Merge pull request #1717 from pipecat-ai/local_smart_turn_torch
Local smart turn torch
This commit is contained in:
@@ -9,6 +9,9 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
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### Added
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- Added support for cross-platform local smart turn detection. You can use
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`LocalSmartTurnAnalyzer` for on-device inference using Torch.
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- `BaseOutputTransport` now allows multiple destinations if the transport
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implementation supports it (e.g. Daily's custom tracks). With multiple
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destinations it is possible to send different audio or video tracks with a
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128
examples/foundational/38b-smart-turn-local.py
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128
examples/foundational/38b-smart-turn-local.py
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#
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# Copyright (c) 2024–2025, Daily
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#
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# SPDX-License-Identifier: BSD 2-Clause License
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#
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import argparse
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import os
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from dotenv import load_dotenv
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from loguru import logger
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from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
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from pipecat.audio.turn.smart_turn.local_smart_turn import LocalSmartTurnAnalyzer
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from pipecat.audio.vad.silero import SileroVADAnalyzer
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from pipecat.audio.vad.vad_analyzer import VADParams
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from pipecat.pipeline.pipeline import Pipeline
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from pipecat.pipeline.runner import PipelineRunner
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from pipecat.pipeline.task import PipelineParams, PipelineTask
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from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
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from pipecat.services.cartesia.tts import CartesiaTTSService
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from pipecat.services.deepgram.stt import DeepgramSTTService
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from pipecat.services.openai.llm import OpenAILLMService
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from pipecat.transports.base_transport import TransportParams
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from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
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from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
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load_dotenv(override=True)
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async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
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logger.info(f"Starting bot")
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# To use this locally, set the environment variable LOCAL_SMART_TURN_MODEL_PATH
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# to the path where the smart-turn repo is cloned.
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#
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# Example setup:
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#
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# # Git LFS (Large File Storage)
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# brew install git-lfs
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# # Hugging Face uses LFS to store large model files, including .mlpackage
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# git lfs install
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# # Clone the repo with the smart_turn_classifier.mlpackage
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# git clone https://huggingface.co/pipecat-ai/smart-turn
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#
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# Then set the env variable:
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# export LOCAL_SMART_TURN_MODEL_PATH=./smart-turn
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# or add it to your .env file
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smart_turn_model_path = os.getenv("LOCAL_SMART_TURN_MODEL_PATH")
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transport = SmallWebRTCTransport(
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webrtc_connection=webrtc_connection,
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params=TransportParams(
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audio_in_enabled=True,
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audio_out_enabled=True,
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vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.2)),
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turn_analyzer=LocalSmartTurnAnalyzer(
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smart_turn_model_path=smart_turn_model_path, params=SmartTurnParams()
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),
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),
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)
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stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
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tts = CartesiaTTSService(
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api_key=os.getenv("CARTESIA_API_KEY"),
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voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
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)
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llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
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messages = [
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{
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"role": "system",
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"content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way.",
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},
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]
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context = OpenAILLMContext(messages)
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context_aggregator = llm.create_context_aggregator(context)
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pipeline = Pipeline(
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[
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transport.input(), # Transport user input
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stt,
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context_aggregator.user(), # User responses
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llm, # LLM
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tts, # TTS
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transport.output(), # Transport bot output
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context_aggregator.assistant(), # Assistant spoken responses
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]
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)
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task = PipelineTask(
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pipeline,
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params=PipelineParams(
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allow_interruptions=True,
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enable_metrics=True,
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enable_usage_metrics=True,
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report_only_initial_ttfb=True,
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),
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)
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@transport.event_handler("on_client_connected")
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async def on_client_connected(transport, client):
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logger.info(f"Client connected")
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# Kick off the conversation.
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messages.append({"role": "system", "content": "Please introduce yourself to the user."})
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await task.queue_frames([context_aggregator.user().get_context_frame()])
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@transport.event_handler("on_client_disconnected")
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async def on_client_disconnected(transport, client):
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logger.info(f"Client disconnected")
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@transport.event_handler("on_client_closed")
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async def on_client_closed(transport, client):
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logger.info(f"Client closed connection")
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await task.cancel()
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runner = PipelineRunner(handle_sigint=False)
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await runner.run(task)
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if __name__ == "__main__":
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from run import main
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main()
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73
src/pipecat/audio/turn/smart_turn/local_smart_turn.py
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73
src/pipecat/audio/turn/smart_turn/local_smart_turn.py
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#
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# Copyright (c) 2024–2025, Daily
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#
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# SPDX-License-Identifier: BSD 2-Clause License
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#
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from typing import Any, Dict
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import numpy as np
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from loguru import logger
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from pipecat.audio.turn.smart_turn.base_smart_turn import BaseSmartTurn
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try:
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import torch
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from transformers import AutoFeatureExtractor, Wav2Vec2BertForSequenceClassification
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except ModuleNotFoundError as e:
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logger.error(f"Exception: {e}")
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logger.error(
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"In order to use the LocalSmartTurnAnalyzer, you need to `pip install pipecat-ai[local-smart-turn]`."
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)
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raise Exception(f"Missing module: {e}")
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class LocalSmartTurnAnalyzer(BaseSmartTurn):
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def __init__(self, *, smart_turn_model_path: str, **kwargs):
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super().__init__(**kwargs)
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if not smart_turn_model_path:
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# Define the path to the pretrained model on Hugging Face
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smart_turn_model_path = "pipecat-ai/smart-turn"
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logger.debug("Loading Local Smart Turn model...")
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# Load the pretrained model for sequence classification
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self._turn_model = Wav2Vec2BertForSequenceClassification.from_pretrained(
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smart_turn_model_path
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)
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# Load the corresponding feature extractor for preprocessing audio
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self._turn_processor = AutoFeatureExtractor.from_pretrained(smart_turn_model_path)
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# Set device to GPU if available, else CPU
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self._device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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# Move model to selected device and set it to evaluation mode
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self._turn_model = self._turn_model.to(self._device)
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self._turn_model.eval()
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logger.debug("Loaded Local Smart Turn")
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async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
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inputs = self._turn_processor(
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audio_array,
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sampling_rate=16000,
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padding="max_length",
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truncation=True,
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max_length=800, # Maximum length as specified in training
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return_attention_mask=True,
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return_tensors="pt",
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)
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# Move input tensors to the same device as the model
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inputs = {k: v.to(self._device) for k, v in inputs.items()}
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# Disable gradient calculation for inference
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with torch.no_grad():
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outputs = self._turn_model(**inputs)
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logits = outputs.logits
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probabilities = torch.nn.functional.softmax(logits, dim=1)
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completion_prob = probabilities[0, 1].item() # Probability of class 1 (Complete)
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prediction = 1 if completion_prob > 0.5 else 0
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return {
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"prediction": prediction,
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"probability": completion_prob,
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}
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