Mercurial > libervia-backend

annotate tests/unit/frontends/test_webrtc.py @ 4182:4dc00e848961

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core (tools/trigger): new `add_with_check` method: this method add a wrapper which will skip the trigger is the plugin is not available in the session (e.g. a component which doesn't use the plugin).
author Goffi <goffi@goffi.org>
date Sat, 09 Dec 2023 19:19:26 +0100
parents 13dd5660c28f
children dfccc90cacc6
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
rev   line source
4140
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1 #!/usr/bin/env python3
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2
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3 # Libervia: an XMPP client
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4 # Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org)
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5
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6 # This program is free software: you can redistribute it and/or modify
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7 # it under the terms of the GNU Affero General Public License as published by
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8 # the Free Software Foundation, either version 3 of the License, or
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9 # (at your option) any later version.
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10
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11 # This program is distributed in the hope that it will be useful,
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12 # but WITHOUT ANY WARRANTY; without even the implied warranty of
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13 # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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14 # GNU Affero General Public License for more details.
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15
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16 # You should have received a copy of the GNU Affero General Public License
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17 # along with this program. If not, see <http://www.gnu.org/licenses/>.
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18 from unittest.mock import AsyncMock, MagicMock
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19
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20 from gi.repository import Gst
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21 import pytest
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22
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23 from libervia.backend.core import exceptions
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24 from libervia.backend.tools.common import data_format
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25 from libervia.frontends.tools import webrtc as webrtc_mod
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26
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27
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28
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29 @pytest.fixture
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30 def host(monkeypatch):
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31 host = MagicMock()
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32 host.bridge = AsyncMock()
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33 host.app.expand = lambda s: s
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34 return host
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35
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36 @pytest.fixture(scope="function")
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37 def webrtc(host):
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38 """Fixture for WebRTC instantiation."""
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39 profile = "test_profile"
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40 instance = webrtc_mod.WebRTC(host.bridge, profile)
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41
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42 instance._set_media_types = MagicMock()
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43 instance.start_pipeline = MagicMock()
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44 instance.link_element_or_pad = MagicMock()
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45 instance.webrtcbin = MagicMock()
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46 instance.webrtcbin.emit = MagicMock()
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47
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48 instance.GstSdp_SDPMessage_new_from_text = MagicMock()
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49 instance.GstWebRTC_WebRTCSessionDescription_new = MagicMock()
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50 instance.Gst_Promise_new_with_change_func = MagicMock()
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51
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52 return instance
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53
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54
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55 class TestWebRtc:
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56 def test_get_payload_types(self, webrtc):
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57 """The method can identify the correct payload types for video and audio."""
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58 fake_sdpmsg = MagicMock()
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59 fake_media = MagicMock()
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60 fake_caps = MagicMock()
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61 fake_structure = MagicMock()
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62
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63 # This side effect will return 'fake_video_encoding' first, then
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64 # 'fake_audio_encoding'.
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65 fake_structure.__getitem__.side_effect = [
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66 "fake_video_encoding",
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67 "fake_audio_encoding",
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68 ]
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69 fake_caps.get_structure.return_value = fake_structure
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70 fake_media.get_format.side_effect = ["webrtc-datachannel", "10", "20"]
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71 fake_media.get_caps_from_media.return_value = fake_caps
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72 fake_sdpmsg.get_media.return_value = fake_media
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73 fake_sdpmsg.medias_len.return_value = 1
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74 fake_media.formats_len.return_value = 3
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75
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76 result = webrtc.get_payload_types(
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77 fake_sdpmsg, "fake_video_encoding", "fake_audio_encoding"
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78 )
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79 expected_result = {"fake_video_encoding": 10, "fake_audio_encoding": 20}
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80
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81 assert result == expected_result
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82
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83 def test_on_accepted_call(self, webrtc):
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84 """The method correctly sets the remote SDP upon acceptance of an outgoing call."""
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85 sdp_str = "mock_sdp_string"
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86 profile_str = "test_profile"
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87
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88 webrtc.on_accepted_call(sdp_str, profile_str)
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89
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90 # remote description must be set
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91 assert webrtc.webrtcbin.emit.call_count == 1
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92 assert webrtc.webrtcbin.emit.call_args[0][0] == "set-remote-description"
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93
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94 @pytest.mark.asyncio
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95 async def test_answer_call(self, webrtc, monkeypatch):
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96 """The method correctly answers an incoming call."""
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97 mock_setup_call = AsyncMock()
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98
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99 def mock_get_payload_types(sdpmsg, video_encoding, audio_encoding):
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100 return {"VP8": 96, "OPUS": 97}
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101
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102 monkeypatch.setattr(webrtc, "setup_call", mock_setup_call)
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103 monkeypatch.setattr(webrtc, "get_payload_types", mock_get_payload_types)
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104
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105 sdp_str = "mock_sdp_string"
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106 profile_str = "mock_profile"
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107
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108 await webrtc.answer_call(sdp_str, profile_str)
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109
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110 mock_setup_call.assert_called_once_with("responder", audio_pt=97, video_pt=96)
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111
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112 # remote description must be set
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113 assert webrtc.webrtcbin.emit.call_count == 1
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114 assert webrtc.webrtcbin.emit.call_args[0][0] == "set-remote-description"
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115
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116 def test_on_remote_decodebin_stream_video(self, webrtc, monkeypatch):
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117 """The method correctly handles video streams from the remote decodebin."""
