Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc.py @ 4259:49019947cc76
component AP Gateway: implement HTTP GET signature.
author | Goffi <goffi@goffi.org> |
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date | Wed, 05 Jun 2024 22:34:09 +0200 |
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1 #!/usr/bin/env python3 |
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2 |
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3 # Libervia WebRTC implementation |
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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 |
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19 from collections.abc import Awaitable |
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20 import gi |
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21 |
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22 gi.require_versions({"Gst": "1.0", "GstWebRTC": "1.0"}) |
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23 from gi.repository import Gst, GstWebRTC, GstSdp |
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24 |
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25 from libervia.backend.core import exceptions |
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26 |
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27 try: |
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28 from gi.overrides import Gst as _ |
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29 except ImportError: |
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30 raise exceptions.MissingModule( |
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31 "No GStreamer Python overrides available. Please install relevant packages on " |
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32 "your system (e.g., `python3-gst-1.0` on Debian and derivatives)." |
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33 ) |
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34 import asyncio |
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35 from datetime import datetime |
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36 import logging |
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37 import re |
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38 from typing import Callable, Final |
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39 from urllib.parse import quote_plus |
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40 |
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41 from libervia.backend.tools.common import data_format |
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42 from libervia.frontends.tools import aio, display_servers, jid |
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43 from .webrtc_models import ( |
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44 CallData, |
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45 SinksApp, |
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46 SinksAuto, |
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47 SinksData, |
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48 SinksDataChannel, |
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49 SinksNone, |
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50 SourcesAuto, |
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51 SourcesData, |
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52 SourcesDataChannel, |
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53 SourcesNone, |
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54 SourcesPipeline, |
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55 SourcesTest, |
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56 ) |
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57 |
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58 current_server = display_servers.detect() |
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59 if current_server == display_servers.X11: |
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60 # GSTreamer's ximagesrc documentation asks to run this function |
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61 import ctypes |
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62 |
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63 ctypes.CDLL("libX11.so.6").XInitThreads() |
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64 |
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65 |
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66 log = logging.getLogger(__name__) |
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67 VIDEO_SOURCE_AUTO: Final = "v4l2src" |
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68 AUDIO_SOURCE_AUTO: Final = "pulsesrc" |
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69 NONE_NOT_IMPLEMENTED_MSG: Final = "None value is not handled yet." |
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70 |
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71 Gst.init(None) |
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72 |
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73 |
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74 class WebRTC: |
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75 """GSTreamer based WebRTC implementation for audio and video communication. |
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76 |
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77 This class encapsulates the WebRTC functionalities required for initiating and |
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78 handling audio and video calls, and data channels. |
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79 """ |
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80 |
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81 def __init__( |
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82 self, |
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83 bridge, |
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84 profile: str, |
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85 sources_data: SourcesData|None = None, |
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86 sinks_data: SinksData|None = None, |
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87 reset_cb: Callable | None = None, |
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88 merge_pip: bool | None = None, |
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89 target_size: tuple[int, int] | None = None, |
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90 call_start_cb: Callable[[str, dict, str], Awaitable[str]] | None = None, |
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91 dc_data_list: list[SourcesDataChannel|SinksDataChannel]|None = None |
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92 ) -> None: |
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93 """Initializes a new WebRTC instance. |
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94 |
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95 @param bridge: An instance of backend bridge. |
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96 @param profile: Libervia profile. |
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97 @param sources_data: Data of the sources. |
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98 The model used will determine which sources to use. |
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99 SourcesDataChannel can be used here as a convenience. It will then be moved |
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100 to ``data_channels`` and ``SourcesNone`` will be used instead for |
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101 ``sources_data``. |
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102 If None, SourcesAuto will be used. |
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103 @param sinks_data: Data of the sinks. |
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104 The model used will determine which sinks to use. |
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105 SinksDataChannel can be used here as a convenience. It will then be moved |
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106 to ``data_channels`` and ``SinksNone`` will be used instead for |
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107 ``sinks_data``. |
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108 If None, SinksAuto will be used. |
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109 @param reset_cb: An optional Callable that is triggered on reset events. Can be |
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110 used to reset UI data on new calls. |
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111 @param merge_pip: A boolean flag indicating whether Picture-in-Picture mode is |
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112 enabled. When PiP is used, local feedback is merged to remote video stream. |
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113 Only one video stream is then produced (the local one). |
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114 If None, PiP mode is selected automatically according to selected sink (it's |
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115 used for SinksAuto only for now). |
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116 @param target_size: Expected size of the final sink stream. Mainly use by composer |
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117 when ``merge_pip`` is set. |
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118 None to autodetect (no real autodetection implemeted yet, default to |
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119 (1280,720)). |
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120 @param call_start_cb: Called when call is started. |
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121 @param dc_data_list: Data Channels to create. |
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122 If a SourcesDataChannel is used as ``sources_data``, or a SinksDataChannel is |
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123 used as ``sinks_data``, they will be automatically added to this list. |
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124 """ |
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125 self.main_loop = asyncio.get_event_loop() |
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126 self.bridge = bridge |
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127 self.profile = profile |
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128 self.pipeline = None |
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129 self._audio_muted = False |
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130 self._video_muted = False |
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131 self._desktop_sharing = False |
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132 self.desktop_sharing_data = None |
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133 if dc_data_list is None: |
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134 dc_data_list = [] |
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135 self.dc_data_list = dc_data_list |
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136 if sources_data is None: |
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137 sources_data = SourcesAuto() |
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138 elif isinstance(sources_data, SourcesDataChannel): |
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139 dc_data_list.append(sources_data) |
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140 sources_data = SourcesNone() |
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141 self.sources_data = sources_data |
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142 if sinks_data is None: |
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143 sinks_data = SinksAuto() |
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144 elif isinstance(sinks_data, SinksDataChannel): |
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145 dc_data_list.append(sinks_data) |
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146 sinks_data = SinksNone() |
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147 self.sinks_data = sinks_data |
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148 if target_size is None: |
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149 target_size = (1280, 720) |
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150 self.target_width, self.target_height = target_size |
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151 if merge_pip is None: |
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152 merge_pip = isinstance(sinks_data, SinksAuto) |
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153 self.merge_pip = merge_pip |
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154 if call_start_cb is None: |
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155 call_start_cb = self._call_start |
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156 self.call_start_cb = call_start_cb |
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157 if isinstance(sinks_data, SinksApp): |
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158 if merge_pip and sinks_data.remote_video_cb is not None: |
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159 raise ValueError("Remote_video_cb can't be used when merge_pip is used!") |
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160 self.reset_cb = reset_cb |
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161 if current_server == display_servers.WAYLAND: |
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162 from .portal_desktop import DesktopPortal |
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163 |
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164 self.desktop_portal = DesktopPortal( |
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165 on_session_closed_cb=self.on_portal_session_closed |
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166 ) |
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167 else: |
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168 self.desktop_portal = None |
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169 self.reset_instance() |
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170 |
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171 @property |
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172 def audio_muted(self): |
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173 return self._audio_muted |
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174 |
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175 @audio_muted.setter |
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176 def audio_muted(self, muted: bool) -> None: |
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177 if muted != self._audio_muted: |
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178 self._audio_muted = muted |
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179 self.on_audio_mute(muted) |
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180 |
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181 @property |
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182 def video_muted(self): |
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183 return self._video_muted |
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184 |
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185 @video_muted.setter |
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186 def video_muted(self, muted: bool) -> None: |
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187 if muted != self._video_muted: |
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188 self._video_muted = muted |
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189 self.on_video_mute(muted) |
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190 |
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191 @property |
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192 def desktop_sharing(self): |
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193 return self._desktop_sharing |
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194 |
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195 @desktop_sharing.setter |
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196 def desktop_sharing(self, active: bool) -> None: |
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197 if active != self._desktop_sharing: |
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198 self._desktop_sharing = active |
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199 self.on_desktop_switch(active) |
