feat: 站点 JSON、语音终端 WebSocket 指派与客户端联调
- 用 OR_SITE_CONFIG_JSON_FILE 统一术间配置(video_rtsp_urls + voice_or_room_bindings) - VoiceTerminalHub:assignment、WS 推送与 HTTP 查询;开录/停录后 notify - 一键联调 orchestrate-and-start 与 /client/surgeries/start 共用指派逻辑,修复 demo 路径不发 WS - 语音桌面端:SIGINT 退出、shutdown 清理、仅 WS 指派、固定 pending 轮询间隔、界面仅保留录音时长 - 新增/调整契约与绑定测试,文档与示例配置同步 Made-with: Cursor
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134
tests/test_voice_terminal_binding.py
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134
tests/test_voice_terminal_binding.py
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"""术间绑定与站点配置(OR_SITE_CONFIG_JSON)解析。"""
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from __future__ import annotations
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import json
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from pathlib import Path
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import pytest
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from app.config import Settings
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from app.or_site_config import (
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load_or_site_config_from_path,
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merge_video_rtsp_urls_into_file,
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parse_or_site_config_object,
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)
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from app.services.voice_terminal_binding import VoiceTerminalBindingIndex
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def test_parse_and_resolve() -> None:
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idx = VoiceTerminalBindingIndex.from_binding_list(
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[
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{
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"or_room_id": "OR-05",
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"camera_ids": ["or-cam-01", "or-cam-02"],
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"voice_terminal_id": "T-05",
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}
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]
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)
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assert idx is not None
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assert idx.resolve_terminal(["or-cam-02", "or-cam-01"]) == "T-05"
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assert idx.resolve_terminal(["or-cam-01"]) == "T-05"
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def test_resolve_subset_uses_smallest_superset_when_ambiguous() -> None:
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idx = VoiceTerminalBindingIndex.from_binding_list(
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[
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{
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"or_room_id": "A",
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"camera_ids": ["c1", "c2"],
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"voice_terminal_id": "T-AB",
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},
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{
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"or_room_id": "B",
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"camera_ids": ["c1", "c2", "c3"],
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"voice_terminal_id": "T-ABC",
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},
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]
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)
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assert idx is not None
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assert idx.resolve_terminal(["c1"]) == "T-AB"
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def test_duplicate_terminal_rejected() -> None:
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assert (
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VoiceTerminalBindingIndex.from_binding_list(
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[
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{"or_room_id": "A", "camera_ids": ["c1"], "voice_terminal_id": "T"},
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{"or_room_id": "B", "camera_ids": ["c2"], "voice_terminal_id": "T"},
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]
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)
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is None
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)
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def test_duplicate_camera_set_rejected() -> None:
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assert (
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VoiceTerminalBindingIndex.from_binding_list(
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[
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{"or_room_id": "A", "camera_ids": ["c1"], "voice_terminal_id": "T1"},
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{"or_room_id": "B", "camera_ids": ["c1"], "voice_terminal_id": "T2"},
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]
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)
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is None
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)
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def test_from_binding_list_empty() -> None:
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idx = VoiceTerminalBindingIndex.from_binding_list([])
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assert idx is not None
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assert idx.resolve_terminal(["x"]) is None
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def test_parse_site_config_rejects_unknown_keys() -> None:
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with pytest.raises(ValueError, match="unknown top-level"):
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parse_or_site_config_object(
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{
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"video_rtsp_urls": {},
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"voice_or_room_bindings": [],
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"extra": 1,
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}
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)
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def test_parse_site_config_requires_both_sections() -> None:
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with pytest.raises(ValueError, match="must include"):
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parse_or_site_config_object({"video_rtsp_urls": {}})
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def test_load_from_path_roundtrip(tmp_path: Path) -> None:
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p = tmp_path / "site.json"
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p.write_text(
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'{"video_rtsp_urls":{"c1":"rtsp://h"},"voice_or_room_bindings":['
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'{"or_room_id":"R","camera_ids":["a"],"voice_terminal_id":"T"}]}',
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encoding="utf-8",
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)
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cfg = load_or_site_config_from_path(p)
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assert cfg.video_rtsp_urls == {"c1": "rtsp://h"}
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assert cfg.voice_bindings.resolve_terminal(["a"]) == "T"
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def test_settings_video_map(tmp_path: Path) -> None:
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p = tmp_path / "site.json"
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p.write_text(
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'{"video_rtsp_urls":{"c1":"rtsp://x"},"voice_or_room_bindings":[]}',
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encoding="utf-8",
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)
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s = Settings(or_site_config_json_file=str(p))
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assert s.video_rtsp_url_map() == {"c1": "rtsp://x"}
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def test_merge_preserves_bindings(tmp_path: Path) -> None:
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p = tmp_path / "site.json"
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p.write_text(
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'{"video_rtsp_urls":{"old":"rtsp://old"},'
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'"voice_or_room_bindings":[{"or_room_id":"R","camera_ids":["x"],'
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'"voice_terminal_id":"T"}]}',
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encoding="utf-8",
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)
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merge_video_rtsp_urls_into_file(
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p, {"x": "rtsp://127.0.0.1:1/p"}, replace_host="127.0.0.1"
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)
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cfg = load_or_site_config_from_path(p)
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assert cfg.video_rtsp_urls == {"x": "rtsp://127.0.0.1:1/p"}
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assert cfg.voice_bindings.resolve_terminal(["x"]) == "T"
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