api: implement first server API + tests

server/poetry.lock

@@ -682,6 +682,78 @@ humanfriendly = ">=9.1"
 [package.extras]
 cron = ["capturer (>=2.4)"]
 
+[[package]]
+name = "coverage"
+version = "7.2.7"
+description = "Code coverage measurement for Python"
+optional = false
+python-versions = ">=3.7"
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+    {file = "coverage-7.2.7-pp37.pp38.pp39-none-any.whl", hash = "sha256:b7b4c971f05e6ae490fef852c218b0e79d4e52f79ef0c8475566584a8fb3e01d"},
+    {file = "coverage-7.2.7.tar.gz", hash = "sha256:924d94291ca674905fe9481f12294eb11f2d3d3fd1adb20314ba89e94f44ed59"},
+]
+
+[package.extras]
+toml = ["tomli"]
+
 [[package]]
 name = "cryptography"
 version = "41.0.2"
@@ -1169,6 +1241,23 @@ cli = ["click (==8.*)", "pygments (==2.*)", "rich (>=10,<14)"]
 http2 = ["h2 (>=3,<5)"]
 socks = ["socksio (==1.*)"]
 
+[[package]]
+name = "httpx-ws"
+version = "0.4.1"
+description = "WebSockets support for HTTPX"
+optional = false
+python-versions = ">=3.8"
+files = [
+    {file = "httpx_ws-0.4.1-py3-none-any.whl", hash = "sha256:01bdaeb66add8196485dc39912abd0a3e95b67c244aededc151156ac6adca850"},
+    {file = "httpx_ws-0.4.1.tar.gz", hash = "sha256:5f3e291e8fb99c89f994329d883e5679d02a0b5b12a1e414f7f8630c276b6744"},
+]
+
+[package.dependencies]
+anyio = "*"
+httpcore = ">=0.17.3,<0.18"
+httpx = ">=0.23.1"
+wsproto = "*"
+
 [[package]]
 name = "huggingface-hub"
 version = "0.16.4"
@@ -1883,6 +1972,24 @@ pytest = ">=7.0.0"
 docs = ["sphinx (>=5.3)", "sphinx-rtd-theme (>=1.0)"]
 testing = ["coverage (>=6.2)", "flaky (>=3.5.0)", "hypothesis (>=5.7.1)", "mypy (>=0.931)", "pytest-trio (>=0.7.0)"]
 
+[[package]]
+name = "pytest-cov"
+version = "4.1.0"
+description = "Pytest plugin for measuring coverage."
+optional = false
+python-versions = ">=3.7"
+files = [
+    {file = "pytest-cov-4.1.0.tar.gz", hash = "sha256:3904b13dfbfec47f003b8e77fd5b589cd11904a21ddf1ab38a64f204d6a10ef6"},
+    {file = "pytest_cov-4.1.0-py3-none-any.whl", hash = "sha256:6ba70b9e97e69fcc3fb45bfeab2d0a138fb65c4d0d6a41ef33983ad114be8c3a"},
+]
+
+[package.dependencies]
+coverage = {version = ">=5.2.1", extras = ["toml"]}
+pytest = ">=4.6"
+
+[package.extras]
+testing = ["fields", "hunter", "process-tests", "pytest-xdist", "six", "virtualenv"]
+
 [[package]]
 name = "python-dateutil"
 version = "2.8.2"
@@ -2537,6 +2644,20 @@ files = [
     {file = "wrapt-1.15.0.tar.gz", hash = "sha256:d06730c6aed78cee4126234cf2d071e01b44b915e725a6cb439a879ec9754a3a"},
 ]
 