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200 try: |
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201 cb = self.bindings["desktop_sharing"] |
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202 except KeyError: |
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203 pass |
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204 else: |
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205 cb(active) |
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206 |
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207 @property |
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208 def sdp_set(self): |
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209 return self._sdp_set |
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210 |
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211 @sdp_set.setter |
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212 def sdp_set(self, is_set: bool): |
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213 self._sdp_set = is_set |
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214 if is_set: |
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215 self.on_ice_candidates_new(self.remote_candidates_buffer) |
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216 for data in self.remote_candidates_buffer.values(): |
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217 data["candidates"].clear() |
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218 |
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219 @property |
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220 def media_types(self): |
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221 if self._media_types is None: |
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222 raise Exception("self._media_types should not be None!") |
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223 return self._media_types |
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224 |
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225 @media_types.setter |
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226 def media_types(self, new_media_types: dict) -> None: |
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227 self._media_types = new_media_types |
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228 self._media_types_inv = {v: k for k, v in new_media_types.items()} |
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229 |
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230 @property |
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231 def media_types_inv(self) -> dict: |
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232 if self._media_types_inv is None: |
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233 raise Exception("self._media_types_inv should not be None!") |
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234 return self._media_types_inv |
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235 |
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236 def bind(self, **kwargs: Callable) -> None: |
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237 self.bindings.clear() |
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238 for key, cb in kwargs.items(): |
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239 if key not in ("desktop_sharing",): |
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240 raise ValueError( |
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241 'Only "desktop_sharing" is currently allowed for binding' |
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242 ) |
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243 self.bindings[key] = cb |
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244 |
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245 def generate_dot_file( |
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246 self, |
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247 filename: str = "pipeline", |
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248 details: Gst.DebugGraphDetails = Gst.DebugGraphDetails.ALL, |
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249 with_timestamp: bool = True, |
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250 bin_: Gst.Bin | None = None, |
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251 ) -> None: |
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252 """Generate Dot File for debugging |
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253 |
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254 ``GST_DEBUG_DUMP_DOT_DIR`` environment variable must be set to destination dir. |
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255 ``dot -Tpng -o <filename>.png <filename>.dot`` can be use to convert to a PNG file. |
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256 See |
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257 https://gstreamer.freedesktop.org/documentation/gstreamer/debugutils.html?gi-language=python#GstDebugGraphDetails |
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258 for details. |
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259 |
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260 @param filename: name of the generated file |
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261 @param details: which details to print |
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262 @param with_timestamp: if True, add a timestamp to filename |
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263 @param bin_: which bin to output. By default, the whole pipeline |
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264 (``self.pipeline``) will be used. |
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265 """ |
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266 if bin_ is None: |
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267 bin_ = self.pipeline |
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268 if with_timestamp: |
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269 timestamp = datetime.now().isoformat(timespec="milliseconds") |
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270 filename = f"{timestamp}_filename" |
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271 |
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272 Gst.debug_bin_to_dot_file(bin_, details, filename) |
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273 |
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274 def get_sdp_mline_index(self, media_type: str) -> int: |
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275 """Gets the sdpMLineIndex for a given media type. |
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276 |
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277 @param media_type: The type of the media. |
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278 """ |
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279 for index, m_type in self.media_types.items(): |
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280 if m_type == media_type: |
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281 return index |
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282 raise ValueError(f"Media type '{media_type}' not found") |
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283 |
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284 def _set_media_types(self, offer_sdp: str) -> None: |
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285 """Sets media types from offer SDP |
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286 |
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287 @param offer: RTC session description containing the offer |
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288 """ |
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289 sdp_lines = offer_sdp.splitlines() |
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290 media_types = {} |
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291 mline_index = 0 |
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292 |
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293 for line in sdp_lines: |
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294 if line.startswith("m="): |
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295 media_types[mline_index] = line[2 : line.find(" ")] |
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296 mline_index += 1 |
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297 |
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298 self.media_types = media_types |
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299 |
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300 def _a_call(self, method, *args, **kwargs): |
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301 """Call an async method in main thread""" |
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302 aio.run_from_thread(method, *args, **kwargs, loop=self.main_loop) |
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303 |
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304 def get_payload_types( |
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305 self, sdpmsg, video_encoding: str, audio_encoding: str |
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306 ) -> dict[str, int | None]: |
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307 """Find the payload types for the specified video and audio encoding. |
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308 |
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309 Very simplistically finds the first payload type matching the encoding |
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310 name. More complex applications will want to match caps on |
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311 profile-level-id, packetization-mode, etc. |
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312 """ |
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313 # method coming from gstreamer example (Matthew Waters, Nirbheek Chauhan) at |
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314 # subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py |
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315 video_pt = None |
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316 audio_pt = None |
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317 for i in range(0, sdpmsg.medias_len()): |
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318 media = sdpmsg.get_media(i) |
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319 for j in range(0, media.formats_len()): |
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320 fmt = media.get_format(j) |
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321 if fmt == "webrtc-datachannel": |
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322 continue |
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323 pt = int(fmt) |
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324 caps = media.get_caps_from_media(pt) |
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325 s = caps.get_structure(0) |
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326 encoding_name = s["encoding-name"] |
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327 if video_pt is None and encoding_name == video_encoding: |
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328 video_pt = pt |
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329 elif audio_pt is None and encoding_name == audio_encoding: |
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330 audio_pt = pt |
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331 return {video_encoding: video_pt, audio_encoding: audio_pt} |
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332 |
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333 def parse_ice_candidate(self, candidate_string): |
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334 """Parses the ice candidate string. |
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335 |
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336 @param candidate_string: The ice candidate string to be parsed. |
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337 """ |
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338 pattern = re.compile( |
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339 r"candidate:(?P<foundation>\S+) (?P<component_id>\d+) (?P<transport>\S+) " |
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340 r"(?P<priority>\d+) (?P<address>\S+) (?P<port>\d+) typ " |
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341 r"(?P<type>\S+)(?: raddr (?P<rel_addr>\S+) rport " |
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342 r"(?P<rel_port>\d+))?(?: generation (?P<generation>\d+))?" |
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343 ) |
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344 match = pattern.match(candidate_string) |
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345 if match: |
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346 candidate_dict = match.groupdict() |
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347 |
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348 # Apply the correct types to the dictionary values |
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349 candidate_dict["component_id"] = int(candidate_dict["component_id"]) |
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350 candidate_dict["priority"] = int(candidate_dict["priority"]) |
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351 candidate_dict["port"] = int(candidate_dict["port"]) |
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352 |
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353 if candidate_dict["rel_port"]: |
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354 candidate_dict["rel_port"] = int(candidate_dict["rel_port"]) |
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355 |
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356 if candidate_dict["generation"]: |
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357 candidate_dict["generation"] = candidate_dict["generation"] |
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358 |
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359 # Remove None values |
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360 return {k: v for k, v in candidate_dict.items() if v is not None} |
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361 else: |
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362 log.warning(f"can't parse candidate: {candidate_string!r}") |
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363 return None |
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364 |
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365 def build_ice_candidate(self, parsed_candidate): |
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366 """Builds ICE candidate |
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367 |
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368 @param parsed_candidate: Dictionary containing parsed ICE candidate |
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369 """ |
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370 base_format = ( |
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371 "candidate:{foundation} {component_id} {transport} {priority} " |
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372 "{address} {port} typ {type}" |
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373 ) |
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374 |
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375 if parsed_candidate.get("rel_addr") and parsed_candidate.get("rel_port"): |
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376 base_format += " raddr {rel_addr} rport {rel_port}" |
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377 |
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378 if parsed_candidate.get("generation"): |
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379 base_format += " generation {generation}" |
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380 |
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381 return base_format.format(**parsed_candidate) |
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382 |
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383 def extract_ufrag_pwd(self, sdp: str) -> None: |
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384 """Retrieves ICE password and user fragment for SDP offer. |
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385 |
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386 @param sdp: The Session Description Protocol offer string. |
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387 """ |
