode("utf-8")}"
|
|
691
|
-
}}
|
|
692
|
-
}}
|
|
693
|
-
}}
|
|
694
|
-
'''
|
|
695
|
-
await self._send_client_event(audio_event)
|
|
696
|
-
|
|
697
|
-
async def _send_session_end_events(self):
|
|
698
|
-
if not self._stream or not self._prompt_name:
|
|
699
|
-
return
|
|
700
|
-
|
|
701
|
-
prompt_end = f'''
|
|
702
|
-
{{
|
|
703
|
-
"event": {{
|
|
704
|
-
"promptEnd": {{
|
|
705
|
-
"promptName": "{self._prompt_name}"
|
|
706
|
-
}}
|
|
707
|
-
}}
|
|
708
|
-
}}
|
|
709
|
-
'''
|
|
710
|
-
await self._send_client_event(prompt_end)
|
|
711
|
-
|
|
712
|
-
session_end = """
|
|
713
|
-
{
|
|
714
|
-
"event": {
|
|
715
|
-
"sessionEnd": {}
|
|
716
|
-
}
|
|
717
|
-
}
|
|
718
|
-
"""
|
|
719
|
-
await self._send_client_event(session_end)
|
|
720
|
-
|
|
721
|
-
async def _send_tool_result(self, tool_call_id, result):
|
|
722
|
-
if not self._stream or not self._prompt_name:
|
|
723
|
-
return
|
|
724
|
-
|
|
725
|
-
content_name = str(uuid.uuid4())
|
|
726
|
-
|
|
727
|
-
result_content_start = f'''
|
|
728
|
-
{{
|
|
729
|
-
"event": {{
|
|
730
|
-
"contentStart": {{
|
|
731
|
-
"promptName": "{self._prompt_name}",
|
|
732
|
-
"contentName": "{content_name}",
|
|
733
|
-
"interactive": false,
|
|
734
|
-
"type": "TOOL",
|
|
735
|
-
"role": "TOOL",
|
|
736
|
-
"toolResultInputConfiguration": {{
|
|
737
|
-
"toolUseId": "{tool_call_id}",
|
|
738
|
-
"type": "TEXT",
|
|
739
|
-
"textInputConfiguration": {{
|
|
740
|
-
"mediaType": "text/plain"
|
|
741
|
-
}}
|
|
742
|
-
}}
|
|
743
|
-
}}
|
|
744
|
-
}}
|
|
745
|
-
}}
|
|
746
|
-
'''
|
|
747
|
-
await self._send_client_event(result_content_start)
|
|
748
|
-
|
|
749
|
-
result_content = json.dumps(
|
|
750
|
-
{
|
|
751
|
-
"event": {
|
|
752
|
-
"toolResult": {
|
|
753
|
-
"promptName": self._prompt_name,
|
|
754
|
-
"contentName": content_name,
|
|
755
|
-
"content": json.dumps(result) if isinstance(result, dict) else result,
|
|
756
|
-
}
|
|
757
|
-
}
|
|
758
|
-
}
|
|
759
|
-
)
|
|
760
|
-
await self._send_client_event(result_content)
|
|
761
|
-
|
|
762
|
-
result_content_end = f"""
|
|
763
|
-
{{
|
|
764
|
-
"event": {{
|
|
765
|
-
"contentEnd": {{
|
|
766
|
-
"promptName": "{self._prompt_name}",
|
|
767
|
-
"contentName": "{content_name}"
|
|
768
|
-
}}
|
|
769
|
-
}}
|
|
770
|
-
}}
|
|
771
|
-
"""
|
|
772
|
-
await self._send_client_event(result_content_end)
|
|
773
|
-
|
|
774
|
-
async def _send_client_event(self, event_json: str):
|
|
775
|
-
if not self._stream: # should never happen
|
|
776
|
-
return
|
|
777
|
-
|
|
778
|
-
event = InvokeModelWithBidirectionalStreamInputChunk(
|
|
779
|
-
value=BidirectionalInputPayloadPart(bytes_=event_json.encode("utf-8"))
|
|
780
|
-
)
|
|
781
|
-
await self._stream.input_stream.send(event)
