tests (unit/frontends): tests for webrtc implementation:

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Goffi 2023-11-01 14:04:25 +01:00
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#!/usr/bin/env python3
# Libervia: an XMPP client
# Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org)
# This program is free software: you can redistribute it and/or modify
# it under the terms of the GNU Affero General Public License as published by
# the Free Software Foundation, either version 3 of the License, or
# (at your option) any later version.
# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU Affero General Public License for more details.
# You should have received a copy of the GNU Affero General Public License
# along with this program. If not, see <http://www.gnu.org/licenses/>.
from unittest.mock import AsyncMock, MagicMock
from gi.repository import Gst
import pytest
from libervia.backend.core import exceptions
from libervia.backend.tools.common import data_format
from libervia.frontends.tools import webrtc as webrtc_mod
@pytest.fixture
def host(monkeypatch):
host = MagicMock()
host.bridge = AsyncMock()
host.app.expand = lambda s: s
return host
@pytest.fixture(scope="function")
def webrtc(host):
"""Fixture for WebRTC instantiation."""
profile = "test_profile"
instance = webrtc_mod.WebRTC(host.bridge, profile)
instance._set_media_types = MagicMock()
instance.start_pipeline = MagicMock()
instance.link_element_or_pad = MagicMock()
instance.webrtcbin = MagicMock()
instance.webrtcbin.emit = MagicMock()
instance.GstSdp_SDPMessage_new_from_text = MagicMock()
instance.GstWebRTC_WebRTCSessionDescription_new = MagicMock()
instance.Gst_Promise_new_with_change_func = MagicMock()
return instance
class TestWebRtc:
def test_get_payload_types(self, webrtc):
"""The method can identify the correct payload types for video and audio."""
fake_sdpmsg = MagicMock()
fake_media = MagicMock()
fake_caps = MagicMock()
fake_structure = MagicMock()
# This side effect will return 'fake_video_encoding' first, then
# 'fake_audio_encoding'.
fake_structure.__getitem__.side_effect = [
"fake_video_encoding",
"fake_audio_encoding",
]
fake_caps.get_structure.return_value = fake_structure
fake_media.get_format.side_effect = ["webrtc-datachannel", "10", "20"]
fake_media.get_caps_from_media.return_value = fake_caps
fake_sdpmsg.get_media.return_value = fake_media
fake_sdpmsg.medias_len.return_value = 1
fake_media.formats_len.return_value = 3
result = webrtc.get_payload_types(
fake_sdpmsg, "fake_video_encoding", "fake_audio_encoding"
)
expected_result = {"fake_video_encoding": 10, "fake_audio_encoding": 20}
assert result == expected_result
def test_on_accepted_call(self, webrtc):
"""The method correctly sets the remote SDP upon acceptance of an outgoing call."""
sdp_str = "mock_sdp_string"
profile_str = "test_profile"
webrtc.on_accepted_call(sdp_str, profile_str)
# remote description must be set
assert webrtc.webrtcbin.emit.call_count == 1
assert webrtc.webrtcbin.emit.call_args[0][0] == "set-remote-description"
@pytest.mark.asyncio
async def test_answer_call(self, webrtc, monkeypatch):
"""The method correctly answers an incoming call."""
mock_setup_call = AsyncMock()
def mock_get_payload_types(sdpmsg, video_encoding, audio_encoding):
return {"VP8": 96, "OPUS": 97}
monkeypatch.setattr(webrtc, "setup_call", mock_setup_call)
monkeypatch.setattr(webrtc, "get_payload_types", mock_get_payload_types)
sdp_str = "mock_sdp_string"
profile_str = "mock_profile"
await webrtc.answer_call(sdp_str, profile_str)
mock_setup_call.assert_called_once_with("responder", audio_pt=97, video_pt=96)
# remote description must be set
assert webrtc.webrtcbin.emit.call_count == 1
assert webrtc.webrtcbin.emit.call_args[0][0] == "set-remote-description"
def test_on_remote_decodebin_stream_video(self, webrtc, monkeypatch):
"""The method correctly handles video streams from the remote decodebin."""