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388 lines = sdp.splitlines() |
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389 media = '' |
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390 mid_media_map = {} |
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391 bundle_media = set() |
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392 bundle_ufrag = '' |
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393 bundle_pwd = '' |
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394 in_bundle = False |
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395 |
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396 for line in lines: |
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397 if line.startswith('m='): |
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398 media = line.split('=')[1].split()[0] |
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399 elif line.startswith('a=mid:'): |
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400 mid = line.split(':')[1].strip() |
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401 mid_media_map[mid] = media |
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402 elif line.startswith('a=group:BUNDLE'): |
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403 in_bundle = True |
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404 bundle_media = set(line.split(':')[1].strip().split()) |
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405 elif line.startswith('a=ice-ufrag:'): |
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406 if in_bundle: |
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407 bundle_ufrag = line.split(':')[1].strip() |
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408 else: |
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409 self.ufrag[media] = line.split(':')[1].strip() |
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410 elif line.startswith('a=ice-pwd:'): |
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411 if in_bundle: |
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412 bundle_pwd = line.split(':')[1].strip() |
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413 else: |
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414 self.pwd[media] = line.split(':')[1].strip() |
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415 else: |
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416 in_bundle = False |
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417 |
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418 if bundle_ufrag and bundle_pwd: |
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419 for mid in bundle_media: |
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420 media = mid_media_map[mid] |
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421 self.ufrag[media] = bundle_ufrag |
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422 self.pwd[media] = bundle_pwd |
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423 |
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424 def reset_instance(self): |
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425 """Inits or resets the instance variables to their default state.""" |
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426 self.role: str | None = None |
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427 if self.pipeline is not None: |
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428 self.pipeline.set_state(Gst.State.NULL) |
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429 self.pipeline = None |
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430 self._remote_video_pad = None |
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431 self.sid: str | None = None |
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432 self.offer: str | None = None |
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433 self.local_candidates_buffer = {} |
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434 self.ufrag: dict[str, str] = {} |
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435 self.pwd: dict[str, str] = {} |
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436 self.callee: jid.JID | None = None |
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437 self._media_types = None |
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438 self._media_types_inv = None |
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439 self._sdp_set: bool = False |
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440 self.remote_candidates_buffer: dict[str, dict[str, list]] = { |
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441 "audio": {"candidates": []}, |
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442 "video": {"candidates": []}, |
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443 } |
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444 self._media_types = None |
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445 self._media_types_inv = None |
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446 self.audio_valve = None |
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447 self.video_valve = None |
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448 self.video_selector = None |
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449 if self.desktop_portal is not None: |
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450 self.desktop_portal.end_session() |
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451 self.desktop_sharing = False |
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452 self.desktop_sink_pad = None |
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453 self.bindings = {} |
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454 if self.reset_cb is not None: |
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455 self.reset_cb() |
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456 self.data_channels: dict[str, GstWebRTC.WebRTCDataChannel] = {} |
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457 |
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458 @property |
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459 def data_channel(self) -> GstWebRTC.WebRTCDataChannel: |
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460 """Convenience method to get WebRTCDataChannel instance when there is only one.""" |
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461 if len(self.data_channels) != 1: |
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462 raise exceptions.InternalError( |
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463 "self.data_channel can only be used in a single Data Channel scenario. " |
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464 "Use self.data_channels dict instead." |
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465 ) |
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466 return next(iter(self.data_channels.values())) |
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467 |
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468 async def setup_call( |
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469 self, |
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470 role: str, |
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471 audio_pt: int | None = 96, |
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472 video_pt: int | None = 97, |
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473 ) -> None: |
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474 """Sets up the call. |
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475 |
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476 This method establishes the Gstreamer pipeline for audio and video communication. |
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477 The method also manages STUN and TURN server configurations, signal watchers, and |
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478 various connection handlers for the webrtcbin. |
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479 |
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480 @param role: The role for the call, either 'initiator' or 'responder'. |
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481 @param audio_pt: The payload type for the audio stream. |
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482 @param video_pt: The payload type for the video stream |
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483 |
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484 @raises NotImplementedError: If audio_pt or video_pt is set to None. |
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485 @raises AssertionError: If the role is not 'initiator' or 'responder'. |
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486 """ |
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487 assert role in ("initiator", "responder") |
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488 self.role = role |
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489 |
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490 |
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491 if isinstance(self.sources_data, SourcesPipeline): |
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492 if self.sources_data.video_pipeline!= "" and video_pt is None: |
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493 raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG) |
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494 if self.sources_data.audio_pipeline!= "" and audio_pt is None: |
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495 raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG) |
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496 elif isinstance(self.sources_data, SourcesNone): |
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497 pass |
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498 else: |
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499 if audio_pt is None or video_pt is None: |
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500 raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG) |
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501 |
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502 match self.sources_data: |
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503 case SourcesAuto(): |
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504 video_source_elt = VIDEO_SOURCE_AUTO |
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505 audio_source_elt = AUDIO_SOURCE_AUTO |
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506 case SourcesNone(): |
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507 video_source_elt = "" |
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508 audio_source_elt = "" |
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509 case SourcesPipeline() as source: |
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510 if source.video_pipeline is None: |
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511 video_source_elt = VIDEO_SOURCE_AUTO |
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512 else: |
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513 video_source_elt = source.video_pipeline |
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514 if source.audio_pipeline is None: |
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515 audio_source_elt = AUDIO_SOURCE_AUTO |
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516 else: |
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517 audio_source_elt = source.audio_pipeline |
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518 case SourcesTest(): |
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519 video_source_elt = "videotestsrc is-live=true pattern=ball" |
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520 audio_source_elt = "audiotestsrc" |
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521 case _: |
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522 raise exceptions.InternalError( |
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523 f'Unexpected "sources_data" value: {self.sources_data!r}' |
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524 ) |
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525 |
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526 match self.sinks_data: |
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527 case SinksApp(): |
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528 local_video_sink_elt = ( |
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529 "appsink name=local_video_sink emit-signals=true drop=true " |
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530 "max-buffers=1 sync=True" |
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531 ) |
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532 case SinksAuto(): |
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533 if isinstance(self.sources_data, SourcesNone): |
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534 local_video_sink_elt = "" |
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535 else: |
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536 local_video_sink_elt = "autovideosink" |
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537 case SinksNone(): |
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538 local_video_sink_elt = "" |
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539 case _: |
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540 raise exceptions.InternalError( |
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541 f'Unexpected "sinks_data" value {self.sinks_data!r}' |
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|
542 ) |
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543 |
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544 gst_pipe_elements = [ |
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545 "webrtcbin latency=30 name=sendrecv bundle-policy=max-bundle" |
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546 ] |
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547 |
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548 if self.merge_pip and local_video_sink_elt: |
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549 # Compositor is used to merge local video feedback in video sink, useful when |
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550 # we have only a single video sink. |
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551 gst_pipe_elements.append( |
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552 "compositor name=compositor background=black " |
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553 f"! video/x-raw,width={self.target_width}," |
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554 f"height={self.target_height},framerate=30/1 " |
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555 f"! {local_video_sink_elt}" |
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556 ) |
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557 local_video_sink_elt = "compositor.sink_1" |
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558 |
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559 if video_source_elt: |
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560 # Video source with an input-selector to switch between normal and video mute |
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561 # (or desktop sharing). |
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562 gst_pipe_elements.append(f""" |
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563 input-selector name=video_selector |
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564 ! videorate drop-only=1 max-rate=30 |
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565 ! video/x-raw,framerate=30/1 |
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|
566 ! tee name=t |
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diff
changeset
|
567 |
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568 {video_source_elt} name=video_src |
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|
569 ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream |
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570 ! video_selector. |
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|
571 |
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572 videotestsrc name=muted_src is-live=true pattern=black |
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573 ! queue leaky=downstream |
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|
574 ! video_selector. |
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|
575 |
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|
576 t. |
