Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc.py @ 4232:0fbe5c605eb6
tests (unit/webrtc,XEP-0176, XEP-0234): Fix tests and add webrtc file transfer tests:
fix 441
author | Goffi <goffi@goffi.org> |
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date | Sat, 06 Apr 2024 12:59:50 +0200 |
parents | fe29fbdabce6 |
children | d01b8d002619 |
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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 import gi |
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20 gi.require_versions({ |
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21 "Gst": "1.0", |
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22 "GstWebRTC": "1.0" |
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23 }) |
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24 from gi.repository import Gst, GstWebRTC, GstSdp |
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25 |
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26 from libervia.backend.core import exceptions |
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27 |
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28 try: |
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29 from gi.overrides import Gst as _ |
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30 except ImportError: |
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31 raise exceptions.MissingModule( |
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32 "No GStreamer Python overrides available. Please install relevant packages on " |
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33 "your system (e.g., `python3-gst-1.0` on Debian and derivatives)." |
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34 ) |
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35 import asyncio |
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36 from dataclasses import dataclass |
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37 from datetime import datetime |
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38 import logging |
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39 from random import randint |
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40 import re |
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41 from typing import Callable |
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42 from urllib.parse import quote_plus |
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43 |
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44 from libervia.backend.tools.common import data_format |
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45 from libervia.frontends.tools import aio, display_servers |
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46 |
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47 current_server = display_servers.detect() |
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48 if current_server == display_servers.X11: |
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49 # GSTreamer's ximagesrc documentation asks to run this function |
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50 import ctypes |
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51 ctypes.CDLL('libX11.so.6').XInitThreads() |
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52 |
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53 |
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54 log = logging.getLogger(__name__) |
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55 |
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56 Gst.init(None) |
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57 |
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58 SOURCES_AUTO = "auto" |
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59 SOURCES_TEST = "test" |
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60 SINKS_APP = "app" |
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61 SINKS_AUTO = "auto" |
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62 SINKS_TEST = "test" |
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63 |
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64 |
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65 class ScreenshareError(Exception): |
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66 pass |
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67 |
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68 |
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69 @dataclass |
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70 class AppSinkData: |
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71 local_video_cb: Callable |
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72 remote_video_cb: Callable|None |
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73 |
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74 |
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75 class DesktopPortal: |
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76 |
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77 def __init__(self, webrtc: "WebRTC"): |
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78 import dbus |
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79 from dbus.mainloop.glib import DBusGMainLoop |
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80 # we want monitors + windows, see https://flatpak.github.io/xdg-desktop-portal/docs/doc-org.freedesktop.portal.ScreenCast.html#org-freedesktop-portal-screencast-availablesourcetypes |
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81 self.dbus = dbus |
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82 self.webrtc = webrtc |
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83 self.sources_type = dbus.UInt32(7) |
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84 DBusGMainLoop(set_as_default=True) |
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85 self.session_bus = dbus.SessionBus() |
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86 portal_object = self.session_bus.get_object( |
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87 'org.freedesktop.portal.Desktop', |
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88 '/org/freedesktop/portal/desktop' |
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89 ) |
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90 self.screencast_interface = dbus.Interface( |
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91 portal_object, |
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92 'org.freedesktop.portal.ScreenCast' |
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93 ) |
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94 self.session_interface = None |
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95 self.session_signal = None |
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96 self.handle_counter = 0 |
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97 self.session_handle = None |
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98 self.stream_data: dict|None = None |
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99 |
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100 @property |
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101 def handle_token(self): |
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102 self.handle_counter += 1 |
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103 return f"libervia{self.handle_counter}" |
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104 |
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105 def on_session_closed(self, details: dict) -> None: |
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106 if self.session_interface is not None: |
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107 self.session_interface = None |
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108 self.webrtc.desktop_sharing = False |
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109 if self.session_signal is not None: |
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110 self.session_signal.remove() |
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111 self.session_signal = None |
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112 |
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113 |
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114 async def dbus_call(self, method_name: str, *args) -> dict: |
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115 """Call a screenshare portal method |
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116 |
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117 This method handle the signal response. |
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118 @param method_name: method to call |
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119 @param args: extra args |
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120 `handle_token` will be automatically added to the last arg (option dict) |
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121 @return: method result |
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122 """ |
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123 if self.session_handle is not None: |