+[[package]]
+name = "wsproto"
+version = "1.2.0"
+description = "WebSockets state-machine based protocol implementation"
+optional = false
+python-versions = ">=3.7.0"
+files = [
+    {file = "wsproto-1.2.0-py3-none-any.whl", hash = "sha256:b9acddd652b585d75b20477888c56642fdade28bdfd3579aa24a4d2c037dd736"},
+    {file = "wsproto-1.2.0.tar.gz", hash = "sha256:ad565f26ecb92588a3e43bc3d96164de84cd9902482b130d0ddbaa9664a85065"},
+]
+
+[package.dependencies]
+h11 = ">=0.9.0,<1"
+
 [[package]]
 name = "yarl"
 version = "1.9.2"
@@ -2627,4 +2748,4 @@ multidict = ">=4.0"
 [metadata]
 lock-version = "2.0"
 python-versions = "^3.11"
-content-hash = "a51d7d26b88683875685ede2298f0f02ab42b1f303657b47e0a5dee9be0dc9e6"
+content-hash = "c984979825947f67fc42e4553d5ff347f2f9194e4acccdeacb67a406a332009a"

server/pyproject.toml

@@ -35,9 +35,11 @@ stamina = "^23.1.0"
 
 
 [tool.poetry.group.tests.dependencies]
+pytest-cov = "^4.1.0"
 pytest-aiohttp = "^1.0.4"
 pytest-asyncio = "^0.21.1"
 pytest = "^7.4.0"
+httpx-ws = "^0.4.1"
 
 
 [tool.poetry.group.aws.dependencies]
@@ -46,3 +48,13 @@ aioboto3 = "^11.2.0"
 [build-system]
 requires = ["poetry-core"]
 build-backend = "poetry.core.masonry.api"
+
+[tool.coverage.run]
+source = ["reflector"]
+
+[tool.pytest.ini_options]
+addopts = "-ra -q --disable-pytest-warnings --cov --cov-report html -v"
+testpaths = ["tests"]
+asyncio_mode = "auto"
+
+

server/reflector/processors/types.py

@@ -1,9 +1,8 @@
-from dataclasses import dataclass
+from pydantic import BaseModel
 from pathlib import Path
 
 
-@dataclass
-class AudioFile:
+class AudioFile(BaseModel):
     path: Path
     sample_rate: int
     channels: int
@@ -14,15 +13,13 @@ class AudioFile:
         self.path.unlink()
 
 
-@dataclass
-class Word:
+class Word(BaseModel):
     text: str
     start: float
     end: float
 
 
-@dataclass
-class Transcript:
+class Transcript(BaseModel):
     text: str = ""
     words: list[Word] = None
 
@@ -59,8 +56,7 @@ class Transcript:
         return Transcript(text=self.text, words=words)
 
 
-@dataclass
-class TitleSummary:
+class TitleSummary(BaseModel):
     title: str
     summary: str
     timestamp: float
@@ -75,7 +71,6 @@ class TitleSummary:
         return f"{minutes:02d}:{seconds:02d}.{milliseconds:03d}"
 
 
-@dataclass
-class FinalSummary:
+class FinalSummary(BaseModel):
     summary: str
     duration: float

server/reflector/stream_client.py

@@ -3,7 +3,6 @@ import time
 import uuid
 
 import httpx
-import pyaudio
 import stamina
 from aiortc import RTCPeerConnection, RTCSessionDescription
 from aiortc.contrib.media import MediaPlayer, MediaRelay
@@ -24,7 +23,6 @@ class StreamClient:
         self.server_url = url
         self.play_from = play_from
         self.ping_pong = ping_pong
-        self.paudio = pyaudio.PyAudio()
 
         self.pc = RTCPeerConnection()
 
@@ -87,6 +85,7 @@ class StreamClient:
                 self.logger.info(f"Track {track.kind} ended")
 
         self.pc.addTrack(audio)
+        self.track_audio = audio
 
         channel = pc.createDataChannel("data-channel")
         self.logger = self.logger.bind(channel=channel.label)
@@ -142,3 +141,6 @@ class StreamClient:
         coro = self.run_offer(self.pc, self.signaling)
         task = asyncio.create_task(coro)
         await task
+
+    def is_ended(self):
+        return self.track_audio is None or self.track_audio.readyState == "ended"

server/reflector/views/transcripts.py

@@ -3,7 +3,8 @@ from pydantic import BaseModel, Field
 from uuid import UUID, uuid4
 from datetime import datetime
 from fastapi_pagination import Page, paginate
-from .rtc_offer import rtc_offer, RtcOffer, PipelineEvent
+from reflector.logger import logger
+from .rtc_offer import rtc_offer_base, RtcOffer, PipelineEvent
 import asyncio
 from typing import Optional
 
