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