|
|
782
|
-
|
|
783
|
-
#
|
|
784
|
-
# LLM communication: output events (LLM -> pipecat)
|
|
785
|
-
#
|
|
786
|
-
|
|
787
|
-
# Receive events for the session.
|
|
788
|
-
# A few different kinds of content can be delivered:
|
|
789
|
-
# - Transcription of user audio
|
|
790
|
-
# - Tool use
|
|
791
|
-
# - Text preview of planned response speech before audio delivered
|
|
792
|
-
# - User interruption notification
|
|
793
|
-
# - Text of response speech that whose audio was actually delivered
|
|
794
|
-
# - Audio of response speech
|
|
795
|
-
# Each piece of content is wrapped by "contentStart" and "contentEnd" events. The content is
|
|
796
|
-
# delivered sequentially: one piece of content will end before another starts.
|
|
797
|
-
# The overall completion is wrapped by "completionStart" and "completionEnd" events.
|
|
798
|
-
async def _receive_task_handler(self):
|
|
799
|
-
try:
|
|
800
|
-
while self._stream and not self._disconnecting:
|
|
801
|
-
output = await self._stream.await_output()
|
|
802
|
-
result = await output[1].receive()
|
|
803
|
-
|
|
804
|
-
if result.value and result.value.bytes_:
|
|
805
|
-
response_data = result.value.bytes_.decode("utf-8")
|
|
806
|
-
json_data = json.loads(response_data)
|
|
807
|
-
|
|
808
|
-
if "event" in json_data:
|
|
809
|
-
event_json = json_data["event"]
|
|
810
|
-
if "completionStart" in event_json:
|
|
811
|
-
# Handle the LLM completion starting
|
|
812
|
-
await self._handle_completion_start_event(event_json)
|
|
813
|
-
elif "contentStart" in event_json:
|
|
814
|
-
# Handle a piece of content starting
|
|
815
|
-
await self._handle_content_start_event(event_json)
|
|
816
|
-
elif "textOutput" in event_json:
|
|
817
|
-
# Handle text output content
|
|
818
|
-
await self._handle_text_output_event(event_json)
|
|
819
|
-
elif "audioOutput" in event_json:
|
|
820
|
-
# Handle audio output content
|
|
821
|
-
await self._handle_audio_output_event(event_json)
|
|
822
|
-
elif "toolUse" in event_json:
|
|
823
|
-
# Handle tool use
|
|
824
|
-
await self._handle_tool_use_event(event_json)
|
|
825
|
-
elif "contentEnd" in event_json:
|
|
826
|
-
# Handle a piece of content ending
|
|
827
|
-
await self._handle_content_end_event(event_json)
|
|
828
|
-
elif "completionEnd" in event_json:
|
|
829
|
-
# Handle the LLM completion ending
|
|
830
|
-
await self._handle_completion_end_event(event_json)
|
|
831
|
-
except Exception as e:
|
|
832
|
-
logger.error(f"{self} error processing responses: {e}")
|
|
833
|
-
if self._wants_connection:
|
|
834
|
-
await self.reset_conversation()
|
|
835
|
-
|
|
836
|
-
async def _handle_completion_start_event(self, event_json):
|
|
837
|
-
pass
|
|
838
|
-
|
|
839
|
-
async def _handle_content_start_event(self, event_json):
|
|
840
|
-
content_start = event_json["contentStart"]
|
|
841
|
-
type = content_start["type"]
|
|
842
|
-
role = content_start["role"]
|
|
843
|
-
generation_stage = None
|
|
844
|
-
if "additionalModelFields" in content_start:
|
|
845
|
-
additional_model_fields = json.loads(content_start["additionalModelFields"])
|
|
846
|
-
generation_stage = additional_model_fields.get("generationStage")
|
|
847
|
-
|
|
848
|
-
# Bookkeeping: track current content being received
|
|
849
|
-
content = CurrentContent(
|
|
850
|
-
type=ContentType(type),
|
|
851
|
-
role=Role(role),
|
|
852
|
-
text_stage=TextStage(generation_stage) if generation_stage else None,
|
|
853
|
-
text_content=None,
|
|
854
|
-
)
|
|
855
|
-
self._content_being_received = content
|
|
856
|
-
|
|
857
|
-
if content.role == Role.ASSISTANT:
|
|
858
|
-
if content.type == ContentType.AUDIO:
|
|
859
|
-
# Note that an assistant response can comprise of multiple audio blocks
|
|
860
|
-
if not self._assistant_is_responding:
