test: full integration tests (#916)

* test: full integration tests

* fix: add env vars as secrets in CI

server/tests/integration/test_live_pipeline.py

@@ -0,0 +1,109 @@
+"""
+Integration test: WebRTC stream → LivePostProcessingPipeline → full processing.
+
+Exercises: WebRTC SDP exchange → live audio streaming → connection close →
+Hatchet LivePostPipeline → whisper transcription → LLM summarization/topics → status "ended".
+"""
+
+import asyncio
+import json
+import os
+
+import httpx
+import pytest
+from aiortc import RTCPeerConnection, RTCSessionDescription
+from aiortc.contrib.media import MediaPlayer
+
+SERVER_URL = os.environ.get("SERVER_URL", "http://server:1250")
+
+
+@pytest.mark.asyncio
+async def test_live_pipeline_end_to_end(
+    api_client, test_records_dir, poll_transcript_status
+):
+    """Stream audio via WebRTC and verify the full post-processing pipeline completes."""
+    # 1. Create transcript
+    resp = await api_client.post(
+        "/transcripts",
+        json={"name": "integration-live-test"},
+    )
+    assert resp.status_code == 200, f"Failed to create transcript: {resp.text}"
+    transcript = resp.json()
+    transcript_id = transcript["id"]
+
+    # 2. Set up WebRTC peer connection with audio from test file
+    audio_path = test_records_dir / "test_short.wav"
+    assert audio_path.exists(), f"Test audio file not found: {audio_path}"
+
+    pc = RTCPeerConnection()
+    player = MediaPlayer(audio_path.as_posix())
+
+    # Add audio track
+    audio_track = player.audio
+    pc.addTrack(audio_track)
+
+    # Create data channel (server expects this for STOP command)
+    channel = pc.createDataChannel("data-channel")
+
+    # 3. Generate SDP offer
+    offer = await pc.createOffer()
+    await pc.setLocalDescription(offer)
+
+    sdp_payload = {
+        "sdp": pc.localDescription.sdp,
+        "type": pc.localDescription.type,
+    }
+
+    # 4. Send offer to server and get answer
+    webrtc_url = f"{SERVER_URL}/v1/transcripts/{transcript_id}/record/webrtc"
+    async with httpx.AsyncClient(timeout=httpx.Timeout(30.0)) as client:
+        resp = await client.post(webrtc_url, json=sdp_payload)
+    assert resp.status_code == 200, f"WebRTC offer failed: {resp.text}"
+
+    answer_data = resp.json()
+    answer = RTCSessionDescription(sdp=answer_data["sdp"], type=answer_data["type"])
+    await pc.setRemoteDescription(answer)
+
+    # 5. Wait for audio playback to finish
+    max_stream_wait = 60
+    elapsed = 0
+    while elapsed < max_stream_wait:
+        if audio_track.readyState == "ended":
+            break
+        await asyncio.sleep(0.5)
+        elapsed += 0.5
+
+    # 6. Send STOP command and close connection
+    try:
+        channel.send(json.dumps({"cmd": "STOP"}))
+        await asyncio.sleep(1)
+    except Exception:
+        pass  # Channel may not be open if track ended quickly
+
+    await pc.close()
+
+    # 7. Poll until post-processing pipeline completes
+    data = await poll_transcript_status(
+        api_client, transcript_id, target="ended", max_wait=300
+    )
+
+    # 8. Assertions
+    assert data["status"] == "ended"
+    assert data.get("title") and len(data["title"]) > 0, "Title should be non-empty"
+    assert (
+        data.get("long_summary") and len(data["long_summary"]) > 0
+    ), "Long summary should be non-empty"
+    assert (
+        data.get("short_summary") and len(data["short_summary"]) > 0
+    ), "Short summary should be non-empty"
+
+    # Topics are served from a separate endpoint
+    topics_resp = await api_client.get(f"/transcripts/{transcript_id}/topics")
+    assert topics_resp.status_code == 200, f"Failed to get topics: {topics_resp.text}"
+    topics = topics_resp.json()
+    assert len(topics) >= 1, "Should have at least 1 topic"
+    for topic in topics:
+        assert topic.get("title"), "Each topic should have a title"
+        assert topic.get("summary"), "Each topic should have a summary"
+
+    assert data.get("duration", 0) > 0, "Duration should be positive"