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doc (components): Update Email Gateway documentation: A section has been added to explain how attachments are handled. fix 453
author Goffi <goffi@goffi.org>
date Tue, 03 Dec 2024 00:53:18 +0100
parents f1d0cde61af7
children
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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 import pytest
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21
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22 try:
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23 from gi.repository import Gst
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24 except ImportError:
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25 pytest.skip("Gst not available.", allow_module_level=True)
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26
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27 from libervia.backend.core import exceptions
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28 from libervia.backend.tools.common import data_format
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29 from libervia.frontends.tools import webrtc as webrtc_mod
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30
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31
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32 @pytest.fixture
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33 def host(monkeypatch):
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34 host = MagicMock()
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35 host.bridge = AsyncMock()
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36 host.app.expand = lambda s: s
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37 return host
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38
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39
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40 @pytest.fixture(scope="function")
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41 def webrtc(host):
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42 """Fixture for WebRTC instantiation."""
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43 profile = "test_profile"
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44 instance = webrtc_mod.WebRTC(host.bridge, profile)
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45
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46 instance._set_media_types = MagicMock()
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47 instance.start_pipeline = MagicMock()
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48 instance.link_element_or_pad = MagicMock()
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49 instance.webrtcbin = MagicMock()
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50 instance.webrtcbin.emit = MagicMock()
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51
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52 instance.GstSdp_SDPMessage_new_from_text = MagicMock()
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53 instance.GstWebRTC_WebRTCSessionDescription_new = MagicMock()
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54 instance.Gst_Promise_new_with_change_func = MagicMock()
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55
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56 return instance
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57
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58
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59 class TestWebRtc:
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60 def test_get_payload_types(self, webrtc):
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61 """The method can identify the correct payload types for video and audio."""
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62 fake_sdpmsg = MagicMock()
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63 fake_media = MagicMock()
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64 fake_caps = MagicMock()
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65 fake_structure = MagicMock()
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66
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67 # This side effect will return 'fake_video_encoding' first, then
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68 # 'fake_audio_encoding'.
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69 fake_structure.__getitem__.side_effect = [
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70 "fake_video_encoding",
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71 "fake_audio_encoding",
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72 ]
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73 fake_caps.get_structure.return_value = fake_structure
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74 fake_media.get_format.side_effect = ["webrtc-datachannel", "10", "20"]
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75 fake_media.get_caps_from_media.return_value = fake_caps
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76 fake_sdpmsg.get_media.return_value = fake_media
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77 fake_sdpmsg.medias_len.return_value = 1
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78 fake_media.formats_len.return_value = 3
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79
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80 result = webrtc.get_payload_types(
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81 fake_sdpmsg, "fake_video_encoding", "fake_audio_encoding"
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82 )
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83 expected_result = {"fake_video_encoding": 10, "fake_audio_encoding": 20}
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84
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85 assert result == expected_result
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86
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87 def test_on_accepted_call(self, webrtc):
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88 """The method correctly sets the remote SDP upon acceptance of an outgoing call."""
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89 sdp_str = "mock_sdp_string"
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90 profile_str = "test_profile"
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91
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92 webrtc.on_accepted_call(sdp_str, profile_str)
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93
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94 # remote description must be set
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95 assert webrtc.webrtcbin.emit.call_count == 1
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96 assert webrtc.webrtcbin.emit.call_args[0][0] == "set-remote-description"
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97
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98 @pytest.mark.asyncio
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99 async def test_answer_call(self, webrtc, monkeypatch):
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100 """The method correctly answers an incoming call."""
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101 mock_setup_call = AsyncMock()
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102
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103 def mock_get_payload_types(sdpmsg, video_encoding, audio_encoding):
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104 return {"VP8": 96, "OPUS": 97}
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105
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106 monkeypatch.setattr(webrtc, "setup_call", mock_setup_call)
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107 monkeypatch.setattr(webrtc, "get_payload_types", mock_get_payload_types)
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108
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109 sdp_str = "mock_sdp_string"
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110 profile_str = "mock_profile"
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111
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112 await webrtc.answer_call(sdp_str, profile_str)
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113
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114 mock_setup_call.assert_called_once_with("responder", audio_pt=97, video_pt=96)
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115
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116 # remote description must be set
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117 assert webrtc.webrtcbin.emit.call_count == 1
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118 assert webrtc.webrtcbin.emit.call_args[0][0] == "set-remote-description"
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119
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120 def test_on_remote_decodebin_stream_video(self, webrtc, monkeypatch):
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121 """The method correctly handles video streams from the remote decodebin."""
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122 mock_pipeline = MagicMock()
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123 monkeypatch.setattr(webrtc, "pipeline", mock_pipeline)
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124
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125 mock_pad = MagicMock()
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126 mock_caps = MagicMock()
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127 mock_structure = MagicMock()
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128
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129 mock_pad.has_current_caps.return_value = True
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130 mock_pad.get_current_caps.return_value = mock_caps
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131 mock_caps.__len__.return_value = 1
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132 mock_caps.__getitem__.return_value = mock_structure
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133 mock_structure.get_name.return_value = "video/x-h264"
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134 # We use non-standard resolution as example to trigger the workaround
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135 mock_structure.get_int.side_effect = lambda x: MagicMock(
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136 value=990 if x == "width" else 557
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137 )
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138
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139 webrtc.on_remote_decodebin_stream(None, mock_pad)
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140
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141 assert webrtc._remote_video_pad == mock_pad
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142 mock_pipeline.add.assert_called()
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143 mock_pad.link.assert_called()
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144
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145 def test_on_remote_decodebin_stream_audio(self, webrtc, monkeypatch):
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146 """The method correctly handles audio streams from the remote decodebin."""