@@ -46,6 +47,7 @@ class Transcript(BaseModel):
 
     def add_event(self, event: str, data):
         self.events.append(TranscriptEvent(event=event, data=data))
+        return {"event": event, "data": data}
 
     def upsert_topic(self, topic: TranscriptTopic):
         existing_topic = next((t for t in self.topics if t.id == topic.id), None)
@@ -239,14 +241,37 @@ async def handle_rtc_event(event: PipelineEvent, args, data):
     # OFC the current implementation is not good,
     # but it's just a POC before persistence. It won't query the
     # transcript from the database for each event.
-    print(f"Event: {event}", args, data)
+    # print(f"Event: {event}", args, data)
     transcript_id = args
     transcript = transcripts_controller.get_by_id(transcript_id)
     if not transcript:
         return
-    transcript.add_event(event=event, data=data)
-    if event == PipelineEvent.TOPIC:
-        transcript.upsert_topic(TranscriptTopic(**data))
+
+    # event send to websocket clients may not be the same as the event
+    # received from the pipeline. For example, the pipeline will send
+    # a TRANSCRIPT event with all words, but this is not what we want
+    # to send to the websocket client.
+
+    # FIXME don't do copy
+    if event == PipelineEvent.TRANSCRIPT:
+        resp = transcript.add_event(event=event, data={
+            "text": data.text,
+        })
+    elif event == PipelineEvent.TOPIC:
+        topic = TranscriptTopic(
+            title=data.title,
+            summary=data.summary,
+            transcript=data.transcript,
+            timestamp=data.timestamp,
+        )
+        resp = transcript.add_event(event=event, data=topic.model_dump())
+        transcript.upsert_topic(topic)
+    else:
+        logger.warning(f"Unknown event: {event}")
+        return
+
+    # transmit to websocket clients
+    await ws_manager.send_json(transcript_id, resp)
 
 
 @router.post("/transcripts/{transcript_id}/record/webrtc")
@@ -261,9 +286,9 @@ async def transcript_record_webrtc(
         raise HTTPException(status_code=400, detail="Transcript is locked")
 
     # FIXME do not allow multiple recording at the same time
-    return await rtc_offer(
+    return await rtc_offer_base(
         params,
         request,
-        event_callback=transcript.handle_event,
+        event_callback=handle_rtc_event,
         event_callback_args=transcript_id,
     )

server/tests/test_transcripts.py

@@ -61,9 +61,9 @@ async def test_transcripts_list():
         assert "testxx1" in names
         assert "testxx2" in names
 
+
 @pytest.mark.asyncio
 async def test_transcript_delete():
-
     async with AsyncClient(app=app, base_url="http://test/v1") as ac:
         response = await ac.post("/transcripts", json={"name": "testdel1"})
         assert response.status_code == 200
@@ -76,4 +76,3 @@ async def test_transcript_delete():
 
         response = await ac.get(f"/transcripts/{tid}")
         assert response.status_code == 404
-