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124 self.end_screenshare() |
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125 method = getattr(self.screencast_interface, method_name) |
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126 options = args[-1] |
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127 reply_fut = asyncio.Future() |
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128 signal_fut = asyncio.Future() |
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129 # cf. https://flatpak.github.io/xdg-desktop-portal/docs/doc-org.freedesktop.portal.Request.html |
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130 handle_token = self.handle_token |
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131 sender = self.session_bus.get_unique_name().replace(".", "_")[1:] |
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132 path = f"/org/freedesktop/portal/desktop/request/{sender}/{handle_token}" |
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133 signal_match = None |
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134 |
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135 def on_signal(response, results): |
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136 assert signal_match is not None |
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137 signal_match.remove() |
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138 if response == 0: |
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139 signal_fut.set_result(results) |
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140 elif response == 1: |
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141 signal_fut.set_exception( |
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142 exceptions.CancelError("Cancelled by user.") |
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143 ) |
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144 else: |
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145 signal_fut.set_exception(ScreenshareError( |
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146 f"Can't get signal result" |
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147 )) |
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148 |
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149 signal_match = self.session_bus.add_signal_receiver( |
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150 on_signal, |
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151 signal_name="Response", |
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152 dbus_interface="org.freedesktop.portal.Request", |
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153 path=path |
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154 ) |
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155 |
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156 options["handle_token"] = handle_token |
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157 |
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158 method( |
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159 *args, |
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160 reply_handler=reply_fut.set_result, |
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161 error_handler=reply_fut.set_exception |
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162 ) |
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163 try: |
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164 await reply_fut |
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165 except Exception as e: |
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166 raise ScreenshareError(f"Can't ask screenshare permission: {e}") |
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167 return await signal_fut |
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168 |
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169 async def request_screenshare(self) -> dict: |
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170 session_data = await self.dbus_call( |
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171 "CreateSession", |
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172 { |
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173 "session_handle_token": str(randint(1, 2**32)), |
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174 } |
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175 ) |
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176 try: |
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177 session_handle = session_data["session_handle"] |
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178 except KeyError: |
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179 raise ScreenshareError("Can't get session handle") |
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180 self.session_handle = session_handle |
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181 |
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182 |
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183 await self.dbus_call( |
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184 "SelectSources", |
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185 session_handle, |
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186 { |
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187 "multiple": True, |
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188 "types": self.sources_type, |
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189 "modal": True |
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190 } |
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191 ) |
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192 screenshare_data = await self.dbus_call( |
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193 "Start", |
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194 session_handle, |
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195 "", |
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196 {} |
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197 ) |
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198 |
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199 session_object = self.session_bus.get_object( |
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200 'org.freedesktop.portal.Desktop', |
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201 session_handle |
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202 ) |
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203 self.session_interface = self.dbus.Interface( |
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204 session_object, |
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205 'org.freedesktop.portal.Session' |
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206 ) |
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207 |
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208 self.session_signal = self.session_bus.add_signal_receiver( |
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209 self.on_session_closed, |
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210 signal_name="Closed", |
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211 dbus_interface="org.freedesktop.portal.Session", |
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212 path=session_handle |
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213 ) |
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214 |
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215 try: |
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216 node_id, stream_data = screenshare_data["streams"][0] |
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217 source_type = int(stream_data["source_type"]) |
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218 except (IndexError, KeyError): |
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219 raise ScreenshareError("Can't parse stream data") |
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220 self.stream_data = stream_data = { |
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221 "session_handle": session_handle, |
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222 "node_id": node_id, |
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223 "source_type": source_type |
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224 } |
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225 try: |
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226 height = int(stream_data["size"][0]) |
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227 weight = int(stream_data["size"][1]) |
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228 except (IndexError, KeyError): |
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229 pass |
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230 else: |
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231 stream_data["size"] = (height, weight) |
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232 |
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233 return self.stream_data |
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234 |
