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
This commit is contained in:
Kevin
2026-04-27 11:21:16 +08:00
parent 4c3f9a367b
commit 6b3adb4ad8
36 changed files with 1194 additions and 162 deletions

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@@ -2,7 +2,10 @@
from __future__ import annotations
import asyncio
import json
from datetime import datetime, timezone
from pathlib import Path
from unittest.mock import AsyncMock, MagicMock
import pytest
@@ -10,7 +13,12 @@ from fastapi import FastAPI
from fastapi.testclient import TestClient
from app.api import router as api_router
from app.dependencies import get_surgery_pipeline
from app.dependencies import get_surgery_pipeline, get_voice_terminal_hub
from app.services.voice_terminal_hub import (
VoiceTerminalHub,
assign_voice_terminal_after_recording_started,
)
from app.config import Settings
from app.schemas import (
SurgeryConsumptionDetail,
SurgeryPendingConfirmationResponse,
@@ -36,6 +44,8 @@ def api_app(monkeypatch: pytest.MonkeyPatch) -> FastAPI:
app = FastAPI()
app.include_router(api_router)
hub = VoiceTerminalHub(Settings())
app.dependency_overrides[get_voice_terminal_hub] = lambda: hub
return app
@@ -66,6 +76,98 @@ def test_start_surgery_accepted(api_app: FastAPI, instant_sleep: None) -> None:
pipeline.start_recording.assert_awaited_once()
def test_assign_voice_terminal_helper_matches_start_surgery_behavior(
tmp_path: Path,
) -> None:
site = {
"video_rtsp_urls": {"cam1": "rtsp://x/c1"},
"voice_or_room_bindings": [
{
"or_room_id": "R1",
"camera_ids": ["cam1"],
"voice_terminal_id": "TERM-X",
}
],
}
p = tmp_path / "site.json"
p.write_text(json.dumps(site), encoding="utf-8")
hub = VoiceTerminalHub(Settings(or_site_config_json_file=str(p)))
set_vtid = MagicMock()
async def _run() -> None:
await assign_voice_terminal_after_recording_started(
hub,
surgery_id="123456",
camera_ids=["cam1"],
set_voice_terminal_id=set_vtid,
)
asyncio.run(_run())
set_vtid.assert_called_once_with("123456", "TERM-X")
assert hub.get_assignment("TERM-X") == "123456"
def test_start_surgery_notifies_voice_terminal_when_binding_matches(
api_app: FastAPI, instant_sleep: None, tmp_path: Path
) -> None:
site = {
"video_rtsp_urls": {"cam1": "rtsp://x/c1", "cam2": "rtsp://x/c2"},
"voice_or_room_bindings": [
{
"or_room_id": "R1",
"camera_ids": ["cam1", "cam2"],
"voice_terminal_id": "TERM-OR1",
}
],
}
p = tmp_path / "site.json"
p.write_text(json.dumps(site), encoding="utf-8")
hub = VoiceTerminalHub(Settings(or_site_config_json_file=str(p)))
api_app.dependency_overrides[get_voice_terminal_hub] = lambda: hub
pipeline = MagicMock()
pipeline.start_recording = AsyncMock(return_value=None)
pipeline.set_voice_terminal_id = MagicMock()
api_app.dependency_overrides[get_surgery_pipeline] = lambda: pipeline
client = TestClient(api_app)
r = client.post(
"/client/surgeries/start",
json={
"surgery_id": "123456",
"camera_ids": ["cam2", "cam1"],
"candidate_consumables": ["纱布"],
},
)
assert r.status_code == 200
pipeline.set_voice_terminal_id.assert_called_once_with("123456", "TERM-OR1")
assert hub.get_assignment("TERM-OR1") == "123456"
def test_voice_terminal_assignment_get(api_app: FastAPI) -> None:
hub = api_app.dependency_overrides[get_voice_terminal_hub]()
asyncio.run(hub.notify_start("t-assign", "999999"))
client = TestClient(api_app)
r = client.get("/client/voice-terminals/t-assign/assignment")
assert r.status_code == 200
assert r.json() == {
"voice_terminal_id": "t-assign",
"active_surgery_id": "999999",
}
def test_end_surgery_notifies_voice_terminal(
api_app: FastAPI, instant_sleep: None
) -> None:
hub = api_app.dependency_overrides[get_voice_terminal_hub]()
asyncio.run(hub.notify_start("t-end", "123456"))
pipeline = MagicMock()
pipeline.stop_recording = AsyncMock(return_value="t-end")
api_app.dependency_overrides[get_surgery_pipeline] = lambda: pipeline
client = TestClient(api_app)
r = client.post("/client/surgeries/end", json={"surgery_id": "123456"})
assert r.status_code == 200
assert hub.get_assignment("t-end") is None
def test_start_surgery_503_on_pipeline_error(
api_app: FastAPI, instant_sleep: None
) -> None:

