mirror of
https://github.com/Monadical-SAS/reflector.git
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feat: migrate file and live post-processing pipelines from Celery to Hatchet workflow engine (#911)
* feat: migrate file and live post-processing pipelines from Celery to Hatchet workflow engine * fix: always force reprocessing * fix: ci tests with live pipelines * fix: ci tests with live pipelines
This commit is contained in:
Juan Diego García
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GitHub
GitHub
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72dca7cacc
commit
37a1f01850
@@ -2,6 +2,10 @@
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# FIXME test status of transcript
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# FIXME test websocket connection after RTC is finished still send the full events
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# FIXME try with locked session, RTC should not work
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# TODO: add integration tests for post-processing (LivePostPipeline) with a real
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# Hatchet instance. These tests currently only cover the live pipeline.
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# Post-processing events (WAVEFORM, FINAL_*, DURATION, STATUS=ended, mp3)
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# are now dispatched via Hatchet and tested in test_hatchet_live_post_pipeline.py.
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import asyncio
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import json
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@@ -49,7 +53,7 @@ class ThreadedUvicorn:
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@pytest.fixture
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def appserver(tmpdir, setup_database, celery_session_app, celery_session_worker):
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def appserver(tmpdir, setup_database):
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import threading
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from reflector.app import app
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@@ -119,8 +123,6 @@ def appserver(tmpdir, setup_database, celery_session_app, celery_session_worker)
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@pytest.mark.usefixtures("setup_database")
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@pytest.mark.usefixtures("celery_session_app")
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@pytest.mark.usefixtures("celery_session_worker")
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@pytest.mark.asyncio
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async def test_transcript_rtc_and_websocket(
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tmpdir,
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@@ -134,6 +136,7 @@ async def test_transcript_rtc_and_websocket(
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appserver,
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client,
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monkeypatch,
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mock_hatchet_client,
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):
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# goal: start the server, exchange RTC, receive websocket events
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# because of that, we need to start the server in a thread
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@@ -208,35 +211,30 @@ async def test_transcript_rtc_and_websocket(
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stream_client.channel.send(json.dumps({"cmd": "STOP"}))
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await stream_client.stop()
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# wait the processing to finish
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timeout = 120
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# Wait for live pipeline to flush (it dispatches post-processing to Hatchet)
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timeout = 30
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while True:
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# fetch the transcript and check if it is ended
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resp = await client.get(f"/transcripts/{tid}")
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assert resp.status_code == 200
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if resp.json()["status"] in ("ended", "error"):
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if resp.json()["status"] in ("processing", "ended", "error"):
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break
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await asyncio.sleep(1)
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timeout -= 1
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if timeout < 0:
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raise TimeoutError("Timeout while waiting for transcript to be ended")
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if resp.json()["status"] != "ended":
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raise TimeoutError("Transcript processing failed")
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raise TimeoutError("Timeout waiting for live pipeline to finish")
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# stop websocket task
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websocket_task.cancel()
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# check events
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# check live pipeline events
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assert len(events) > 0
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from pprint import pprint
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pprint(events)
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# get events list
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eventnames = [e["event"] for e in events]
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# check events
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# Live pipeline produces TRANSCRIPT and TOPIC events during RTC
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assert "TRANSCRIPT" in eventnames
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ev = events[eventnames.index("TRANSCRIPT")]
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assert ev["data"]["text"].startswith("Hello world.")
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@@ -249,50 +247,18 @@ async def test_transcript_rtc_and_websocket(
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assert ev["data"]["transcript"].startswith("Hello world.")
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assert ev["data"]["timestamp"] == 0.0
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assert "FINAL_LONG_SUMMARY" in eventnames
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ev = events[eventnames.index("FINAL_LONG_SUMMARY")]
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assert ev["data"]["long_summary"] == "LLM LONG SUMMARY"
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assert "FINAL_SHORT_SUMMARY" in eventnames
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ev = events[eventnames.index("FINAL_SHORT_SUMMARY")]
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assert ev["data"]["short_summary"] == "LLM SHORT SUMMARY"
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assert "FINAL_TITLE" in eventnames
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ev = events[eventnames.index("FINAL_TITLE")]
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assert ev["data"]["title"] == "Llm Title"
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assert "WAVEFORM" in eventnames
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ev = events[eventnames.index("WAVEFORM")]
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assert isinstance(ev["data"]["waveform"], list)
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assert len(ev["data"]["waveform"]) >= 250
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waveform_resp = await client.get(f"/transcripts/{tid}/audio/waveform")
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assert waveform_resp.status_code == 200
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assert waveform_resp.headers["content-type"] == "application/json"
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assert isinstance(waveform_resp.json()["data"], list)
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assert len(waveform_resp.json()["data"]) >= 250
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# check status order
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# Live pipeline status progression
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statuses = [e["data"]["value"] for e in events if e["event"] == "STATUS"]
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assert "recording" in statuses
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assert "processing" in statuses
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assert statuses.index("recording") < statuses.index("processing")
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assert statuses.index("processing") < statuses.index("ended")
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# ensure the last event received is ended
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assert events[-1]["event"] == "STATUS"
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assert events[-1]["data"]["value"] == "ended"
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# check on the latest response that the audio duration is > 0
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assert resp.json()["duration"] > 0
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assert "DURATION" in eventnames
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# check that audio/mp3 is available
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audio_resp = await client.get(f"/transcripts/{tid}/audio/mp3")
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assert audio_resp.status_code == 200
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assert audio_resp.headers["Content-Type"] == "audio/mpeg"
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# Post-processing (WAVEFORM, FINAL_*, DURATION, mp3, STATUS=ended) is now
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# dispatched to Hatchet via LivePostPipeline — not tested here.