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|
577 ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream |
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578 ! videoscale |
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|
579 ! videoconvert |
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|
580 ! vp8enc deadline=1 keyframe-max-dist=30 |
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|
581 ! rtpvp8pay picture-id-mode=15-bit |
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582 ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt} |
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583 ! sendrecv. |
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|
584 """) |
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|
585 |
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|
586 if local_video_sink_elt: |
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|
587 # Local video feedback. |
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|
588 gst_pipe_elements.append(f""" |
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|
589 t. |
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|
590 ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream |
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|
591 ! videoconvert |
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592 ! {local_video_sink_elt} |
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593 """) |
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594 |
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595 if audio_source_elt: |
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596 # Audio with a valve for muting. |
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597 gst_pipe_elements.append(r""" |
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598 {audio_source_elt} name=audio_src |
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599 ! valve |
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600 ! queue max-size-buffers=10 max-size-time=0 max-size-bytes=0 leaky=downstream |
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601 ! audioconvert |
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602 ! audioresample |
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603 ! opusenc audio-type=voice |
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604 ! rtpopuspay |
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605 ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} |
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606 ! sendrecv. |
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607 """) |
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608 |
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609 self.gst_pipe_desc = "\n\n".join(gst_pipe_elements) |
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610 |
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611 log.debug(f"Gstreamer pipeline: {self.gst_pipe_desc}") |
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612 |
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613 # Create the pipeline |
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614 try: |
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615 self.pipeline = Gst.parse_launch(self.gst_pipe_desc) |
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616 except Exception: |
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617 log.exception("Can't parse pipeline") |
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618 self.pipeline = None |
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619 if not self.pipeline: |
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620 raise exceptions.InternalError("Failed to create Gstreamer pipeline.") |
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621 |
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622 if not isinstance(self.pipeline, Gst.Pipeline): |
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623 # in the case of Data Channel there is a single element, and Gst.parse_launch |
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624 # doesn't create a Pipeline in this case, so we do it manually. |
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625 pipeline = Gst.Pipeline() |
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626 pipeline.add(self.pipeline) |
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627 self.pipeline = pipeline |
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628 |
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629 self.webrtcbin = self.pipeline.get_by_name("sendrecv") |
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630 if self.webrtcbin is None: |
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631 raise exceptions.InternalError("Can't get the pipeline.") |
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632 |
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633 # If video or audio sources are not created, ``get_by_name`` will return None. |
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634 self.video_src = self.pipeline.get_by_name("video_src") |
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635 self.muted_src = self.pipeline.get_by_name("muted_src") |
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636 self.video_selector = self.pipeline.get_by_name("video_selector") |
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637 if self.video_src and isinstance(self.sources_data, SourcesPipeline): |
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638 for name, value in self.sources_data.video_properties.items(): |
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639 self.video_src.set_property(name, value) |
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640 |
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641 self.audio_src = self.pipeline.get_by_name("audio_src") |
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642 if self.audio_src and isinstance(self.sources_data, SourcesPipeline): |
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643 for name, value in self.sources_data.audio_properties.items(): |
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644 self.audio_src.set_property(name, value) |
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645 |
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646 self.audio_valve = self.pipeline.get_by_name("audio_valve") |
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647 |
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648 if self.video_muted: |
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649 self.on_video_mute(True) |
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650 if self.audio_muted: |
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651 self.on_audio_mute(True) |
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652 |
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653 # set STUN and TURN servers |
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654 external_disco = data_format.deserialise( |
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655 await self.bridge.external_disco_get("", self.profile), type_check=list |
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656 ) |
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657 |
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658 for server in external_disco: |
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659 if server["type"] == "stun": |
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660 if server["transport"] == "tcp": |
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661 log.info( |
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662 "ignoring TCP STUN server, GStreamer only support one STUN server" |
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663 ) |
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664 url = f"stun://{server['host']}:{server['port']}" |
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665 log.debug(f"adding stun server: {url}") |
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666 self.webrtcbin.set_property("stun-server", url) |
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667 elif server["type"] == "turn": |
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668 url = "{scheme}://{username}:{password}@{host}:{port}".format( |
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669 scheme="turns" if server["transport"] == "tcp" else "turn", |
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670 username=quote_plus(server["username"]), |
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671 password=quote_plus(server["password"]), |
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672 host=server["host"], |
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673 port=server["port"], |
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674 ) |
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675 log.debug(f"adding turn server: {url}") |
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676 |
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677 if not self.webrtcbin.emit("add-turn-server", url): |
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678 log.warning(f"Erreur while adding TURN server {url}") |
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679 |
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680 # local video feedback |
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681 if isinstance(self.sinks_data, SinksApp): |
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682 local_video_sink = self.pipeline.get_by_name("local_video_sink") |
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683 if local_video_sink is not None: |
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684 local_video_sink.set_property("emit-signals", True) |
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685 local_video_sink.connect("new-sample", self.sinks_data.local_video_cb) |
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686 local_video_sink_caps = Gst.Caps.from_string(f"video/x-raw,format=RGB") |
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687 local_video_sink.set_property("caps", local_video_sink_caps) |
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688 |
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689 # Create bus and associate signal watchers |
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690 self.bus = self.pipeline.get_bus() |
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691 if not self.bus: |
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692 log.error("Failed to get bus from pipeline.") |
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693 return |
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694 |
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695 self.bus.add_signal_watch() |
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696 self.webrtcbin.connect("pad-added", self.on_pad_added) |
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697 self.bus.connect("message::error", self.on_bus_error) |
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698 self.bus.connect("message::eos", self.on_bus_eos) |
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699 self.webrtcbin.connect("on-negotiation-needed", self.on_negotiation_needed) |
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700 self.webrtcbin.connect("on-ice-candidate", self.on_ice_candidate) |
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701 self.webrtcbin.connect( |
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702 "notify::ice-gathering-state", self.on_ice_gathering_state_change |
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703 ) |
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704 self.webrtcbin.connect( |
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705 "notify::ice-connection-state", self.on_ice_connection_state |
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706 ) |
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707 |
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708 for dc_data in self.dc_data_list: |
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709 self.create_data_channel(dc_data) |
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710 |
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711 def create_data_channel(self, dc_data: SourcesDataChannel|SinksDataChannel) -> None: |
79c8a70e1813
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|
712 """Create a Data Channel and connect relevant callbacks.""" |
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713 assert self.pipeline is not None |
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|
714 if isinstance(dc_data, SourcesDataChannel): |
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715 # Data channel configuration for compatibility with browser defaults |
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716 data_channel_options = Gst.Structure.new_empty("data-channel-options") |
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parents:
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|
717 data_channel_options.set_value("ordered", True) |
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718 data_channel_options.set_value("protocol", "") |
d01b8d002619
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|
719 |
d01b8d002619
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|
720 # Create the data channel |
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|
721 self.pipeline.set_state(Gst.State.READY) |
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722 self.data_channels[dc_data.name] = data_channel = self.webrtcbin.emit( |
79c8a70e1813
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|
723 "create-data-channel", dc_data.name, data_channel_options |
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|
724 ) |
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|
725 if data_channel is None: |
4233
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|
726 log.error("Failed to create data channel") |
d01b8d002619
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diff
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|
727 return |
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|
728 data_channel.connect("on-open", dc_data.dc_open_cb) |
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729 elif isinstance(dc_data, SinksDataChannel): |
79c8a70e1813
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|
730 self.webrtcbin.connect("on-data-channel", dc_data.dc_on_data_channel) |
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|
731 else: |
79c8a70e1813
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|
732 raise ValueError( |
79c8a70e1813
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|
733 "Only SourcesDataChannel or SinksDataChannel are allowed." |
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|
734 ) |
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|
735 |
4139
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|
736 def start_pipeline(self) -> None: |
6745c6bd4c7a
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|
737 """Starts the GStreamer pipeline.""" |
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|
738 log.debug("starting the pipeline") |
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|
739 self.pipeline.set_state(Gst.State.PLAYING) |
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|
740 |
6745c6bd4c7a
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|
741 def on_negotiation_needed(self, webrtc): |
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742 """Initiate SDP offer when negotiation is needed.""" |
6745c6bd4c7a
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|
743 log.debug("Negotiation needed.") |
6745c6bd4c7a
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|
744 if self.role == "initiator": |
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|
745 log.debug("Creating offer…") |
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|
746 promise = Gst.Promise.new_with_change_func(self.on_offer_created) |
6745c6bd4c7a
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747 self.webrtcbin.emit("create-offer", None, promise) |
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|
748 |
6745c6bd4c7a
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|
749 def on_offer_created(self, promise): |
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|
750 """Callback for when SDP offer is created.""" |
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|
751 log.info("on_offer_created called") |
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|
752 assert promise.wait() == Gst.PromiseResult.REPLIED |