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147 mock_pipeline = MagicMock()
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148 monkeypatch.setattr(webrtc, "pipeline", mock_pipeline)
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149
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150 mock_pad = MagicMock()
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151 mock_caps = MagicMock()
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152 mock_structure = MagicMock()
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153
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154 mock_pad.has_current_caps.return_value = True
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155 mock_pad.get_current_caps.return_value = mock_caps
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156 mock_caps.__len__.return_value = 1
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157 mock_caps.__getitem__.return_value = mock_structure
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158 mock_structure.get_name.return_value = "audio/x-raw"
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159
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160 webrtc.on_remote_decodebin_stream(None, mock_pad)
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161
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162 mock_pipeline.add.assert_called()
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163 mock_pad.link.assert_called()
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164
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165 @pytest.mark.skipif(Gst is None, reason="GStreamer is not available")
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166 @pytest.mark.asyncio
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167 async def test_setup_call_correct_role(self, host, webrtc, monkeypatch):
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168 """Roles are set in setup_call."""
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169 monkeypatch.setattr(Gst, "parse_launch", MagicMock())
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170 # we use MagicMock class and not instance on purpose, to pass the "isinstance"
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171 # test of "setup_call".
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172 monkeypatch.setattr(Gst, "Pipeline", MagicMock)
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173 monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
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174
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175 await webrtc.setup_call("initiator")
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176 assert webrtc.role == "initiator"
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177
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178 await webrtc.setup_call("responder")
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179 assert webrtc.role == "responder"
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180
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181 with pytest.raises(AssertionError):
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182 await webrtc.setup_call("invalid_role")
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183
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184 @pytest.mark.asyncio
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185 async def test_setup_call_test_mode(self, host, webrtc, monkeypatch):
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186 """Test mode use fake video and audio in setup_call."""
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187 monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
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188 monkeypatch.setattr(webrtc, "sources_data", webrtc_mod.SourcesTest())
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189 await webrtc.setup_call("initiator")
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190 assert "videotestsrc" in webrtc.gst_pipe_desc
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191 assert "audiotestsrc" in webrtc.gst_pipe_desc
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192
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193 @pytest.mark.asyncio
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194 async def test_setup_call_normal_mode(self, host, webrtc, monkeypatch):
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195 """Normal mode use real video and audio in setup_call."""
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196 monkeypatch.setattr(data_format, "deserialise", MagicMock(return_value=[]))
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197 monkeypatch.setattr(webrtc, "sources_data", webrtc_mod.SourcesAuto())
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198 await webrtc.setup_call("initiator")
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199 assert "v4l2src" in webrtc.gst_pipe_desc
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200 assert "pulsesrc" in webrtc.gst_pipe_desc
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201
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202 @pytest.mark.skipif(Gst is None, reason="GStreamer is not available")
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203 @pytest.mark.asyncio
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204 async def test_setup_call_with_stun_and_turn(self, host, webrtc, monkeypatch):
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205 """STUN and TURN server configurations are done in setup_call."""
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206 mock_pipeline = MagicMock()
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207 mock_parse_launch = MagicMock()
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208 mock_parse_launch.return_value = mock_pipeline
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209 # As for "test_setup_call_correct_role" we user MagicMock class and not instance
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210 # on purpose here.
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211 monkeypatch.setattr(Gst, "Pipeline", MagicMock)
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212 monkeypatch.setattr(Gst, "parse_launch", mock_parse_launch)
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213
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214 mock_pipeline.get_by_name.return_value = webrtc.webrtcbin
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215
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216 mock_external_disco = [
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217 {"type": "stun", "transport": "udp", "host": "stun.host", "port": "3478"},
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218 {
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219 "type": "turn",
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220 "transport": "udp",
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221 "host": "turn.host",
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222 "port": "3478",
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223 "username": "user",
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224 "password": "pass",
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225 },
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226 ]
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227
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228 monkeypatch.setattr(
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229 data_format, "deserialise", MagicMock(return_value=mock_external_disco)
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230 )
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231
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232 mock_emit = AsyncMock()
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233 monkeypatch.setattr(webrtc.webrtcbin, "emit", mock_emit)
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234
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235 mock_set_property = AsyncMock()
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236 monkeypatch.setattr(webrtc.webrtcbin, "set_property", mock_set_property)
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237
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238 await webrtc.setup_call("initiator")
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239
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240 host.bridge.external_disco_get.assert_called_once_with("", webrtc.profile)
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241 mock_set_property.assert_any_call("stun-server", "stun://stun.host:3478")
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242 mock_emit.assert_called_once_with(
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243 "add-turn-server", "turn://user:pass@turn.host:3478"
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244 )
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245
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246 @pytest.mark.skipif(Gst is None, reason="GStreamer is not available")
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247 @pytest.mark.asyncio
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248 async def test_setup_call_gstreamer_pipeline_failure(self, webrtc, monkeypatch):
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249 """Test setup_call method handling Gstreamer pipeline failure."""
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250 monkeypatch.setattr(Gst, "parse_launch", lambda _: None)
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251 with pytest.raises(exceptions.InternalError):
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252 await webrtc.setup_call("initiator")