Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc.py @ 4242:8acf46ed7f36
frontends: remote control implementation:
This is the frontends common part of remote control implementation. It handle the creation
of WebRTC session, and management of inputs. For now the reception use freedesktop.org
Desktop portal, and works mostly with Wayland based Desktop Environments.
rel 436
author | Goffi <goffi@goffi.org> |
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date | Sat, 11 May 2024 13:52:43 +0200 |
parents | 79c8a70e1813 |
children | 0d7bb4df2343 |
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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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|
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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diff
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|
706 ) |
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|
707 |
4240
79c8a70e1813
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|
708 for dc_data in self.dc_data_list: |
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4233
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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 |
79c8a70e1813
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|
714 if isinstance(dc_data, SourcesDataChannel): |
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cli (call, file), frontends: implement webRTC data channel transfer:
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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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diff
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|
717 data_channel_options.set_value("ordered", True) |
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Goffi <goffi@goffi.org>
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|
718 data_channel_options.set_value("protocol", "") |
d01b8d002619
cli (call, file), frontends: implement webRTC data channel transfer:
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diff
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|
719 |
d01b8d002619
cli (call, file), frontends: implement webRTC data channel transfer:
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diff
changeset
|
720 # Create the data channel |
d01b8d002619
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changeset
|
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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parents:
4233
diff
changeset
|
723 "create-data-channel", dc_data.name, data_channel_options |
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|
724 ) |
4240
79c8a70e1813
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|
725 if data_channel is None: |
4233
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diff
changeset
|
726 log.error("Failed to create data channel") |
d01b8d002619
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parents:
4209
diff
changeset
|
727 return |
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|
728 data_channel.connect("on-open", dc_data.dc_open_cb) |
79c8a70e1813
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Goffi <goffi@goffi.org>
parents:
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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) |
79c8a70e1813
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|
731 else: |
79c8a70e1813
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parents:
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|
732 raise ValueError( |
79c8a70e1813
backend, frontend: prepare remote control:
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parents:
4233
diff
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|
733 "Only SourcesDataChannel or SinksDataChannel are allowed." |
79c8a70e1813
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Goffi <goffi@goffi.org>
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|
734 ) |
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|
735 |
4139
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|
736 def start_pipeline(self) -> None: |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
737 """Starts the GStreamer pipeline.""" |
6745c6bd4c7a
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|
738 log.debug("starting the pipeline") |
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|
739 self.pipeline.set_state(Gst.State.PLAYING) |
6745c6bd4c7a
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|
740 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
741 def on_negotiation_needed(self, webrtc): |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
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|
742 """Initiate SDP offer when negotiation is needed.""" |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
743 log.debug("Negotiation needed.") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
744 if self.role == "initiator": |
6745c6bd4c7a
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|
745 log.debug("Creating offer…") |
6745c6bd4c7a
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changeset
|
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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diff
changeset
|
748 |
6745c6bd4c7a
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diff
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|
749 def on_offer_created(self, promise): |
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diff
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|
750 """Callback for when SDP offer is created.""" |
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changeset
|
751 log.info("on_offer_created called") |
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Goffi <goffi@goffi.org>
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changeset
|
752 assert promise.wait() == Gst.PromiseResult.REPLIED |
6745c6bd4c7a
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|
753 reply = promise.get_reply() |
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|
754 if reply is None: |
6745c6bd4c7a
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|
755 log.error("Promise reply is None. Offer creation might have failed.") |
6745c6bd4c7a
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diff
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|
756 return |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
757 offer = reply["offer"] |
6745c6bd4c7a
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|
758 self.offer = offer.sdp.as_text() |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
diff
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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diff
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|
763 promise.interrupt() |
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diff
changeset
|
764 self._a_call(self._start_call) |
6745c6bd4c7a
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diff
changeset
|
765 |
6745c6bd4c7a
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|
766 def on_answer_set(self, promise): |
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changeset
|
767 assert promise.wait() == Gst.PromiseResult.REPLIED |
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diff
changeset
|
768 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
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|
769 def on_answer_created(self, promise, _, __): |
6745c6bd4c7a
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diff
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|
770 """Callback for when SDP answer is created.""" |
6745c6bd4c7a
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diff
changeset
|
771 assert promise.wait() == Gst.PromiseResult.REPLIED |
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diff
changeset
|
772 reply = promise.get_reply() |
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|
773 answer = reply["answer"] |
6745c6bd4c7a
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diff
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|
774 promise = Gst.Promise.new() |
6745c6bd4c7a
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|
775 self.webrtcbin.emit("set-local-description", answer, promise) |
6745c6bd4c7a
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diff
changeset
|
776 promise.interrupt() |
6745c6bd4c7a
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diff
changeset
|
777 answer_sdp = answer.sdp.as_text() |
6745c6bd4c7a
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diff
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|
778 log.info(f"SDP answer set: \n{answer_sdp}") |
6745c6bd4c7a
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|
779 self.sdp_set = True |
6745c6bd4c7a
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changeset
|
780 self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile) |
6745c6bd4c7a
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diff
changeset
|
781 |
6745c6bd4c7a
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diff
changeset
|
782 def on_offer_set(self, promise): |
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diff
changeset
|
783 assert promise.wait() == Gst.PromiseResult.REPLIED |
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diff
changeset
|
784 promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None) |
6745c6bd4c7a
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diff
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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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diff
changeset
|
788 self, source: Gst.Element, dest: Gst.Element | Gst.Pad |
6745c6bd4c7a
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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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changeset
|
791 src_pad = source.get_static_pad("src") |
6745c6bd4c7a
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diff
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|
792 |
6745c6bd4c7a
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diff
