Smart Turn V3 support
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102
src/pipecat/audio/turn/smart_turn/local_smart_turn_v3.py
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102
src/pipecat/audio/turn/smart_turn/local_smart_turn_v3.py
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#
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# Copyright (c) 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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"""Local PyTorch turn analyzer for on-device ML inference using the smart-turn-v3 model.
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This module provides a smart turn analyzer that uses an ONNX model for
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local end-of-turn detection without requiring network connectivity.
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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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from transformers import WhisperFeatureExtractor
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import onnxruntime as ort
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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 LocalSmartTurnAnalyzerV3, you need to `pip install pipecat-ai[local-smart-turn-v3]`."
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)
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raise Exception(f"Missing module: {e}")
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class LocalSmartTurnAnalyzerV3(BaseSmartTurn):
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"""Local turn analyzer using the smart-turn-v2 PyTorch model.
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Provides end-of-turn detection using locally-stored PyTorch models,
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enabling offline operation without network dependencies. Uses
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Wav2Vec2 architecture for audio sequence classification.
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"""
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def __init__(self, *, smart_turn_model_path: str, **kwargs):
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"""Initialize the local PyTorch smart-turn-v3 analyzer.
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Args:
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smart_turn_model_path: Path to the ONNX model file.
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**kwargs: Additional arguments passed to BaseSmartTurn.
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"""
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super().__init__(**kwargs)
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if not smart_turn_model_path:
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raise ValueError("smart_turn_model_path must be provided")
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logger.debug("Loading Local Smart Turn v3 model...")
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self._feature_extractor = WhisperFeatureExtractor(chunk_length=8)
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self._session = ort.InferenceSession(smart_turn_model_path)
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logger.debug("Loaded Local Smart Turn v3")
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async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
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"""Predict end-of-turn using local ONNX model."""
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def truncate_audio_to_last_n_seconds(audio_array, n_seconds=8, sample_rate=16000):
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"""Truncate audio to last n seconds or pad with zeros to meet n seconds."""
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max_samples = n_seconds * sample_rate
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if len(audio_array) > max_samples:
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return audio_array[-max_samples:]
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elif len(audio_array) < max_samples:
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# Pad with zeros at the beginning
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padding = max_samples - len(audio_array)
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return np.pad(audio_array, (padding, 0), mode='constant', constant_values=0)
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return audio_array
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# Truncate to 8 seconds (keeping the end) or pad to 8 seconds
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audio_array = truncate_audio_to_last_n_seconds(audio_array, n_seconds=8)
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# Process audio using Whisper's feature extractor
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inputs = self._feature_extractor(
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audio_array,
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sampling_rate=16000,
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return_tensors="pt",
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padding="max_length",
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max_length=8 * 16000,
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truncation=True,
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do_normalize=True,
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)
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# Convert to numpy and ensure correct shape for ONNX
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input_features = inputs.input_features.squeeze(0).numpy().astype(np.float32)
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input_features = np.expand_dims(input_features, axis=0) # Add batch dimension
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# Run ONNX inference
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outputs = self._session.run(None, {"input_features": input_features})
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# Extract probability (ONNX model returns sigmoid probabilities)
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probability = outputs[0][0].item()
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# Make prediction (1 for Complete, 0 for Incomplete)
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prediction = 1 if probability > 0.5 else 0
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return {
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"prediction": prediction,
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"probability": probability,
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}
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