|
|
861
|
-
# The assistant has started responding.
|
|
862
|
-
self._assistant_is_responding = True
|
|
863
|
-
await self._report_user_transcription_ended() # Consider user turn over
|
|
864
|
-
await self._report_assistant_response_started()
|
|
865
|
-
|
|
866
|
-
async def _handle_text_output_event(self, event_json):
|
|
867
|
-
if not self._content_being_received: # should never happen
|
|
868
|
-
return
|
|
869
|
-
content = self._content_being_received
|
|
870
|
-
|
|
871
|
-
text_content = event_json["textOutput"]["content"]
|
|
872
|
-
|
|
873
|
-
# Bookkeeping: augment the current content being received with text
|
|
874
|
-
# Assumption: only one text content per content block
|
|
875
|
-
content.text_content = text_content
|
|
876
|
-
|
|
877
|
-
async def _handle_audio_output_event(self, event_json):
|
|
878
|
-
if not self._content_being_received: # should never happen
|
|
879
|
-
return
|
|
880
|
-
|
|
881
|
-
# Get audio
|
|
882
|
-
audio_content = event_json["audioOutput"]["content"]
|
|
883
|
-
|
|
884
|
-
# Push audio frame
|
|
885
|
-
audio = base64.b64decode(audio_content)
|
|
886
|
-
frame = TTSAudioRawFrame(
|
|
887
|
-
audio=audio,
|
|
888
|
-
sample_rate=self._params.output_sample_rate,
|
|
889
|
-
num_channels=self._params.output_channel_count,
|
|
890
|
-
)
|
|
891
|
-
await self.push_frame(frame)
|
|
892
|
-
|
|
893
|
-
async def _handle_tool_use_event(self, event_json):
|
|
894
|
-
if not self._content_being_received or not self._context: # should never happen
|
|
895
|
-
return
|
|
896
|
-
|
|
897
|
-
# Consider user turn over
|
|
898
|
-
await self._report_user_transcription_ended()
|
|
899
|
-
|
|
900
|
-
# Get tool use details
|
|
901
|
-
tool_use = event_json["toolUse"]
|
|
902
|
-
function_name = tool_use["toolName"]
|
|
903
|
-
tool_call_id = tool_use["toolUseId"]
|
|
904
|
-
arguments = json.loads(tool_use["content"])
|
|
905
|
-
|
|
906
|
-
# Call tool function
|
|
907
|
-
if self.has_function(function_name):
|
|
908
|
-
if function_name in self._functions.keys() or None in self._functions.keys():
|
|
909
|
-
function_calls_llm = [
|
|
910
|
-
FunctionCallFromLLM(
|
|
911
|
-
context=self._context,
|
|
912
|
-
tool_call_id=tool_call_id,
|
|
913
|
-
function_name=function_name,
|
|
914
|
-
arguments=arguments,
|
|
915
|
-
)
|
|
916
|
-
]
|
|
917
|
-
|
|
918
|
-
await self.run_function_calls(function_calls_llm)
|
|
919
|
-
else:
|
|
920
|
-
raise AWSNovaSonicUnhandledFunctionException(
|
|
921
|
-
f"The LLM tried to call a function named '{function_name}', but there isn't a callback registered for that function."
|
|
922
|
-
)
|
|
923
|
-
|
|
924
|
-
async def _handle_content_end_event(self, event_json):
|
|
925
|
-
if not self._content_being_received: # should never happen
|
|
926
|
-
return
|
|
927
|
-
content = self._content_being_received
|
|
928
|
-
|
|
929
|
-
content_end = event_json["contentEnd"]
|
|
930
|
-
stop_reason = content_end["stopReason"]
|
|
931
|
-
|
|
932
|
-
# Bookkeeping: clear current content being received
|
|
933
|
-
self._content_being_received = None
|
|
934
|
-
|
|
935
|
-
if content.role == Role.ASSISTANT:
|
|
936
|
-
if content.type == ContentType.TEXT:
|
|
937
|
-
# Ignore non-final text, and the "interrupted" message (which isn't meaningful text)
|
|
938
|
-
if content.text_stage == TextStage.FINAL and stop_reason != "INTERRUPTED":
|
|
939
|
-
if self._assistant_is_responding:
|
|
940
|
-
# Text added to the ongoing assistant response
|
|
941
|
-
await self._report_assistant_response_text_added(content.text_content)
|
|
942
|
-
elif content.role == Role.USER:
|
|
943
|
-
if content.type == ContentType.TEXT:
|
|
944
|
-
if content.text_stage == TextStage.FINAL:
|
|
945
|
-
# User transcription text added
|
|
946
|
-
await self._report_user_transcription_text_added(content.text_content)
|
|
947
|
-
|
|
948
|
-
async def _handle_completion_end_event(self, event_json):
|
|
949
|
-
pass
|
|
950
|
-
|
|
951
|
-
#
|
|
952
|
-
# assistant response reporting
|
|
953
|
-
#
|
|
954
|
-
# 1. Started
|
|
955
|
-
# 2. Text added
|
|
956
|
-
# 3. Ended
|
|
957
|
-
#
|
|
958
|
-
|
|
959
|
-
async def _report_assistant_response_started(self):
|
|
960
|
-
logger.debug("Assistant response started")