changeset
|
793 if isinstance(dest, Gst.Pad): |
6745c6bd4c7a
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diff
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|
794 # If the dest is a pad, link directly |
6745c6bd4c7a
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parents:
diff
changeset
|
795 if not src_pad.link(dest) == Gst.PadLinkReturn.OK: |
6745c6bd4c7a
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|
796 log.error( |
6745c6bd4c7a
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diff
changeset
|
797 "Failed to link 'conv' to the compositor's newly requested pad!" |
6745c6bd4c7a
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diff
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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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parents:
diff
changeset
|
801 return source.link(dest) |
6745c6bd4c7a
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changeset
|
802 else: |
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|
803 log.error(f"Unexpected type for dest: {type(dest)}") |
4139
6745c6bd4c7a
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|
804 return False |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
805 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
806 return True |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
807 |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
808 def scaled_dimensions( |
6745c6bd4c7a
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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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diff
changeset
|
810 ) -> tuple[int, int]: |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
811 """Calculates the scaled dimensions preserving aspect ratio. |
6745c6bd4c7a
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diff
changeset
|
812 |
6745c6bd4c7a
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parents:
diff
changeset
|
813 @param original_width: Original width of the video stream. |
6745c6bd4c7a
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parents:
diff
changeset
|
814 @param original_height: Original height of the video stream. |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
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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parents:
diff
changeset
|
817 @return: The width and height of the scaled video. |
6745c6bd4c7a
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diff
changeset
|
818 """ |
6745c6bd4c7a
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parents:
diff
changeset
|
819 aspect_ratio = original_width / original_height |
6745c6bd4c7a
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parents:
diff
changeset
|
820 new_width = int(max_height * aspect_ratio) |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
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|
821 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
822 if new_width <= max_width: |
6745c6bd4c7a
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diff
changeset
|
823 return new_width, max_height |
6745c6bd4c7a
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824 |
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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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diff
changeset
|
971 When a new source pad is added to the element, this method creates a decodebin, |
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Goffi <goffi@goffi.org>
parents:
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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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parents:
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|
978 if pad.direction != Gst.PadDirection.SRC: |
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parents:
diff
changeset
|
979 return |
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parents:
diff
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|
980 |
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diff
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|
981 decodebin = Gst.ElementFactory.make("decodebin") |
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parents:
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|
982 decodebin.connect("pad-added", self.on_remote_decodebin_stream) |
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parents:
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|
983 self.pipeline.add(decodebin) |
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|
984 decodebin.sync_state_with_parent() |
6745c6bd4c7a
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parents:
diff
changeset
|
985 pad.link(decodebin.get_static_pad("sink")) |
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parents:
diff
changeset
|
986 |
4233
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parents:
4209
diff
changeset
|
987 async def _call_start(self, callee: jid.JID, call_data: dict, profile: str) -> str: |
d01b8d002619
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parents:
4209
diff
changeset
|
988 return await self.bridge.call_start( |
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parents:
4209
diff
changeset
|
989 str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile |
d01b8d002619
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parents:
4209
diff
changeset
|
990 ) |
d01b8d002619
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parents:
4209
diff
changeset
|
991 |
4139
6745c6bd4c7a
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|
992 async def _start_call(self) -> None: |
6745c6bd4c7a
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diff
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|
993 """Initiate the call. |
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parents:
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changeset
|
994 |
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diff
changeset
|
995 Initiates a call with the callee using the stored offer. If there are any buffered |
6745c6bd4c7a
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|
996 local ICE candidates, they are sent as part of the initiation. |
6745c6bd4c7a
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diff
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|
997 """ |
6745c6bd4c7a
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|
998 assert self.callee |
4233
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diff
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|
999 assert self.call_start_cb is not None |
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parents:
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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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parents:
diff
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|
1004 log.debug( |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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changeset
|
1005 f"sending buffered local ICE candidates: {self.local_candidates_buffer}" |
6745c6bd4c7a
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diff
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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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parents:
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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 |
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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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|
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 |
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|
1075 @param webrtc: The webrtcbin element. |
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1076 @param mlineindex: The mline index. |
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|
1077 @param candidate: The ICE candidate. |
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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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|
1094 parsed_candidate |
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|
1095 ) |
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|
1096 else: |
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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( |
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|
1106 self.bridge.ice_candidates_add, |
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|
1107 self.sid, |
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|
1108 data_format.serialise({media_type: ice_data}), |
6745c6bd4c7a
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|
1109 self.profile, |
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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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|
1113 """Handle new ICE candidates. |
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|
1114 |
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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. |
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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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|
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}") |
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|
1144 |
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|
1145 def on_ice_connection_state(self, pspec, __): |
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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") |
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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() |
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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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|
1175 self.audio_valve.set_property("drop", muted) |
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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() |