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753 reply = promise.get_reply() |
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|
754 if reply is None: |
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|
755 log.error("Promise reply is None. Offer creation might have failed.") |
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|
756 return |
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|
757 offer = reply["offer"] |
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|
758 self.offer = offer.sdp.as_text() |
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changeset
|
759 log.info(f"SDP offer created: \n{self.offer}") |
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|
760 self._set_media_types(self.offer) |
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|
761 promise = Gst.Promise.new() |
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|
762 self.webrtcbin.emit("set-local-description", offer, promise) |
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|
763 promise.interrupt() |
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changeset
|
764 self._a_call(self._start_call) |
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|
765 |
6745c6bd4c7a
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|
766 def on_answer_set(self, promise): |
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767 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
768 |
6745c6bd4c7a
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|
769 def on_answer_created(self, promise, _, __): |
6745c6bd4c7a
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|
770 """Callback for when SDP answer is created.""" |
6745c6bd4c7a
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|
771 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
772 reply = promise.get_reply() |
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|
773 answer = reply["answer"] |
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|
774 promise = Gst.Promise.new() |
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|
775 self.webrtcbin.emit("set-local-description", answer, promise) |
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|
776 promise.interrupt() |
6745c6bd4c7a
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|
777 answer_sdp = answer.sdp.as_text() |
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|
778 log.info(f"SDP answer set: \n{answer_sdp}") |
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|
779 self.sdp_set = True |
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|
780 self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile) |
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changeset
|
781 |
6745c6bd4c7a
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|
782 def on_offer_set(self, promise): |
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|
783 assert promise.wait() == Gst.PromiseResult.REPLIED |
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changeset
|
784 promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None) |
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|
785 self.webrtcbin.emit("create-answer", None, promise) |
6745c6bd4c7a
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|
786 |
6745c6bd4c7a
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|
787 def link_element_or_pad( |
6745c6bd4c7a
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|
788 self, source: Gst.Element, dest: Gst.Element | Gst.Pad |
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|
789 ) -> bool: |
6745c6bd4c7a
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diff
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|
790 """Check if dest is a pad or an element, and link appropriately""" |
6745c6bd4c7a
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|
791 src_pad = source.get_static_pad("src") |
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|
792 |
6745c6bd4c7a
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|
793 if isinstance(dest, Gst.Pad): |
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|
794 # If the dest is a pad, link directly |
6745c6bd4c7a
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|
795 if not src_pad.link(dest) == Gst.PadLinkReturn.OK: |
6745c6bd4c7a
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|
796 log.error( |
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diff
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|
797 "Failed to link 'conv' to the compositor's newly requested pad!" |
6745c6bd4c7a
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|
798 ) |
6745c6bd4c7a
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|
799 return False |
6745c6bd4c7a
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|
800 elif isinstance(dest, Gst.Element): |
6745c6bd4c7a
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|
801 return source.link(dest) |
6745c6bd4c7a
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|
802 else: |
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|
803 log.error(f"Unexpected type for dest: {type(dest)}") |
4139
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|
804 return False |
6745c6bd4c7a
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diff
changeset
|
805 |
6745c6bd4c7a
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|
806 return True |
6745c6bd4c7a
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diff
changeset
|
807 |
6745c6bd4c7a
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parents:
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changeset
|
808 def scaled_dimensions( |
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diff
changeset
|
809 self, original_width: int, original_height: int, max_width: int, max_height: int |
6745c6bd4c7a
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|
810 ) -> tuple[int, int]: |
6745c6bd4c7a
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changeset
|
811 """Calculates the scaled dimensions preserving aspect ratio. |
6745c6bd4c7a
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|
812 |
6745c6bd4c7a
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diff
changeset
|
813 @param original_width: Original width of the video stream. |
6745c6bd4c7a
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diff
changeset
|
814 @param original_height: Original height of the video stream. |
6745c6bd4c7a
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diff
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|
815 @param max_width: Maximum desired width for the scaled video. |
6745c6bd4c7a
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parents:
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changeset
|
816 @param max_height: Maximum desired height for the scaled video. |
6745c6bd4c7a
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diff
changeset
|
817 @return: The width and height of the scaled video. |
6745c6bd4c7a
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changeset
|
818 """ |
6745c6bd4c7a
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diff
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|
819 aspect_ratio = original_width / original_height |
6745c6bd4c7a
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diff
changeset
|
820 new_width = int(max_height * aspect_ratio) |
6745c6bd4c7a
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diff
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|
821 |
6745c6bd4c7a
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parents:
diff
changeset
|
822 if new_width <= max_width: |
6745c6bd4c7a
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|
823 return new_width, max_height |
6745c6bd4c7a
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changeset
|
824 |
6745c6bd4c7a
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|
825 new_height = int(max_width / aspect_ratio) |
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826 return max_width, new_height |
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827 |
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828 def on_remote_decodebin_stream(self, _, pad: Gst.Pad) -> None: |
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829 """Handle the stream from the remote decodebin. |
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830 |
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831 This method processes the incoming stream from the remote decodebin, determining |
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832 whether it's video or audio. It then sets up the appropriate GStreamer elements |
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833 for video/audio processing and adds them to the pipeline. |
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834 |
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835 @param pad: The Gst.Pad from the remote decodebin producing the stream. |
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836 """ |
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837 assert self.pipeline is not None |
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838 if not pad.has_current_caps(): |
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839 log.error(f"{pad} has no caps, ignoring") |
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840 return |
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841 |
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842 caps = pad.get_current_caps() |
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843 assert len(caps) |
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844 s = caps[0] |
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845 name = s.get_name() |
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846 log.debug(f"====> NAME START: {name}") |
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847 |
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848 q = Gst.ElementFactory.make("queue") |
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849 |
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850 if name.startswith("video"): |
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851 log.debug("===> VIDEO OK") |
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852 |
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853 self._remote_video_pad = pad |
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854 |
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855 # Check and log the original size of the video |
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856 width = self.target_width |
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857 height = self.target_height |
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858 log.info(f"Original video size: {width}x{height}") |
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859 |
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860 # This is a fix for an issue found with Movim on desktop: a non standard |
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861 # resolution is used (990x557) resulting in bad alignement and no color in |
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862 # rendered image |
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863 adjust_resolution = width % 4 != 0 or height % 4 != 0 |
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864 if adjust_resolution: |
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865 log.info("non standard resolution, we need to adjust size") |
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|
866 width = (width + 3) // 4 * 4 |
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867 height = (height + 3) // 4 * 4 |
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868 log.info(f"Adjusted video size: {width}x{height}") |
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|
869 |
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870 conv = Gst.ElementFactory.make("videoconvert") |
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871 if self.merge_pip: |
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872 # with ``merge_pip`` set, we plug the remote stream to the composer |
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873 compositor = self.pipeline.get_by_name("compositor") |
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874 |
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875 sink1_pad = compositor.get_static_pad("sink_1") |
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876 |
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|
877 local_width, local_height = self.scaled_dimensions( |
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878 sink1_pad.get_property("width"), |
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879 sink1_pad.get_property("height"), |
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|
880 width // 3, |
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|
881 height // 3, |
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|
882 ) |
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|
883 |
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|
884 sink1_pad.set_property("xpos", width - local_width) |
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|
885 sink1_pad.set_property("ypos", height - local_height) |
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|
886 sink1_pad.set_property("width", local_width) |
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|
887 sink1_pad.set_property("height", local_height) |
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888 sink1_pad.set_property("sizing-policy", 1) |
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|
889 sink1_pad.set_property("zorder", 1) |
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|
890 |
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|
891 # Request a new pad for the remote stream |
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|
892 sink_pad_template = compositor.get_pad_template("sink_%u") |
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|
893 remote_video_sink = compositor.request_pad(sink_pad_template, None, None) |
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|
894 remote_video_sink.set_property("zorder", 0) |
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895 remote_video_sink.set_property("width", width) |
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896 remote_video_sink.set_property("height", height) |
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|
897 remote_video_sink.set_property("sizing-policy", 1) |
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898 elif isinstance(self.sinks_data, SinksApp): |
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|
899 # ``app`` sink without ``self.merge_pip`` set, be create the sink and |
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900 # connect it to the ``remote_video_cb``. |
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901 remote_video_sink = Gst.ElementFactory.make("appsink") |
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|
902 |
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903 remote_video_caps = Gst.Caps.from_string("video/x-raw,format=RGB") |
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904 remote_video_sink.set_property("caps", remote_video_caps) |
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905 |
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|
906 remote_video_sink.set_property("emit-signals", True) |
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907 remote_video_sink.set_property("drop", True) |
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908 remote_video_sink.set_property("max-buffers", 1) |
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|
909 remote_video_sink.set_property("sync", True) |
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|
910 remote_video_sink.connect("new-sample", self.sinks_data.remote_video_cb) |
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|
911 self.pipeline.add(remote_video_sink) |
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912 elif isinstance(self.sinks_data, SinksAuto): |
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913 # if ``self.merge_pip`` is not set, we create a dedicated |
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|
914 # ``autovideosink`` for remote stream. |
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|
915 remote_video_sink = Gst.ElementFactory.make("autovideosink") |
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|
916 self.pipeline.add(remote_video_sink) |
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|
917 else: |
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|