|
|
961
|
-
|
|
962
|
-
# Report that the assistant has started their response.
|
|
963
|
-
await self.push_frame(LLMFullResponseStartFrame())
|
|
964
|
-
|
|
965
|
-
# Report that equivalent of TTS (this is a speech-to-speech model) started
|
|
966
|
-
await self.push_frame(TTSStartedFrame())
|
|
967
|
-
|
|
968
|
-
async def _report_assistant_response_text_added(self, text):
|
|
969
|
-
if not self._context: # should never happen
|
|
970
|
-
return
|
|
971
|
-
|
|
972
|
-
logger.debug(f"Assistant response text added: {text}")
|
|
973
|
-
|
|
974
|
-
# Report some text added to the ongoing assistant response
|
|
975
|
-
await self.push_frame(LLMTextFrame(text))
|
|
976
|
-
|
|
977
|
-
# Report some text added to the *equivalent* of TTS (this is a speech-to-speech model)
|
|
978
|
-
await self.push_frame(TTSTextFrame(text))
|
|
979
|
-
|
|
980
|
-
# TODO: this is a (hopefully temporary) HACK. Here we directly manipulate the context rather
|
|
981
|
-
# than relying on the frames pushed to the assistant context aggregator. The pattern of
|
|
982
|
-
# receiving full-sentence text after the assistant has spoken does not easily fit with the
|
|
983
|
-
# Pipecat expectation of chunks of text streaming in while the assistant is speaking.
|
|
984
|
-
# Interruption handling was especially challenging. Rather than spend days trying to fit a
|
|
985
|
-
# square peg in a round hole, I decided on this hack for the time being. We can most cleanly
|
|
986
|
-
# abandon this hack if/when AWS Nova Sonic implements streaming smaller text chunks
|
|
987
|
-
# interspersed with audio. Note that when we move away from this hack, we need to make sure
|
|
988
|
-
# that on an interruption we avoid sending LLMFullResponseEndFrame, which gets the
|
|
989
|
-
# LLMAssistantContextAggregator into a bad state.
|
|
990
|
-
self._context.buffer_assistant_text(text)
|
|
991
|
-
|
|
992
|
-
async def _report_assistant_response_ended(self):
|
|
993
|
-
if not self._context: # should never happen
|
|
994
|
-
return
|
|
995
|
-
|
|
996
|
-
logger.debug("Assistant response ended")
|
|
997
|
-
|
|
998
|
-
# Report that the assistant has finished their response.
|
|
999
|
-
await self.push_frame(LLMFullResponseEndFrame())
|
|
1000
|
-
|
|
1001
|
-
# Report that equivalent of TTS (this is a speech-to-speech model) stopped.
|
|
1002
|
-
await self.push_frame(TTSStoppedFrame())
|
|
1003
|
-
|
|
1004
|
-
# For an explanation of this hack, see _report_assistant_response_text_added.
|
|
1005
|
-
self._context.flush_aggregated_assistant_text()
|
|
1006
|
-
|
|
1007
|
-
#
|
|
1008
|
-
# user transcription reporting
|
|
1009
|
-
#
|
|
1010
|
-
# 1. Text added
|
|
1011
|
-
# 2. Ended
|
|
1012
|
-
#
|
|
1013
|
-
# Note: "started" does not need to be reported
|
|
1014
|
-
#
|
|
1015
|
-
|
|
1016
|
-
async def _report_user_transcription_text_added(self, text):
|
|
1017
|
-
if not self._context: # should never happen
|
|
1018
|
-
return
|
|
1019
|
-
|
|
1020
|
-
logger.debug(f"User transcription text added: {text}")
|
|
1021
|
-
|
|
1022
|
-
# Manually add new user transcription text to context.
|
|
1023
|
-
# We can't rely on the user context aggregator to do this since it's upstream from the LLM.
|
|
1024
|
-
self._context.buffer_user_text(text)