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@@ -72,9 +72,12 @@ class _StubCameraSessionManager:
)
await self._registry.register(surgery_id, run)
def set_voice_terminal_id(self, surgery_id: str, terminal_id: str | None) -> None:
self._real.set_voice_terminal_id(surgery_id, terminal_id)
async def stop_surgery(
self, surgery_id: str, *, require_active: bool = True
) -> None:
) -> str | None:
run = await self._registry.unregister(surgery_id)
if run is None:
if require_active:
@@ -84,9 +87,11 @@ class _StubCameraSessionManager:
"RECORDING_NOT_STOPPED",
"停录未能完成:当前没有该手术的活跃录制会话。",
)
return
return None
voice_tid = run.state.voice_terminal_id
details = list(run.state.details)
await self._archive.persist_or_archive(surgery_id, details)
return voice_tid
def __getattr__(self, name: str) -> Any:
return getattr(self._real, name)

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@@ -0,0 +1,134 @@
"""术间绑定与站点配置OR_SITE_CONFIG_JSON解析。"""
from __future__ import annotations
import json
from pathlib import Path
import pytest
from app.config import Settings
from app.or_site_config import (
load_or_site_config_from_path,
merge_video_rtsp_urls_into_file,
parse_or_site_config_object,
)
from app.services.voice_terminal_binding import VoiceTerminalBindingIndex
def test_parse_and_resolve() -> None:
idx = VoiceTerminalBindingIndex.from_binding_list(
[
{
"or_room_id": "OR-05",
"camera_ids": ["or-cam-01", "or-cam-02"],
"voice_terminal_id": "T-05",
}
]
)
assert idx is not None
assert idx.resolve_terminal(["or-cam-02", "or-cam-01"]) == "T-05"
assert idx.resolve_terminal(["or-cam-01"]) == "T-05"
def test_resolve_subset_uses_smallest_superset_when_ambiguous() -> None:
idx = VoiceTerminalBindingIndex.from_binding_list(
[
{
"or_room_id": "A",
"camera_ids": ["c1", "c2"],
"voice_terminal_id": "T-AB",
},
{
"or_room_id": "B",
"camera_ids": ["c1", "c2", "c3"],
"voice_terminal_id": "T-ABC",
},
]
)
assert idx is not None
assert idx.resolve_terminal(["c1"]) == "T-AB"
def test_duplicate_terminal_rejected() -> None:
assert (
VoiceTerminalBindingIndex.from_binding_list(
[
{"or_room_id": "A", "camera_ids": ["c1"], "voice_terminal_id": "T"},
{"or_room_id": "B", "camera_ids": ["c2"], "voice_terminal_id": "T"},
]
)
is None
)
def test_duplicate_camera_set_rejected() -> None:
assert (
VoiceTerminalBindingIndex.from_binding_list(
[
{"or_room_id": "A", "camera_ids": ["c1"], "voice_terminal_id": "T1"},
{"or_room_id": "B", "camera_ids": ["c1"], "voice_terminal_id": "T2"},
]
)
is None
)
def test_from_binding_list_empty() -> None:
idx = VoiceTerminalBindingIndex.from_binding_list([])
assert idx is not None
assert idx.resolve_terminal(["x"]) is None
def test_parse_site_config_rejects_unknown_keys() -> None:
with pytest.raises(ValueError, match="unknown top-level"):
parse_or_site_config_object(
{
"video_rtsp_urls": {},
"voice_or_room_bindings": [],
"extra": 1,
}
)
def test_parse_site_config_requires_both_sections() -> None:
with pytest.raises(ValueError, match="must include"):
parse_or_site_config_object({"video_rtsp_urls": {}})
def test_load_from_path_roundtrip(tmp_path: Path) -> None:
p = tmp_path / "site.json"
p.write_text(
'{"video_rtsp_urls":{"c1":"rtsp://h"},"voice_or_room_bindings":['
'{"or_room_id":"R","camera_ids":["a"],"voice_terminal_id":"T"}]}',
encoding="utf-8",
)
cfg = load_or_site_config_from_path(p)
assert cfg.video_rtsp_urls == {"c1": "rtsp://h"}
assert cfg.voice_bindings.resolve_terminal(["a"]) == "T"
def test_settings_video_map(tmp_path: Path) -> None:
p = tmp_path / "site.json"
p.write_text(
'{"video_rtsp_urls":{"c1":"rtsp://x"},"voice_or_room_bindings":[]}',
encoding="utf-8",
)
s = Settings(or_site_config_json_file=str(p))
assert s.video_rtsp_url_map() == {"c1": "rtsp://x"}
def test_merge_preserves_bindings(tmp_path: Path) -> None:
p = tmp_path / "site.json"
p.write_text(
'{"video_rtsp_urls":{"old":"rtsp://old"},'
'"voice_or_room_bindings":[{"or_room_id":"R","camera_ids":["x"],'
'"voice_terminal_id":"T"}]}',
encoding="utf-8",
)
merge_video_rtsp_urls_into_file(
p, {"x": "rtsp://127.0.0.1:1/p"}, replace_host="127.0.0.1"
)
cfg = load_or_site_config_from_path(p)
assert cfg.video_rtsp_urls == {"x": "rtsp://127.0.0.1:1/p"}
assert cfg.voice_bindings.resolve_terminal(["x"]) == "T"