43cc188d85
- Introduce a new parameter `audio_out_end_silence_secs` in the `_base_params` function to control the duration of silence added after the end frame, allowing for smoother call termination. - Set the default value to 0 to ensure immediate hang-up after the end speech, enhancing user experience during call endings.
56 lines
2.0 KiB
Python
56 lines
2.0 KiB
Python
"""Transport 工厂——管线与"输出方式"解耦的关键。
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同一条 STT→LLM→TTS 管线,可以挂在不同 transport 上:
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- WebRTC:浏览器,低延迟,带 NAT 穿透 -> build_webrtc_transport
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- WS: 裸音频流,服务端/话务/自定义客户端,简单 -> build_ws_transport
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未来加电话(Twilio/Vonage)只是再加一个 build_xxx_transport + 对应 serializer。
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对应 dograh 的 transport_setup.py(WebRTC)+ 各 telephony provider 的 transport.py(WS)。
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"""
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from fastapi import WebSocket
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from pipecat.transports.base_transport import TransportParams
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# WebRTC
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from pipecat.transports.smallwebrtc.connection import SmallWebRTCConnection
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from pipecat.transports.smallwebrtc.transport import SmallWebRTCTransport
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# 裸 WS 音频流
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from pipecat.transports.websocket.fastapi import (
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FastAPIWebsocketTransport,
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FastAPIWebsocketParams,
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)
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from pipecat.serializers.protobuf import ProtobufFrameSerializer
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def _base_params() -> dict:
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"""两种 transport 共享的音频参数。"""
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return dict(
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audio_in_enabled=True,
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audio_out_enabled=True,
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# EndFrame 后默认补 2s 静音(防止收尾被截断)。我们的挂断已等到机器人
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# 说完才触发,这段静音纯属空等,置 0 让结束语播完立即挂断。
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audio_out_end_silence_secs=0,
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)
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def build_webrtc_transport(connection: SmallWebRTCConnection) -> SmallWebRTCTransport:
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return SmallWebRTCTransport(
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webrtc_connection=connection,
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params=TransportParams(**_base_params()),
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)
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def build_ws_transport(websocket: WebSocket) -> FastAPIWebsocketTransport:
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"""裸 WS 输出。序列化用 protobuf(自定义客户端用同款解码);
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若对接电话商,把 serializer 换成对应的 TwilioFrameSerializer 等即可。
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"""
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return FastAPIWebsocketTransport(
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websocket=websocket,
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params=FastAPIWebsocketParams(
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serializer=ProtobufFrameSerializer(),
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**_base_params(),
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),
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)
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