|
|
1025
|
-
|
|
1026
|
-
# Report that some new user transcription text is available.
|
|
1027
|
-
if self._send_transcription_frames:
|
|
1028
|
-
await self.push_frame(
|
|
1029
|
-
InterimTranscriptionFrame(text=text, user_id="", timestamp=time_now_iso8601())
|
|
1030
|
-
)
|
|
1031
|
-
|
|
1032
|
-
async def _report_user_transcription_ended(self):
|
|
1033
|
-
if not self._context: # should never happen
|
|
1034
|
-
return
|
|
1035
|
-
|
|
1036
|
-
# Manually add user transcription to context (if any has been buffered).
|
|
1037
|
-
# We can't rely on the user context aggregator to do this since it's upstream from the LLM.
|
|
1038
|
-
transcription = self._context.flush_aggregated_user_text()
|
|
1039
|
-
|
|
1040
|
-
if not transcription:
|
|
1041
|
-
return
|
|
1042
|
-
|
|
1043
|
-
logger.debug(f"User transcription ended")
|
|
1044
|
-
|
|
1045
|
-
if self._send_transcription_frames:
|
|
1046
|
-
await self.push_frame(
|
|
1047
|
-
TranscriptionFrame(text=transcription, user_id="", timestamp=time_now_iso8601())
|
|
1048
|
-
)
|
|
1049
|
-
|
|
1050
|
-
#
|
|
1051
|
-
# context
|
|
1052
|
-
#
|
|
1053
|
-
|
|
1054
|
-
def create_context_aggregator(
|
|
1055
|
-
self,
|
|
1056
|
-
context: OpenAILLMContext,
|
|
1057
|
-
*,
|
|
1058
|
-
user_params: LLMUserAggregatorParams = LLMUserAggregatorParams(),
|
|
1059
|
-
assistant_params: LLMAssistantAggregatorParams = LLMAssistantAggregatorParams(),
|
|
1060
|
-
) -> AWSNovaSonicContextAggregatorPair:
|
|
1061
|
-
"""Create context aggregator pair for managing conversation context.
|
|
1062
|
-
|
|
1063
|
-
Args:
|
|
1064
|
-
context: The OpenAI LLM context to upgrade.
|
|
1065
|
-
user_params: Parameters for the user context aggregator.
|
|
1066
|
-
assistant_params: Parameters for the assistant context aggregator.
|
|
1067
|
-
|
|
1068
|
-
Returns:
|
|
1069
|
-
A pair of user and assistant context aggregators.
|
|
1070
|
-
"""
|
|
1071
|
-
context.set_llm_adapter(self.get_llm_adapter())
|
|
1072
|
-
|
|
1073
|
-
user = AWSNovaSonicUserContextAggregator(context=context, params=user_params)
|
|
1074
|
-
assistant = AWSNovaSonicAssistantContextAggregator(context=context, params=assistant_params)
|
|
1075
|
-
|
|
1076
|
-
return AWSNovaSonicContextAggregatorPair(user, assistant)
|
|
1077
|
-
|
|
1078
|
-
#
|
|
1079
|
-
# assistant response trigger (HACK)
|
|
1080
|
-
#
|
|
1081
|
-
|
|
1082
|
-
# Class variable
|
|
1083
|
-
AWAIT_TRIGGER_ASSISTANT_RESPONSE_INSTRUCTION = (
|
|
1084
|
-
"Start speaking when you hear the user say 'ready', but don't consider that 'ready' to be "
|
|
1085
|
-
"a meaningful part of the conversation other than as a trigger for you to start speaking."
|
|
17
|
+
with warnings.catch_warnings():
|
|
18
|
+
warnings.simplefilter("always")
|
|
19
|
+
warnings.warn(
|
|
20
|
+
"Types in pipecat.services.aws_nova_sonic.aws are deprecated. "
|
|
21
|
+
"Please use the equivalent types from "
|
|
22
|
+
"pipecat.services.aws.nova_sonic.llm instead.",
|
|
23
|
+
DeprecationWarning,
|
|
24
|
+
stacklevel=2,
|
|
1086
25
|
)
|
|
1087
|
-
|
|
1088
|
-
async def trigger_assistant_response(self):
|
|
1089
|
-
"""Trigger an assistant response by sending audio cue.
|
|
1090
|
-
|
|
1091
|
-
Sends a pre-recorded "ready" audio trigger to prompt the assistant
|
|
1092
|
-
to start speaking. This is useful for controlling conversation flow.
|
|
1093
|
-
|
|
1094
|
-
Returns:
|
|
1095
|
-
False if already triggering a response, True otherwise.