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# See test_hatchet_live_post_pipeline.py for post-processing tests.
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@pytest.mark.usefixtures("setup_database")
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@pytest.mark.usefixtures("celery_session_app")
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@pytest.mark.usefixtures("celery_session_worker")
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@pytest.mark.asyncio
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async def test_transcript_rtc_and_websocket_and_fr(
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tmpdir,
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@@ -306,6 +272,7 @@ async def test_transcript_rtc_and_websocket_and_fr(
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appserver,
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client,
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monkeypatch,
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mock_hatchet_client,
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):
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# goal: start the server, exchange RTC, receive websocket events
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# because of that, we need to start the server in a thread
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@@ -382,42 +349,34 @@ async def test_transcript_rtc_and_websocket_and_fr(
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# instead of waiting a long time, we just send a STOP
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stream_client.channel.send(json.dumps({"cmd": "STOP"}))
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# wait the processing to finish
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await asyncio.sleep(2)
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await stream_client.stop()
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# wait the processing to finish
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timeout = 120
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# Wait for live pipeline to flush
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timeout = 30
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while True:
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# fetch the transcript and check if it is ended
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resp = await client.get(f"/transcripts/{tid}")
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assert resp.status_code == 200
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if resp.json()["status"] == "ended":
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if resp.json()["status"] in ("processing", "ended", "error"):
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break
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await asyncio.sleep(1)
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timeout -= 1
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if timeout < 0:
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raise TimeoutError("Timeout while waiting for transcript to be ended")
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if resp.json()["status"] != "ended":
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raise TimeoutError("Transcript processing failed")
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await asyncio.sleep(2)
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raise TimeoutError("Timeout waiting for live pipeline to finish")
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# stop websocket task
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websocket_task.cancel()
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# check events
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# check live pipeline events
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assert len(events) > 0
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from pprint import pprint
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pprint(events)
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# get events list
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eventnames = [e["event"] for e in events]
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# check events
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# Live pipeline produces TRANSCRIPT with translation
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assert "TRANSCRIPT" in eventnames
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ev = events[eventnames.index("TRANSCRIPT")]
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assert ev["data"]["text"].startswith("Hello world.")
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@@ -430,23 +389,11 @@ async def test_transcript_rtc_and_websocket_and_fr(
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assert ev["data"]["transcript"].startswith("Hello world.")
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assert ev["data"]["timestamp"] == 0.0
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assert "FINAL_LONG_SUMMARY" in eventnames
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ev = events[eventnames.index("FINAL_LONG_SUMMARY")]
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assert ev["data"]["long_summary"] == "LLM LONG SUMMARY"
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assert "FINAL_SHORT_SUMMARY" in eventnames
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ev = events[eventnames.index("FINAL_SHORT_SUMMARY")]
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assert ev["data"]["short_summary"] == "LLM SHORT SUMMARY"
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assert "FINAL_TITLE" in eventnames
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ev = events[eventnames.index("FINAL_TITLE")]
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assert ev["data"]["title"] == "Llm Title"
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# check status order
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# Live pipeline status progression
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statuses = [e["data"]["value"] for e in events if e["event"] == "STATUS"]
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assert "recording" in statuses
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assert "processing" in statuses
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assert statuses.index("recording") < statuses.index("processing")
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assert statuses.index("processing") < statuses.index("ended")
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# ensure the last event received is ended
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assert events[-1]["event"] == "STATUS"
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assert events[-1]["data"]["value"] == "ended"
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# Post-processing (FINAL_*, STATUS=ended) is now dispatched to Hatchet
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# via LivePostPipeline — not tested here.
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