Merge pull request #1717 from pipecat-ai/local_smart_turn_torch

Local smart turn torch
This commit is contained in:
Filipi da Silva Fuchter
2025-05-02 15:53:30 -03:00
committed by GitHub
3 changed files with 204 additions and 0 deletions

View File

@@ -9,6 +9,9 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
### Added
- Added support for cross-platform local smart turn detection. You can use
`LocalSmartTurnAnalyzer` for on-device inference using Torch.
- `BaseOutputTransport` now allows multiple destinations if the transport
implementation supports it (e.g. Daily's custom tracks). With multiple
destinations it is possible to send different audio or video tracks with a

View File

@@ -0,0 +1,128 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import argparse
import os
from dotenv import load_dotenv
from loguru import logger
from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
from pipecat.audio.turn.smart_turn.local_smart_turn import LocalSmartTurnAnalyzer
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# To use this locally, set the environment variable LOCAL_SMART_TURN_MODEL_PATH
# to the path where the smart-turn repo is cloned.
#
# Example setup:
#
# # Git LFS (Large File Storage)
# brew install git-lfs
# # Hugging Face uses LFS to store large model files, including .mlpackage
# git lfs install
# # Clone the repo with the smart_turn_classifier.mlpackage
# git clone https://huggingface.co/pipecat-ai/smart-turn
#
# Then set the env variable:
# export LOCAL_SMART_TURN_MODEL_PATH=./smart-turn
# or add it to your .env file
smart_turn_model_path = os.getenv("LOCAL_SMART_TURN_MODEL_PATH")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.2)),
turn_analyzer=LocalSmartTurnAnalyzer(
smart_turn_model_path=smart_turn_model_path, params=SmartTurnParams()
),
),
)
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
messages = [
{
"role": "system",
"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.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(), # Transport user input
stt,
context_aggregator.user(), # User responses
llm, # LLM
tts, # TTS
transport.output(), # Transport bot output
context_aggregator.assistant(), # Assistant spoken responses
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
logger.info(f"Client connected")
# Kick off the conversation.
messages.append({"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([context_aggregator.user().get_context_frame()])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
await runner.run(task)
if __name__ == "__main__":
from run import main
main()

View File

@@ -0,0 +1,73 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
from typing import Any, Dict
import numpy as np
from loguru import logger
from pipecat.audio.turn.smart_turn.base_smart_turn import BaseSmartTurn
try:
import torch
from transformers import AutoFeatureExtractor, Wav2Vec2BertForSequenceClassification
except ModuleNotFoundError as e:
logger.error(f"Exception: {e}")
logger.error(
"In order to use the LocalSmartTurnAnalyzer, you need to `pip install pipecat-ai[local-smart-turn]`."
)
raise Exception(f"Missing module: {e}")
class LocalSmartTurnAnalyzer(BaseSmartTurn):
def __init__(self, *, smart_turn_model_path: str, **kwargs):
super().__init__(**kwargs)
if not smart_turn_model_path:
# Define the path to the pretrained model on Hugging Face
smart_turn_model_path = "pipecat-ai/smart-turn"
logger.debug("Loading Local Smart Turn model...")
# Load the pretrained model for sequence classification
self._turn_model = Wav2Vec2BertForSequenceClassification.from_pretrained(
smart_turn_model_path
)
# Load the corresponding feature extractor for preprocessing audio
self._turn_processor = AutoFeatureExtractor.from_pretrained(smart_turn_model_path)
# Set device to GPU if available, else CPU
self._device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
# Move model to selected device and set it to evaluation mode
self._turn_model = self._turn_model.to(self._device)
self._turn_model.eval()
logger.debug("Loaded Local Smart Turn")
async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
inputs = self._turn_processor(
audio_array,
sampling_rate=16000,
padding="max_length",
truncation=True,
max_length=800, # Maximum length as specified in training
return_attention_mask=True,
return_tensors="pt",
)
# Move input tensors to the same device as the model
inputs = {k: v.to(self._device) for k, v in inputs.items()}
# Disable gradient calculation for inference
with torch.no_grad():
outputs = self._turn_model(**inputs)
logits = outputs.logits
probabilities = torch.nn.functional.softmax(logits, dim=1)
completion_prob = probabilities[0, 1].item() # Probability of class 1 (Complete)
prediction = 1 if completion_prob > 0.5 else 0
return {
"prediction": prediction,
"probability": completion_prob,
}