|
|
1096
|
-
"""
|
|
1097
|
-
if self._triggering_assistant_response:
|
|
1098
|
-
return False
|
|
1099
|
-
|
|
1100
|
-
self._triggering_assistant_response = True
|
|
1101
|
-
|
|
1102
|
-
# Read audio bytes, if we don't already have them cached
|
|
1103
|
-
if not self._assistant_response_trigger_audio:
|
|
1104
|
-
file_path = files("pipecat.services.aws_nova_sonic").joinpath("ready.wav")
|
|
1105
|
-
with wave.open(file_path.open("rb"), "rb") as wav_file:
|
|
1106
|
-
self._assistant_response_trigger_audio = wav_file.readframes(wav_file.getnframes())
|
|
1107
|
-
|
|
1108
|
-
# Send the trigger audio, if we're fully connected and set up
|
|
1109
|
-
if self._connected_time is not None:
|
|
1110
|
-
await self._send_assistant_response_trigger()
|
|
1111
|
-
|
|
1112
|
-
async def _send_assistant_response_trigger(self):
|
|
1113
|
-
if (
|
|
1114
|
-
not self._assistant_response_trigger_audio or self._connected_time is None
|
|
1115
|
-
): # should never happen
|
|
1116
|
-
return
|
|
1117
|
-
|
|
1118
|
-
try:
|
|
1119
|
-
logger.debug("Sending assistant response trigger...")
|
|
1120
|
-
|
|
1121
|
-
chunk_duration = 0.02 # what we might get from InputAudioRawFrame
|
|
1122
|
-
chunk_size = int(
|
|
1123
|
-
chunk_duration
|
|
1124
|
-
* self._params.input_sample_rate
|
|
1125
|
-
* self._params.input_channel_count
|
|
1126
|
-
* (self._params.input_sample_size / 8)
|
|
1127
|
-
) # e.g. 0.02 seconds of 16-bit (2-byte) PCM mono audio at 16kHz is 640 bytes
|
|
1128
|
-
|
|
1129
|
-
# Lead with a bit of blank audio, if needed.
|
|
1130
|
-
# It seems like the LLM can't quite "hear" the first little bit of audio sent on a
|
|
1131
|
-
# connection.
|
|
1132
|
-
current_time = time.time()
|
|
1133
|
-
max_blank_audio_duration = 0.5
|
|
1134
|
-
blank_audio_duration = (
|
|
1135
|
-
max_blank_audio_duration - (current_time - self._connected_time)
|
|
1136
|
-
if self._connected_time is not None
|
|
1137
|
-
and (current_time - self._connected_time) < max_blank_audio_duration
|
|
1138
|
-
else None
|
|
1139
|
-
)
|
|
1140
|
-
if blank_audio_duration:
|
|
1141
|
-
logger.debug(
|
|
1142
|
-
f"Leading assistant response trigger with {blank_audio_duration}s of blank audio"
|
|
1143
|
-
)
|
|
1144
|
-
blank_audio_chunk = b"\x00" * chunk_size
|
|
1145
|
-
num_chunks = int(blank_audio_duration / chunk_duration)
|
|
1146
|
-
for _ in range(num_chunks):
|
|
1147
|
-
await self._send_user_audio_event(blank_audio_chunk)
|
|
1148
|
-
await asyncio.sleep(chunk_duration)
|
|
1149
|
-
|
|
1150
|
-
# Send trigger audio
|
|
1151
|
-
# NOTE: this audio *will* be transcribed and eventually make it into the context. That's OK:
|
|
1152
|
-
# if we ever need to seed this service again with context it would make sense to include it
|
|
1153
|
-
# since the instruction (i.e. the "wait for the trigger" instruction) will be part of the
|
|
1154
|
-
# context as well.
|
|
1155
|
-
audio_chunks = [
|
|
1156
|
-
self._assistant_response_trigger_audio[i : i + chunk_size]
|
|
1157
|
-
for i in range(0, len(self._assistant_response_trigger_audio), chunk_size)
|
|
1158
|
-
]
|
|
1159
|
-
for chunk in audio_chunks:
|
|
1160
|
-
await self._send_user_audio_event(chunk)
|
|
1161
|
-
await asyncio.sleep(chunk_duration)
|
|
1162
|
-
finally:
|
|
1163
|
-
# We need to clean up in case sending the trigger was cancelled, e.g. in the case of a user interruption.
|
|
1164
|
-
# (An asyncio.CancelledError would be raised in that case.)
|
|
1165
|
-
self._triggering_assistant_response = False
|