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ai-video-fullstack/backend/services/pipecat/transports.py
Xin Wang 42cab2a6ef Initial commit: AI Video Assistant fullstack platform.
Add pipecat-based backend with WebRTC/WS voice routes, Next.js frontend, and Docker Compose orchestration.

Co-authored-by: Cursor <cursoragent@cursor.com>
2026-06-08 13:51:28 +08:00

55 lines
1.9 KiB
Python

"""Transport 工厂——管线与"输出方式"解耦的关键。
同一条 STT→LLM→TTS 管线,可以挂在不同 transport 上:
- WebRTC:浏览器,低延迟,带 NAT 穿透 -> build_webrtc_transport
- WS: 裸音频流,服务端/话务/自定义客户端,简单 -> build_ws_transport
未来加电话(Twilio/Vonage)只是再加一个 build_xxx_transport + 对应 serializer。
对应 dograh 的 transport_setup.py(WebRTC)+ 各 telephony provider 的 transport.py(WS)。
"""
from fastapi import WebSocket
from pipecat.transports.base_transport import TransportParams
from pipecat.audio.vad.silero import SileroVADAnalyzer
# WebRTC
from pipecat.transports.smallwebrtc.connection import SmallWebRTCConnection
from pipecat.transports.smallwebrtc.transport import SmallWebRTCTransport
# 裸 WS 音频流
from pipecat.transports.network.fastapi_websocket import (
FastAPIWebsocketTransport,
FastAPIWebsocketParams,
)
from pipecat.serializers.protobuf import ProtobufFrameSerializer
def _base_params() -> dict:
"""两种 transport 共享的音频参数。"""
return dict(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(), # 本地 VAD,打断功能依赖它
)
def build_webrtc_transport(connection: SmallWebRTCConnection) -> SmallWebRTCTransport:
return SmallWebRTCTransport(
webrtc_connection=connection,
params=TransportParams(**_base_params()),
)
def build_ws_transport(websocket: WebSocket) -> FastAPIWebsocketTransport:
"""裸 WS 输出。序列化用 protobuf(自定义客户端用同款解码);
若对接电话商,把 serializer 换成对应的 TwilioFrameSerializer 等即可。
"""
return FastAPIWebsocketTransport(
websocket=websocket,
params=FastAPIWebsocketParams(
serializer=ProtobufFrameSerializer(),
**_base_params(),
),
)