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