918 raise exceptions.InternalError( |
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|
919 f'Unhandled "sinks_data" value: {self.sinks_data!r}' |
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|
920 ) |
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|
921 |
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|
922 if adjust_resolution: |
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|
923 videoscale = Gst.ElementFactory.make("videoscale") |
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|
924 adjusted_caps = Gst.Caps.from_string( |
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|
925 f"video/x-raw,width={width},height={height}" |
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|
926 ) |
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|
927 capsfilter = Gst.ElementFactory.make("capsfilter") |
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|
928 capsfilter.set_property("caps", adjusted_caps) |
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|
929 |
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|
930 self.pipeline.add(q, conv, videoscale, capsfilter) |
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|
931 |
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|
932 self.pipeline.sync_children_states() |
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|
933 ret = pad.link(q.get_static_pad("sink")) |
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|
934 if ret != Gst.PadLinkReturn.OK: |
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|
935 log.error(f"Error linking pad: {ret}") |
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|
936 q.link(conv) |
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|
937 conv.link(videoscale) |
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|
938 videoscale.link(capsfilter) |
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|
939 self.link_element_or_pad(capsfilter.link, remote_video_sink) |
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|
940 |
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|
941 else: |
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|
942 self.pipeline.add(q, conv) |
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|
943 |
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|
944 self.pipeline.sync_children_states() |
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|
945 ret = pad.link(q.get_static_pad("sink")) |
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|
946 if ret != Gst.PadLinkReturn.OK: |
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|
947 log.error(f"Error linking pad: {ret}") |
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|
948 q.link(conv) |
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|
949 self.link_element_or_pad(conv, remote_video_sink) |
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|
950 |
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|
951 elif name.startswith("audio"): |
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|
952 log.debug("===> Audio OK") |
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953 conv = Gst.ElementFactory.make("audioconvert") |
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954 resample = Gst.ElementFactory.make("audioresample") |
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955 remote_audio_sink = Gst.ElementFactory.make("autoaudiosink") |
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|
956 self.pipeline.add(q, conv, resample, remote_audio_sink) |
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|
957 self.pipeline.sync_children_states() |
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958 ret = pad.link(q.get_static_pad("sink")) |
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|
959 if ret != Gst.PadLinkReturn.OK: |
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|
960 log.error(f"Error linking pad: {ret}") |
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|
961 q.link(conv) |
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|
962 conv.link(resample) |
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|
963 resample.link(remote_audio_sink) |
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|
964 |
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965 else: |
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|
966 log.warning(f"unmanaged name: {name!r}") |
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|
967 |
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|
968 def on_pad_added(self, __, pad: Gst.Pad) -> None: |
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969 """Handle the addition of a new pad to the element. |
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|
970 |
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|
971 When a new source pad is added to the element, this method creates a decodebin, |
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|
972 connects it to handle the stream, and links the pad to the decodebin. |
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|
973 |
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|
974 @param __: Placeholder for the signal source. Not used in this method. |
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|
975 @param pad: The newly added pad. |
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|
976 """ |
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|
977 log.debug("on_pad_added") |
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|
978 if pad.direction != Gst.PadDirection.SRC: |
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diff
changeset
|
979 return |
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diff
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|
980 |
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|
981 decodebin = Gst.ElementFactory.make("decodebin") |
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|
982 decodebin.connect("pad-added", self.on_remote_decodebin_stream) |
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|
983 self.pipeline.add(decodebin) |
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|
984 decodebin.sync_state_with_parent() |
6745c6bd4c7a
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|
985 pad.link(decodebin.get_static_pad("sink")) |
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diff
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|
986 |
4233
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parents:
4209
diff
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|
987 async def _call_start(self, callee: jid.JID, call_data: dict, profile: str) -> str: |
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4209
diff
changeset
|
988 return await self.bridge.call_start( |
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4209
diff
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989 str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile |
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4209
diff
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|
990 ) |
d01b8d002619
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parents:
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diff
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|
991 |
4139
6745c6bd4c7a
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|
992 async def _start_call(self) -> None: |
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|
993 """Initiate the call. |
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|
994 |
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diff
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|
995 Initiates a call with the callee using the stored offer. If there are any buffered |
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|
996 local ICE candidates, they are sent as part of the initiation. |
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|
997 """ |
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|
998 assert self.callee |
4233
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|
999 assert self.call_start_cb is not None |
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|
1000 self.sid = await self.call_start_cb( |
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parents:
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1001 self.callee, {"sdp": self.offer}, self.profile |
4139
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|
1002 ) |
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|
1003 if self.local_candidates_buffer: |
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|
1004 log.debug( |
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|
1005 f"sending buffered local ICE candidates: {self.local_candidates_buffer}" |
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|
1006 ) |
4240
79c8a70e1813
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4233
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|
1007 if not self.pwd: |
4139
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|
1008 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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|
1009 self.extract_ufrag_pwd(sdp) |
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|
1010 ice_data = {} |
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|
1011 for media_type, candidates in self.local_candidates_buffer.items(): |
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|
1012 ice_data[media_type] = { |
4240
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1013 "ufrag": self.ufrag[media_type], |
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|
1014 "pwd": self.pwd[media_type], |
4139
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|
1015 "candidates": candidates, |
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|
1016 } |
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|
1017 await self.bridge.ice_candidates_add( |
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|
1018 self.sid, data_format.serialise(ice_data), self.profile |
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|
1019 ) |
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|
1020 self.local_candidates_buffer.clear() |
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|
1021 |
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|
1022 def _remote_sdp_set(self, promise) -> None: |
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|
1023 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
1024 self.sdp_set = True |
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|
1025 |
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|
1026 def on_accepted_call(self, sdp: str, profile: str) -> None: |
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|
1027 """Outgoing call has been accepted. |
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|
1028 |
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|
1029 @param sdp: The SDP answer string received from the other party. |
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|
1030 @param profile: Profile used for the call. |
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|
1031 """ |
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|
1032 log.debug(f"SDP answer received: \n{sdp}") |
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|
1033 |
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|
1034 __, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp) |
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|
1035 answer = GstWebRTC.WebRTCSessionDescription.new( |
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|
1036 GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg |
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|
1037 ) |
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|
1038 promise = Gst.Promise.new_with_change_func(self._remote_sdp_set) |
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|
1039 self.webrtcbin.emit("set-remote-description", answer, promise) |
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|
1040 |
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|
1041 async def answer_call(self, sdp: str, profile: str) -> None: |
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|
1042 """Answer an incoming call |
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|
1043 |
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|
1044 @param sdp: The SDP offer string received from the initiator. |
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|
1045 @param profile: Profile used for the call. |
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|
1046 |
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1047 @raise AssertionError: Raised when either "VP8" or "OPUS" is not present in |
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|
1048 payload types. |
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|
1049 """ |
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|
1050 log.debug(f"SDP offer received: \n{sdp}") |
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1051 self._set_media_types(sdp) |
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|
1052 __, offer_sdp_msg = GstSdp.SDPMessage.new_from_text(sdp) |
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changeset
|
1053 payload_types = self.get_payload_types( |
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|
1054 offer_sdp_msg, video_encoding="VP8", audio_encoding="OPUS" |
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|
1055 ) |
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|
1056 assert "VP8" in payload_types |
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|
1057 assert "OPUS" in payload_types |
4240
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1058 try: |
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1059 await self.setup_call( |
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1060 "responder", audio_pt=payload_types["OPUS"], video_pt=payload_types["VP8"] |
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1061 ) |
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1062 except Exception: |
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|
1063 log.exception("Can't setup call") |
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|
1064 raise |
4139
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|
1065 self.start_pipeline() |
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|
1066 offer = GstWebRTC.WebRTCSessionDescription.new( |
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|
1067 GstWebRTC.WebRTCSDPType.OFFER, offer_sdp_msg |
6745c6bd4c7a
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|
1068 ) |
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|
1069 promise = Gst.Promise.new_with_change_func(self.on_offer_set) |
6745c6bd4c7a
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|
1070 self.webrtcbin.emit("set-remote-description", offer, promise) |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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|
1071 |
6745c6bd4c7a
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|
1072 def on_ice_candidate(self, webrtc, mline_index, candidate_sdp): |
6745c6bd4c7a
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|
1073 """Handles the on-ice-candidate signal of webrtcbin. |
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|
1074 |
6745c6bd4c7a
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|
1075 @param webrtc: The webrtcbin element. |
6745c6bd4c7a
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1076 @param mlineindex: The mline index. |
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|
1077 @param candidate: The ICE candidate. |
6745c6bd4c7a
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|
1078 """ |
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1079 log.debug( |
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1080 f"Local ICE candidate. MLine Index: {mline_index}, Candidate: {candidate_sdp}" |
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1081 ) |
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1082 parsed_candidate = self.parse_ice_candidate(candidate_sdp) |
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1083 if parsed_candidate is None: |
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1084 log.warning(f"Can't parse candidate: {candidate_sdp}") |
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|
1085 return |
4139
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1086 try: |
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|
1087 media_type = self.media_types[mline_index] |
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|
1088 except KeyError: |
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|
1089 raise exceptions.InternalError("can't find media type") |