server/tests/test_transcripts_rtc_ws.py

@@ -0,0 +1,150 @@
+# === further tests
+# FIXME test status of transcript
+# FIXME test websocket connection after RTC is finished still send the full events
+# FIXME try with locked session, RTC should not work
+
+import pytest
+from unittest.mock import patch
+from httpx import AsyncClient
+
+from reflector.app import app
+from uvicorn import Config, Server
+import threading
+import asyncio
+from pathlib import Path
+from httpx_ws import aconnect_ws
+
+
+class ThreadedUvicorn:
+    def __init__(self, config: Config):
+        self.server = Server(config)
+        self.thread = threading.Thread(daemon=True, target=self.server.run)
+
+    async def start(self):
+        self.thread.start()
+        while not self.server.started:
+            await asyncio.sleep(0.1)
+
+    def stop(self):
+        if self.thread.is_alive():
+            self.server.should_exit = True
+            while self.thread.is_alive():
+                continue
+
+
+@pytest.fixture
+async def dummy_transcript():
+    from reflector.processors.audio_transcript import AudioTranscriptProcessor
+    from reflector.processors.types import AudioFile, Transcript, Word
+
+    class TestAudioTranscriptProcessor(AudioTranscriptProcessor):
+        async def _transcript(self, data: AudioFile):
+            return Transcript(
+                text="Hello world",
+                words=[
+                    Word(start=0.0, end=1.0, text="Hello"),
+                    Word(start=1.0, end=2.0, text="world"),
+                ],
+            )
+
+    with patch(
+        "reflector.processors.audio_transcript_auto"
+        ".AudioTranscriptAutoProcessor.get_instance"
+    ) as mock_audio:
+        mock_audio.return_value = TestAudioTranscriptProcessor()
+        yield
+
+
+@pytest.fixture
+async def dummy_llm():
+    from reflector.llm.base import LLM
+
+    class TestLLM(LLM):
+        async def _generate(self, prompt: str, **kwargs):
+            return {"text": "LLM RESULT"}
+
+    with patch("reflector.llm.base.LLM.get_instance") as mock_llm:
+        mock_llm.return_value = TestLLM()
+        yield
+
+
+@pytest.mark.asyncio
+async def test_transcript_rtc_and_websocket(dummy_transcript, dummy_llm):
+    # goal: start the server, exchange RTC, receive websocket events
+    # because of that, we need to start the server in a thread
+    # to be able to connect with aiortc
+
+    # start server
+    host = "127.0.0.1"
+    port = 1255
+    base_url = f"http://{host}:{port}/v1"
+    config = Config(app=app, host=host, port=port)
+    server = ThreadedUvicorn(config)
+    await server.start()
+
+    # create a transcript
+    ac = AsyncClient(base_url=base_url)
+    response = await ac.post("/transcripts", json={"name": "Test RTC"})
+    assert response.status_code == 200
+    tid = response.json()["id"]
+
+    # create a websocket connection as a task
+    events = []
+
+    async def websocket_task():
+        print("Test websocket: TASK STARTED")
+        async with aconnect_ws(f"{base_url}/transcripts/{tid}/events") as ws:
+            print("Test websocket: CONNECTED")
+            try:
+                while True:
+                    msg = await ws.receive_json()
+                    print(f"Test websocket: JSON {msg}")
+                    if msg is None:
+                        break
+                    events.append(msg)
+            except Exception as e:
+                print(f"Test websocket: EXCEPTION {e}")
+            finally:
+                ws.close()
+                print("Test websocket: DISCONNECTED")
+
+    websocket_task = asyncio.get_event_loop().create_task(websocket_task())
+
+    # create stream client
+    import argparse
+    from reflector.stream_client import StreamClient
+    from aiortc.contrib.signaling import add_signaling_arguments, create_signaling
+
+    parser = argparse.ArgumentParser()
+    add_signaling_arguments(parser)
+    args = parser.parse_args(["-s", "tcp-socket"])
+    signaling = create_signaling(args)
+
+    url = f"{base_url}/transcripts/{tid}/record/webrtc"
+    path = Path(__file__).parent / "records" / "test_short.wav"
+    client = StreamClient(signaling, url=url, play_from=path.as_posix())
+    await client.start()
+
+    timeout = 20
+    while not client.is_ended():
+        await asyncio.sleep(1)
+        timeout -= 1
+        if timeout < 0:
+            raise TimeoutError("Timeout while waiting for RTC to end")
+
+    await client.stop()
+
+    # wait the processing to finish
+    await asyncio.sleep(2)
+
+    # stop websocket task
+    websocket_task.cancel()
+
+    # check events
+    print(events)
+    assert len(events) > 0
+    assert events[0]["event"] == "TRANSCRIPT"
+    assert events[0]["data"]["text"] == "Hello world"
+
+    # stop server
+    # server.stop()