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118 mock_pipeline = MagicMock()
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119 monkeypatch.setattr(webrtc, "pipeline", mock_pipeline)
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120
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121 mock_pad = MagicMock()
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122 mock_caps = MagicMock()
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123 mock_structure = MagicMock()
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124
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125 mock_pad.has_current_caps.return_value = True
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126 mock_pad.get_current_caps.return_value = mock_caps
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127 mock_caps.__len__.return_value = 1
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128 mock_caps.__getitem__.return_value = mock_structure
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129 mock_structure.get_name.return_value = "video/x-h264"
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130 # We use non-standard resolution as example to trigger the workaround
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131 mock_structure.get_int.side_effect = lambda x: MagicMock(
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132 value=990 if x == "width" else 557
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133 )
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134
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135 webrtc.on_remote_decodebin_stream(None, mock_pad)
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136
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137 assert webrtc._remote_video_pad == mock_pad
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138 mock_pipeline.add.assert_called()
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139 mock_pad.link.assert_called()
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140
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141 def test_on_remote_decodebin_stream_audio(self, webrtc, monkeypatch):
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142 """The method correctly handles audio streams from the remote decodebin."""
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143 mock_pipeline = MagicMock()
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144 monkeypatch.setattr(webrtc, "pipeline", mock_pipeline)
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145
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146 mock_pad = MagicMock()
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147 mock_caps = MagicMock()
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148 mock_structure = MagicMock()
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149
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150 mock_pad.has_current_caps.return_value = True
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151 mock_pad.get_current_caps.return_value = mock_caps
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152 mock_caps.__len__.return_value = 1
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153 mock_caps.__getitem__.return_value = mock_structure
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154 mock_structure.get_name.return_value = "audio/x-raw"
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155
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156 webrtc.on_remote_decodebin_stream(None, mock_pad)
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157
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158 mock_pipeline.add.assert_called()
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159 mock_pad.link.assert_called()
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160
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161 @pytest.mark.asyncio
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162 async def test_setup_call_correct_role(self, host, webrtc, monkeypatch):
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163 """Roles are set in setup_call."""
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164 monkeypatch.setattr(Gst, "parse_launch", MagicMock())
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165 monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
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166
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167 await webrtc.setup_call("initiator")
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168 assert webrtc.role == "initiator"
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169
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170 await webrtc.setup_call("responder")
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171 assert webrtc.role == "responder"
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172
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173 with pytest.raises(AssertionError):
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174 await webrtc.setup_call("invalid_role")
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175
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176 @pytest.mark.asyncio
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177 async def test_setup_call_test_mode(self, host, webrtc, monkeypatch):
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178 """Test mode use fake video and audio in setup_call."""
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179 monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
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180 monkeypatch.setattr(webrtc, "sources", webrtc_mod.SINKS_TEST)
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181 await webrtc.setup_call("initiator")
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182 assert "videotestsrc" in webrtc.gst_pipe_desc
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183 assert "audiotestsrc" in webrtc.gst_pipe_desc
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184
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185 @pytest.mark.asyncio
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186 async def test_setup_call_normal_mode(self, host, webrtc, monkeypatch):
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187 """Normal mode use real video and audio in setup_call."""
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188 monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
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189 monkeypatch.setattr(webrtc, "sources", webrtc_mod.SOURCES_AUTO)
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190 await webrtc.setup_call("initiator")
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191 assert "v4l2src" in webrtc.gst_pipe_desc
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192 assert "pulsesrc" in webrtc.gst_pipe_desc
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193
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194 @pytest.mark.asyncio
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195 async def test_setup_call_with_stun_and_turn(self, host, webrtc, monkeypatch):
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196 """STUN and TURN server configurations are done in setup_call."""
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197 mock_pipeline = MagicMock()
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198 mock_parse_launch = MagicMock()
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199 mock_parse_launch.return_value = mock_pipeline
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200 monkeypatch.setattr(Gst, "parse_launch", mock_parse_launch)
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201
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202 mock_pipeline.get_by_name.return_value = webrtc.webrtcbin
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203
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204 mock_external_disco = [
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205 {"type": "stun", "transport": "udp", "host": "stun.host", "port": "3478"},
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206 {
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207 "type": "turn",
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208 "transport": "udp",
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209 "host": "turn.host",
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210 "port": "3478",
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211 "username": "user",
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212 "password": "pass",
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213 },
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214 ]
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215
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216 monkeypatch.setattr(
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217 data_format, "deserialise", MagicMock(return_value=mock_external_disco)
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218 )
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219
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220 mock_emit = AsyncMock()
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221 monkeypatch.setattr(webrtc.webrtcbin, "emit", mock_emit)
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222
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223 mock_set_property = AsyncMock()
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224 monkeypatch.setattr(webrtc.webrtcbin, "set_property", mock_set_property)
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225
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226 await webrtc.setup_call("initiator")
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227
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228 host.bridge.external_disco_get.assert_called_once_with("", webrtc.profile)
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229 mock_set_property.assert_any_call("stun-server", "stun://stun.host:3478")
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230 mock_emit.assert_called_once_with(
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231 "add-turn-server", "turn://user:pass@turn.host:3478"
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232 )
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233
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234 @pytest.mark.asyncio
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235 async def test_setup_call_gstreamer_pipeline_failure(self, webrtc, monkeypatch):
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236 """Test setup_call method handling Gstreamer pipeline failure."""
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237 monkeypatch.setattr(Gst, "parse_launch", lambda _: None)
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238 with pytest.raises(exceptions.InternalError):
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239 await webrtc.setup_call("initiator")