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|
1090 |
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|
1091 if self.sid is None: |
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Goffi <goffi@goffi.org>
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|
1092 log.debug("buffering local ICE candidate") |
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|
1093 self.local_candidates_buffer.setdefault(media_type, []).append( |
6745c6bd4c7a
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changeset
|
1094 parsed_candidate |
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changeset
|
1095 ) |
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|
1096 else: |
6745c6bd4c7a
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|
1097 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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|
1098 assert sdp is not None |
4240
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1099 self.extract_ufrag_pwd(sdp) |
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|
1100 ice_data = { |
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1101 "ufrag": self.ufrag[media_type], |
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|
1102 "pwd": self.pwd[media_type], |
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1103 "candidates": [parsed_candidate] |
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1104 } |
4139
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|
1105 self._a_call( |
6745c6bd4c7a
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diff
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|
1106 self.bridge.ice_candidates_add, |
6745c6bd4c7a
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|
1107 self.sid, |
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|
1108 data_format.serialise({media_type: ice_data}), |
6745c6bd4c7a
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changeset
|
1109 self.profile, |
6745c6bd4c7a
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diff
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|
1110 ) |
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|
1111 |
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|
1112 def on_ice_candidates_new(self, candidates: dict) -> None: |
6745c6bd4c7a
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diff
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|
1113 """Handle new ICE candidates. |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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|
1114 |
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diff
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|
1115 @param candidates: A dictionary containing media types ("audio" or "video") as |
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|
1116 keys and corresponding ICE data as values. |
6745c6bd4c7a
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|
1117 |
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|
1118 @raise exceptions.InternalError: Raised when sdp mline index is not found. |
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|
1119 """ |
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|
1120 if not self.sdp_set: |
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|
1121 log.debug("buffering remote ICE candidate") |
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|
1122 for media_type in ("audio", "video"): |
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|
1123 media_candidates = candidates.get(media_type) |
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|
1124 if media_candidates: |
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|
1125 buffer = self.remote_candidates_buffer[media_type] |
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|
1126 buffer["candidates"].extend(media_candidates["candidates"]) |
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diff
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|
1127 return |
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|
1128 for media_type, ice_data in candidates.items(): |
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|
1129 for candidate in ice_data["candidates"]: |
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|
1130 candidate_sdp = self.build_ice_candidate(candidate) |
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|
1131 try: |
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|
1132 mline_index = self.get_sdp_mline_index(media_type) |
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|
1133 except Exception as e: |
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|
1134 raise exceptions.InternalError(f"Can't find sdp mline index: {e}") |
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|
1135 self.webrtcbin.emit("add-ice-candidate", mline_index, candidate_sdp) |
4233
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|
1136 log.warning( |
4139
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|
1137 f"Remote ICE candidate added. MLine Index: {mline_index}, " |
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|
1138 f"{candidate_sdp}" |
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|
1139 ) |
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|
1140 |
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|
1141 def on_ice_gathering_state_change(self, pspec, __): |
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|
1142 state = self.webrtcbin.get_property("ice-gathering-state") |
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|
1143 log.debug(f"ICE gathering state changed to {state}") |
6745c6bd4c7a
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|
1144 |
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|
1145 def on_ice_connection_state(self, pspec, __): |
6745c6bd4c7a
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|
1146 state = self.webrtcbin.props.ice_connection_state |
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|
1147 if state == GstWebRTC.WebRTCICEConnectionState.FAILED: |
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|
1148 log.error("ICE connection failed") |
6745c6bd4c7a
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|
1149 log.info(f"ICE connection state changed to {state}") |
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|
1150 |
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|
1151 def on_bus_error(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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|
1152 """Handles the GStreamer bus error messages. |
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|
1153 |
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|
1154 @param bus: The GStreamer bus. |
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|
1155 @param message: The error message. |
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|
1156 """ |
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|
1157 err, debug = message.parse_error() |
6745c6bd4c7a
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|
1158 log.error(f"Error from {message.src.get_name()}: {err.message}") |
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|
1159 log.error(f"Debugging info: {debug}") |
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|
1160 |
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|
1161 def on_bus_eos(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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|
1162 """Handles the GStreamer bus eos messages. |
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|
1163 |
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|
1164 @param bus: The GStreamer bus. |
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|
1165 @param message: The eos message. |
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|
1166 """ |
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|
1167 log.info("End of stream") |
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|
1168 |
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|
1169 def on_audio_mute(self, muted: bool) -> None: |
4204
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|
1170 """Handles (un)muting of audio. |
879bad48cc2d
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|
1171 |
879bad48cc2d
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|
1172 @param muted: True if audio is muted. |
879bad48cc2d
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|
1173 """ |
4139
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|
1174 if self.audio_valve is not None: |
6745c6bd4c7a
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diff
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|
1175 self.audio_valve.set_property("drop", muted) |
6745c6bd4c7a
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diff
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|
1176 state = "muted" if muted else "unmuted" |
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1177 log.info(f"audio is now {state}") |
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1178 |
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1179 def on_video_mute(self, muted: bool) -> None: |
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1180 """Handles (un)muting of video. |
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1181 |
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1182 @param muted: True if video is muted. |
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1183 """ |
4139
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1184 if self.video_selector is not None: |
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1185 current_source = ( |
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1186 None if muted else "desktop" if self.desktop_sharing else "video" |
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1187 ) |
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1188 self.switch_video_source(current_source) |
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1189 state = "muted" if muted else "unmuted" |
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1190 log.info(f"Video is now {state}") |
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1191 |
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1192 def on_desktop_switch(self, desktop_active: bool) -> None: |
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1193 """Switches the video source between desktop and video. |
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1194 |
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1195 @param desktop_active: True if desktop must be active. False for video. |
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1196 """ |
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1197 if desktop_active and self.desktop_portal is not None: |
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1198 aio.run_async(self.on_desktop_switch_portal(desktop_active)) |
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1199 else: |
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1200 self.do_desktop_switch(desktop_active) |
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1201 |
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1202 async def on_desktop_switch_portal(self, desktop_active: bool) -> None: |
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1203 """Call freedesktop screenshare portal and the activate the shared stream""" |
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1204 assert self.desktop_portal is not None |
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1205 try: |
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1206 screenshare_data = await self.desktop_portal.request_screenshare() |
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1207 except exceptions.CancelError: |
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1208 self.desktop_sharing = False |
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1209 return |
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1210 self.desktop_sharing_data = {"path": str(screenshare_data["node_id"])} |
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1211 self.do_desktop_switch(desktop_active) |
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1212 |
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1213 def on_portal_session_closed(self) -> None: |
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1214 self.desktop_sharing = False |
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1215 |
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1216 def do_desktop_switch(self, desktop_active: bool) -> None: |
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1217 if self.video_muted: |
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1218 # Update the active source state but do not switch |
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1219 self.desktop_sharing = desktop_active |
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1220 return |
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1221 |
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1222 source = "desktop" if desktop_active else "video" |
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1223 self.switch_video_source(source) |
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1224 self.desktop_sharing = desktop_active |
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1225 |
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1226 def switch_video_source(self, source: str | None) -> None: |
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1227 """Activates the specified source while deactivating the others. |
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1228 |
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1229 @param source: 'desktop', 'video', 'muted' or None for muted source. |
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1230 """ |
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1231 if source is None: |
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1232 source = "muted" |
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1233 if source not in ["video", "muted", "desktop"]: |
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1234 raise ValueError( |
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1235 f"Invalid source: {source!r}, use one of {'video', 'muted', 'desktop'}" |
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1236 ) |
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1237 |
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1238 self.pipeline.set_state(Gst.State.PAUSED) |
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1239 |
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1240 # Create a new desktop source if necessary |
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1241 if source == "desktop": |
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1242 self._setup_desktop_source(self.desktop_sharing_data) |
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1243 |
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1244 # Activate the chosen source and deactivate the others |
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1245 for src_name in ["video", "muted", "desktop"]: |
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1246 src_element = self.pipeline.get_by_name(f"{src_name}_src") |
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1247 if src_name == source: |
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1248 if src_element: |
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1249 src_element.set_state(Gst.State.PLAYING) |
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1250 else: |
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1251 if src_element: |
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1252 if src_name == "desktop": |
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1253 self._remove_desktop_source(src_element) |
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1254 else: |
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1255 src_element.set_state(Gst.State.NULL) |
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1256 |
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1257 # Set the video_selector active pad |
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1258 if source == "desktop": |
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1259 if self.desktop_sink_pad: |
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1260 pad = self.desktop_sink_pad |
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1261 else: |
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1262 log.error(f"No desktop pad available") |
