IQ.Pilot Prebuilt Release @ 27f668a

iqpilot/system/webrtc/tests/test_rtc_integration.py

@@ -0,0 +1,95 @@
+import asyncio
+import os
+
+import pytest
+
+from iqpilot.system.webrtc.rtc import WebRTCOfferBuilder
+from iqpilot.system.webrtc.rtc.stream import RTCSessionDescription
+from iqpilot.system.webrtc.session import StreamSession
+
+
+@pytest.mark.asyncio
+async def test_native_video_audio_and_data_channel():
+  if not os.environ.get("CI"):
+    return
+
+  answer_session = None
+  video_received = asyncio.Event()
+  audio_received = asyncio.Event()
+
+  async def connect(offer):
+    nonlocal answer_session
+    browser_offer = offer.sdp.replace("profile-level-id=42e01f", "profile-level-id=640c1f", 1)
+    audio_offset = browser_offer.index("m=audio")
+    browser_offer = browser_offer[:audio_offset] + browser_offer[audio_offset:].replace(
+      "a=recvonly",
+      "a=sendrecv\r\na=msid:ios-microphone ios-audio-track\r\na=ssrc:123456 cname:ios-audio\r\na=ssrc:123456 msid:ios-microphone ios-audio-track",
+      1,
+    )
+    answer_session = StreamSession(browser_offer, offer.video, [], [], [], debug_mode=True)
+    answer = await answer_session.get_answer()
+    assert not any(line.startswith("m=video 0 ") for line in answer.sdp.splitlines())
+    assert answer_session.stream._track_state[0][0] is answer_session.stream._negotiated_tracks["road"]
+    answer_session.start()
+    return RTCSessionDescription(answer.sdp, answer.type)
+
+  builder = WebRTCOfferBuilder(connect, ice_servers=[])
+  builder.offer_to_receive_video_stream("road")
+  builder.offer_to_receive_audio_stream()
+  builder.add_messaging()
+  stream = builder.stream()
+
+  try:
+    await asyncio.wait_for(stream.start(), 10)
+    stream.get_incoming_video_track("road").on_frame(lambda *_: video_received.set())
+    stream.get_incoming_audio_track().on_frame(lambda *_: audio_received.set())
+    await asyncio.wait_for(stream.wait_for_connection(), 10)
+    await asyncio.wait_for(video_received.wait(), 10)
+    await asyncio.wait_for(audio_received.wait(), 10)
+    stream.get_messaging_channel().send('{"type":"timingSei","enabled":true}')
+    await asyncio.sleep(0.1)
+    assert answer_session is not None
+    assert answer_session.video_tracks[0].timing_sei_enabled
+  finally:
+    await stream.stop()
+    if answer_session is not None:
+      await answer_session.stop_async()
+
+
+@pytest.mark.asyncio
+async def test_native_duplex_audio_negotiation():
+  if not os.environ.get("CI"):
+    return
+
+  answer_session = None
+
+  async def connect(offer):
+    nonlocal answer_session
+    sendrecv_offer = offer.sdp.replace(
+      "a=recvonly",
+      "a=sendrecv\r\na=msid:ios-microphone ios-audio-track\r\na=ssrc:123456 cname:ios-audio\r\na=ssrc:123456 msid:ios-microphone ios-audio-track",
+      1,
+    )
+    answer_session = StreamSession(sendrecv_offer, offer.video, [], [], [], debug_mode=True)
+    answer = await answer_session.get_answer()
+    assert "ios-microphone" not in answer.sdp
+    assert "ios-audio-track" not in answer.sdp
+    assert answer.sdp.count("a=msid:audio audio") == 1
+    answer_session.start()
+    return RTCSessionDescription(answer.sdp, answer.type)
+
+  builder = WebRTCOfferBuilder(connect, ice_servers=[])
+  builder.offer_to_receive_audio_stream()
+  stream = builder.stream()
+
+  try:
+    await asyncio.wait_for(stream.start(), 10)
+    await asyncio.wait_for(stream.wait_for_connection(), 10)
+    assert stream.has_incoming_audio_track()
+    assert answer_session is not None
+    assert answer_session.audio_recv_requested
+    assert answer_session.audio_output is not None
+  finally:
+    await stream.stop()
+    if answer_session is not None:
+      await answer_session.stop_async()