mock_pipeline = MagicMock()
monkeypatch.setattr(webrtc, "pipeline", mock_pipeline)
mock_pad = MagicMock()
mock_caps = MagicMock()
mock_structure = MagicMock()
mock_pad.has_current_caps.return_value = True
mock_pad.get_current_caps.return_value = mock_caps
mock_caps.__len__.return_value = 1
mock_caps.__getitem__.return_value = mock_structure
mock_structure.get_name.return_value = "video/x-h264"
# We use non-standard resolution as example to trigger the workaround
mock_structure.get_int.side_effect = lambda x: MagicMock(
value=990 if x == "width" else 557
)
webrtc.on_remote_decodebin_stream(None, mock_pad)
assert webrtc._remote_video_pad == mock_pad
mock_pipeline.add.assert_called()
mock_pad.link.assert_called()
def test_on_remote_decodebin_stream_audio(self, webrtc, monkeypatch):
"""The method correctly handles audio streams from the remote decodebin."""
mock_pipeline = MagicMock()
monkeypatch.setattr(webrtc, "pipeline", mock_pipeline)
mock_pad = MagicMock()
mock_caps = MagicMock()
mock_structure = MagicMock()
mock_pad.has_current_caps.return_value = True
mock_pad.get_current_caps.return_value = mock_caps
mock_caps.__len__.return_value = 1
mock_caps.__getitem__.return_value = mock_structure
mock_structure.get_name.return_value = "audio/x-raw"
webrtc.on_remote_decodebin_stream(None, mock_pad)
mock_pipeline.add.assert_called()
mock_pad.link.assert_called()
@pytest.mark.asyncio
async def test_setup_call_correct_role(self, host, webrtc, monkeypatch):
"""Roles are set in setup_call."""
monkeypatch.setattr(Gst, "parse_launch", MagicMock())
monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
await webrtc.setup_call("initiator")
assert webrtc.role == "initiator"
await webrtc.setup_call("responder")
assert webrtc.role == "responder"
with pytest.raises(AssertionError):
await webrtc.setup_call("invalid_role")
@pytest.mark.asyncio
async def test_setup_call_test_mode(self, host, webrtc, monkeypatch):
"""Test mode use fake video and audio in setup_call."""
monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
monkeypatch.setattr(webrtc, "sources", webrtc_mod.SINKS_TEST)
await webrtc.setup_call("initiator")
assert "videotestsrc" in webrtc.gst_pipe_desc
assert "audiotestsrc" in webrtc.gst_pipe_desc
@pytest.mark.asyncio
async def test_setup_call_normal_mode(self, host, webrtc, monkeypatch):
"""Normal mode use real video and audio in setup_call."""
monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
monkeypatch.setattr(webrtc, "sources", webrtc_mod.SOURCES_AUTO)
await webrtc.setup_call("initiator")
assert "v4l2src" in webrtc.gst_pipe_desc
assert "pulsesrc" in webrtc.gst_pipe_desc
@pytest.mark.asyncio
async def test_setup_call_with_stun_and_turn(self, host, webrtc, monkeypatch):
"""STUN and TURN server configurations are done in setup_call."""
mock_pipeline = MagicMock()
mock_parse_launch = MagicMock()
mock_parse_launch.return_value = mock_pipeline
monkeypatch.setattr(Gst, "parse_launch", mock_parse_launch)
mock_pipeline.get_by_name.return_value = webrtc.webrtcbin
mock_external_disco = [
{"type": "stun", "transport": "udp", "host": "stun.host", "port": "3478"},
{
"type": "turn",
"transport": "udp",
"host": "turn.host",
"port": "3478",
"username": "user",
"password": "pass",
},
]
monkeypatch.setattr(
data_format, "deserialise", MagicMock(return_value=mock_external_disco)
)
mock_emit = AsyncMock()
monkeypatch.setattr(webrtc.webrtcbin, "emit", mock_emit)
mock_set_property = AsyncMock()
monkeypatch.setattr(webrtc.webrtcbin, "set_property", mock_set_property)
await webrtc.setup_call("initiator")
host.bridge.external_disco_get.assert_called_once_with("", webrtc.profile)
mock_set_property.assert_any_call("stun-server", "stun://stun.host:3478")
mock_emit.assert_called_once_with(
"add-turn-server", "turn://user:pass@turn.host:3478"
)
@pytest.mark.asyncio
async def test_setup_call_gstreamer_pipeline_failure(self, webrtc, monkeypatch):
"""Test setup_call method handling Gstreamer pipeline failure."""
monkeypatch.setattr(Gst, "parse_launch", lambda _: None)
with pytest.raises(exceptions.InternalError):
await webrtc.setup_call("initiator")