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1263 pad = None |
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1264 else: |
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1265 pad_name = f"sink_{['video', 'muted'].index(source)}" |
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1266 pad = self.video_selector.get_static_pad(pad_name) |
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1267 |
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1268 if pad is not None: |
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1269 self.video_selector.props.active_pad = pad |
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1270 |
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1271 self.pipeline.set_state(Gst.State.PLAYING) |
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1272 |
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1273 def _setup_desktop_source(self, properties: dict[str, object] | None) -> None: |
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1274 """Set up a new desktop source. |
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1275 |
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1276 @param properties: The properties to set on the desktop source. |
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1277 """ |
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1278 source_elt = "ximagesrc" if self.desktop_portal is None else "pipewiresrc" |
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1279 desktop_src = Gst.ElementFactory.make(source_elt, "desktop_src") |
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1280 if properties is None: |
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1281 properties = {} |
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1282 for key, value in properties.items(): |
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1283 log.debug(f"setting {source_elt} property: {key!r}={value!r}") |
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1284 desktop_src.set_property(key, value) |
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1285 video_convert = Gst.ElementFactory.make("videoconvert", "desktop_videoconvert") |
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1286 queue = Gst.ElementFactory.make("queue", "desktop_queue") |
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1287 queue.set_property("leaky", "downstream") |
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1288 |
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1289 self.pipeline.add(desktop_src) |
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1290 self.pipeline.add(video_convert) |
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1291 self.pipeline.add(queue) |
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1292 |
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1293 desktop_src.link(video_convert) |
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1294 video_convert.link(queue) |
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1295 |
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1296 sink_pad_template = self.video_selector.get_pad_template("sink_%u") |
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1297 self.desktop_sink_pad = self.video_selector.request_pad( |
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1298 sink_pad_template, None, None |
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1299 ) |
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1300 queue_src_pad = queue.get_static_pad("src") |
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1301 queue_src_pad.link(self.desktop_sink_pad) |
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1302 |
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1303 desktop_src.sync_state_with_parent() |
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1304 video_convert.sync_state_with_parent() |
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1305 queue.sync_state_with_parent() |
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1306 |
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1307 def _remove_desktop_source(self, desktop_src: Gst.Element) -> None: |
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1308 """Remove the desktop source from the pipeline. |
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1309 |
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1310 @param desktop_src: The desktop source to remove. |
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1311 """ |
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1312 # Remove elements for the desktop source |
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1313 video_convert = self.pipeline.get_by_name("desktop_videoconvert") |
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1314 queue = self.pipeline.get_by_name("desktop_queue") |
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1315 |
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1316 if video_convert: |
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1317 video_convert.set_state(Gst.State.NULL) |
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1318 desktop_src.unlink(video_convert) |
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1319 self.pipeline.remove(video_convert) |
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1320 |
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1321 if queue: |
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1322 queue.set_state(Gst.State.NULL) |
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1323 self.pipeline.remove(queue) |
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1324 |
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1325 desktop_src.set_state(Gst.State.NULL) |
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1326 self.pipeline.remove(desktop_src) |
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1327 |
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1328 # Release the pad associated with the desktop source |
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1329 if self.desktop_sink_pad: |
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1330 self.video_selector.release_request_pad(self.desktop_sink_pad) |
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1331 self.desktop_sink_pad = None |
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1332 |
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1333 if self.desktop_portal is not None: |
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1334 self.desktop_portal.end_session() |
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1335 |
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1336 async def end_call(self) -> None: |
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1337 """Stop streaming and clean instance""" |
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1338 self.reset_instance() |
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1339 |
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1340 |
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1341 class WebRTCCall: |
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1342 """Helper class to create and handle WebRTC. |
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1343 |
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1344 This class handles signals and communication of connection data with backend. |
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1345 |
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1346 """ |
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1347 |
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1348 def __init__( |
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1349 self, |
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1350 bridge, |
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1351 profile: str, |
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1352 callee: jid.JID, |
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1353 on_call_setup_cb: Callable | None = None, |
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1354 on_call_ended_cb: Callable | None = None, |
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1355 **kwargs, |
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1356 ): |
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1357 """Create and setup a webRTC instance |
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1358 |
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1359 @param bridge: async Bridge. |
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1360 @param profile: profile making or receiving the call |
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1361 @param callee: peer jid |
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1362 @param kwargs: extra kw args to use when instantiating WebRTC |
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1363 """ |
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1364 self.profile = profile |
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1365 self.webrtc = WebRTC(bridge, profile, **kwargs) |
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1366 self.webrtc.callee = callee |
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1367 self.on_call_setup_cb = on_call_setup_cb |
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1368 self.on_call_ended_cb = on_call_ended_cb |
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1369 bridge.register_signal("ice_candidates_new", self.on_ice_candidates_new, "plugin") |
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1370 bridge.register_signal("call_setup", self.on_call_setup, "plugin") |
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1371 bridge.register_signal("call_ended", self.on_call_ended, "plugin") |
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1372 |
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1373 @classmethod |
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1374 async def make_webrtc_call( |
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1375 cls, bridge, profile: str, call_data: CallData, **kwargs |
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1376 ) -> "WebRTCCall": |
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1377 """Create the webrtc_call instance |
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1378 |
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1379 @param call_data: Call data of the command |
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1380 @param kwargs: extra args used to instanciate WebRTCCall |
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1381 |
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1382 """ |
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1383 webrtc_call = cls(bridge, profile, call_data.callee, **call_data.kwargs, **kwargs) |
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1384 if call_data.sid is None: |
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1385 # we are making the call |
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1386 await webrtc_call.start() |
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1387 else: |
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1388 # we are receiving the call |
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1389 webrtc_call.sid = call_data.sid |
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1390 if call_data.action_id is not None: |
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1391 await bridge.action_launch( |
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1392 call_data.action_id, |
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1393 data_format.serialise({"cancelled": False}), |
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1394 profile, |
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1395 ) |
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1396 return webrtc_call |
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1397 |
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1398 @property |
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1399 def sid(self) -> str | None: |
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1400 return self.webrtc.sid |
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1401 |
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1402 @sid.setter |
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1403 def sid(self, new_sid: str | None) -> None: |
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1404 self.webrtc.sid = new_sid |
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1405 |
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1406 async def on_ice_candidates_new( |
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1407 self, sid: str, candidates_s: str, profile: str |
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1408 ) -> None: |
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1409 if sid != self.webrtc.sid or profile != self.profile: |
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1410 return |
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1411 self.webrtc.on_ice_candidates_new( |
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1412 data_format.deserialise(candidates_s), |
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1413 ) |
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1414 |
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1415 async def on_call_setup(self, sid: str, setup_data_s: str, profile: str) -> None: |
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1416 if sid != self.webrtc.sid or profile != self.profile: |
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1417 return |
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1418 setup_data = data_format.deserialise(setup_data_s) |
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1419 try: |
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1420 role = setup_data["role"] |
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1421 sdp = setup_data["sdp"] |
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1422 except KeyError: |
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1423 log.error(f"Invalid setup data received: {setup_data}") |
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1424 return |
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1425 if role == "initiator": |
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1426 self.webrtc.on_accepted_call(sdp, profile) |
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1427 elif role == "responder": |
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1428 await self.webrtc.answer_call(sdp, profile) |
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1429 else: |
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1430 log.error(f"Invalid role received during setup: {setup_data}") |
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1431 if self.on_call_setup_cb is not None: |
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1432 await aio.maybe_async(self.on_call_setup_cb(sid, profile)) |
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1433 |
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1434 async def on_call_ended(self, sid: str, data_s: str, profile: str) -> None: |
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1435 if sid != self.webrtc.sid or profile != self.profile: |
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1436 return |
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1437 await self.webrtc.end_call() |
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1438 if self.on_call_ended_cb is not None: |
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1439 await aio.maybe_async(self.on_call_ended_cb(sid, profile)) |
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1440 |
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1441 async def start(self): |
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1442 """Start a call. |
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1443 |
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1444 To be used only if we are initiator |
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1445 """ |
4240
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1446 try: |
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1447 await self.webrtc.setup_call("initiator") |
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1448 except Exception: |
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1449 log.exception("Can't setup call") |
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|
1450 raise |
4233
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|
1451 self.webrtc.start_pipeline() |