230 lines
8.3 KiB
Python
230 lines
8.3 KiB
Python
from __future__ import annotations
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import uuid
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from loguru import logger
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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.frames.frames import (
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LLMRunFrame,
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OutputTransportMessageUrgentFrame,
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TTSSpeakFrame,
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)
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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.llm_context import LLMContext
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from pipecat.processors.aggregators.llm_response_universal import (
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AssistantTurnStoppedMessage,
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LLMContextAggregatorPair,
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LLMUserAggregatorParams,
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UserTurnStoppedMessage,
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)
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from pipecat.serializers.protobuf import ProtobufFrameSerializer
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from pipecat.serializers.base_serializer import FrameSerializer
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from pipecat.transports.websocket.fastapi import (
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FastAPIWebsocketParams,
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FastAPIWebsocketTransport,
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)
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from pipecat.turns.user_stop.speech_timeout_user_turn_stop_strategy import (
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SpeechTimeoutUserTurnStopStrategy,
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)
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from pipecat.turns.user_turn_strategies import UserTurnStrategies
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from .config import EngineConfig
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from .context_sync import AssistantContextSyncProcessor
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from .product_protocol import ProductWebsocketSerializer
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from .services import create_llm_service, create_stt_service, create_tts_service
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from .text_input import ProductTextInputProcessor
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from .text_stream import ProductTextStreamProcessor, maybe_sync_assistant_context
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from .transcript_stream import ProductTranscriptStreamProcessor
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from .turn_start import InterruptionGateUserTurnStartStrategy
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async def run_voice_pipeline(websocket, config: EngineConfig) -> None:
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await run_pipeline_with_serializer(
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websocket,
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config,
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serializer=ProtobufFrameSerializer(),
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client_label="Pipecat protobuf",
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)
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async def run_product_voice_pipeline(websocket, config: EngineConfig) -> None:
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await run_pipeline_with_serializer(
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websocket,
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config,
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serializer=ProductWebsocketSerializer(
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sample_rate=config.audio.sample_rate_hz,
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channels=config.audio.channels,
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),
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client_label="Product JSON",
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)
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async def run_pipeline_with_serializer(
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websocket,
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config: EngineConfig,
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*,
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serializer: FrameSerializer,
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client_label: str,
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) -> None:
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transport = FastAPIWebsocketTransport(
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websocket=websocket,
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params=FastAPIWebsocketParams(
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audio_in_enabled=True,
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audio_out_enabled=True,
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audio_in_sample_rate=config.audio.sample_rate_hz,
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audio_out_sample_rate=config.audio.sample_rate_hz,
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audio_in_channels=config.audio.channels,
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audio_out_channels=config.audio.channels,
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serializer=serializer,
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session_timeout=None,
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),
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)
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stt = create_stt_service(config.services.stt, config.audio)
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llm_config = config.services.llm
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chat_id = llm_config.chat_id or f"voice_{uuid.uuid4().hex[:16]}"
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llm = create_llm_service(
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llm_config,
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chat_id=chat_id,
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session_variables={"session_id": chat_id, "channel": "voice"},
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greeting_prompt=config.agent.greeting,
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)
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if llm_config.provider == "fastgpt":
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logger.info(f"FastGPT chatId={chat_id}")
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tts = create_tts_service(config.services.tts, config.audio)
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use_fastgpt = llm_config.provider == "fastgpt" and not llm_config.send_system_prompt
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messages: list[dict[str, str]] = []
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if not use_fastgpt:
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messages = [{"role": "system", "content": config.agent.system_prompt}]
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if config.agent.greeting and config.agent.greeting_mode == "generated":
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messages.append({"role": "system", "content": config.agent.greeting})
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context = LLMContext(messages)
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vad_params = VADParams(
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confidence=config.turn.vad.confidence,
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start_secs=config.turn.vad.start_secs,
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stop_secs=config.turn.vad.stop_secs,
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min_volume=config.turn.vad.min_volume,
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)
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# Replace pipecat's default stop strategy (Smart Turn v3) with a simple
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# silence-timeout strategy. Smart Turn v3 was finalizing every short
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# Chinese phrase as a complete turn, which caused one logical utterance
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# to become several LLM calls and several user bubbles in the UI. The
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# timeout strategy waits for `user_speech_timeout_sec` of silence
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# (re-armed every time the user resumes speaking) before declaring the
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# turn finished — which is what we actually want for streaming ASRs.
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user_turn_strategies = UserTurnStrategies(
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start=[
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InterruptionGateUserTurnStartStrategy(
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min_chars_when_bot_speaking=config.turn.interruption_min_chars,
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allowed_short_replies=config.turn.interruption_short_replies,
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use_interim=config.turn.interruption_use_interim,
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),
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],
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stop=[
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SpeechTimeoutUserTurnStopStrategy(
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user_speech_timeout=config.turn.user_speech_timeout_sec,
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),
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],
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)
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user_aggregator, assistant_aggregator = LLMContextAggregatorPair(
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context,
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user_params=LLMUserAggregatorParams(
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vad_analyzer=SileroVADAnalyzer(params=vad_params),
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user_turn_strategies=user_turn_strategies,
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),
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)
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text_stream = ProductTextStreamProcessor()
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context_sync = AssistantContextSyncProcessor(
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text_stream=text_stream,
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assistant_aggregator=assistant_aggregator,
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)
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pipeline = Pipeline(
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[
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transport.input(),
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ProductTextInputProcessor(),
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stt,
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ProductTranscriptStreamProcessor(),
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context_sync,
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user_aggregator,
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llm,
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text_stream,
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tts,
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transport.output(),
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assistant_aggregator,
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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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audio_in_sample_rate=config.audio.sample_rate_hz,
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audio_out_sample_rate=config.audio.sample_rate_hz,
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enable_metrics=True,
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enable_usage_metrics=True,
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enable_heartbeats=True,
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),
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idle_timeout_secs=config.session.inactivity_timeout_sec,
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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_label} websocket client connected")
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if config.agent.greeting_mode == "fixed" and config.agent.greeting:
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await task.queue_frames([TTSSpeakFrame(config.agent.greeting)])
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elif config.agent.greeting_mode == "generated":
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await task.queue_frames([LLMRunFrame()])
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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_label} websocket client disconnected")
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await task.cancel()
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@transport.event_handler("on_session_timeout")
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async def on_session_timeout(_transport, _client):
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logger.info(f"{client_label} websocket session timed out")
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await task.cancel()
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@user_aggregator.event_handler("on_user_turn_stopped")
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async def on_user_turn_stopped(_aggregator, _strategy, message: UserTurnStoppedMessage):
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logger.info(f"User: {message.content}")
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text = (message.content or "").strip()
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if not text:
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return
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await task.queue_frame(
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OutputTransportMessageUrgentFrame(
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message={
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"type": "input.transcript.final",
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"text": text,
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"user_id": message.user_id,
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"timestamp": message.timestamp,
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}
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)
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)
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# NOTE: assistant turn started/final events are emitted by
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# ProductTextStreamProcessor, upstream of TTS, so text streams to the
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# client ahead of audio. This logger is kept for server-side visibility.
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@assistant_aggregator.event_handler("on_assistant_turn_stopped")
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async def on_assistant_turn_stopped(_aggregator, message: AssistantTurnStoppedMessage):
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logger.info(f"Assistant: {message.content}")
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maybe_sync_assistant_context(
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_aggregator,
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text_stream,
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committed_text=message.content or "",
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)
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text_stream.take_interrupted_stream_text()
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runner = PipelineRunner(handle_sigint=False)
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await runner.run(task)
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