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235 def end_screenshare(self) -> None: |
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236 """Close a running screenshare session, if any.""" |
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237 if self.session_interface is None: |
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238 return |
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239 self.session_interface.Close() |
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240 self.on_session_closed({}) |
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241 |
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242 |
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243 class WebRTC: |
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244 """GSTreamer based WebRTC implementation for audio and video communication. |
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245 |
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246 This class encapsulates the WebRTC functionalities required for initiating and |
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247 handling audio and video calls. |
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248 """ |
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249 |
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250 def __init__( |
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251 self, |
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252 bridge, |
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253 profile: str, |
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254 sources: str = SOURCES_AUTO, |
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255 sinks: str = SINKS_AUTO, |
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256 appsink_data: AppSinkData | None = None, |
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257 reset_cb: Callable | None = None, |
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258 merge_pip: bool|None = None, |
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259 target_size: tuple[int, int]|None = None, |
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260 ) -> None: |
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261 """Initializes a new WebRTC instance. |
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262 |
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263 @param bridge: An instance of backend bridge. |
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264 @param profile: Libervia profile. |
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265 @param sources: Which kind of source to use. |
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266 @param sinks: Which kind of sinks to use. |
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267 @param appsink_data: configuration data for appsink (when SINKS_APP is used). Must |
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268 not be used for other sinks. |
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269 @param reset_cb: An optional Callable that is triggered on reset events. Can be |
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270 used to reset UI data on new calls. |
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271 @param merge_pip: A boolean flag indicating whether Picture-in-Picture mode is |
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272 enabled. When PiP is used, local feedback is merged to remote video stream. |
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273 Only one video stream is then produced (the local one). |
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274 If None, PiP mode is selected automatically according to selected sink (it's |
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275 used for SINKS_AUTO only for now). |
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276 @param target_size: Expected size of the final sink stream. Mainly use by composer |
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277 when ``merge_pip`` is set. |
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278 None to autodetect (not real autodetection implemeted yet, default to |
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279 (1280,720)). |
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280 """ |
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281 self.main_loop = asyncio.get_event_loop() |
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282 self.bridge = bridge |
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283 self.profile = profile |
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284 self.pipeline = None |
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285 self._audio_muted = False |
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286 self._video_muted = False |
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287 self._desktop_sharing = False |
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288 self.desktop_sharing_data = None |
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289 self.sources = sources |
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290 self.sinks = sinks |
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291 if target_size is None: |
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292 target_size=(1280, 720) |
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293 self.target_width, self.target_height = target_size |
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294 if merge_pip is None: |
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295 merge_pip = sinks == SINKS_AUTO |
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296 self.merge_pip = merge_pip |
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297 if sinks == SINKS_APP: |
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298 if ( |
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299 merge_pip |
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300 and appsink_data is not None |
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301 and appsink_data.remote_video_cb is not None |
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302 ): |
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303 raise ValueError("Remote_video_cb can't be used when merge_pip is used!") |
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304 self.appsink_data = appsink_data |
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305 elif appsink_data is not None: |
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306 raise exceptions.InternalError( |
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307 "appsink_data can only be used for SINKS_APP sinks" |
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308 ) |
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309 self.reset_cb = reset_cb |
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310 if current_server == display_servers.WAYLAND: |
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311 self.desktop_portal = DesktopPortal(self) |
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312 else: |
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313 self.desktop_portal = None |
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314 self.reset_instance() |
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315 |
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316 @property |
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317 def audio_muted(self): |
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318 return self._audio_muted |
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319 |
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320 @audio_muted.setter |
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321 def audio_muted(self, muted: bool) -> None: |
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322 if muted != self._audio_muted: |
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323 self._audio_muted = muted |
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324 self.on_audio_mute(muted) |
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325 |
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326 @property |
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327 def video_muted(self): |
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328 return self._video_muted |
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329 |
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330 @video_muted.setter |
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331 def video_muted(self, muted: bool) -> None: |
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332 if muted != self._video_muted: |
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333 self._video_muted = muted |
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334 self.on_video_mute(muted) |
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335 |
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336 @property |
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337 def desktop_sharing(self): |
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338 return self._desktop_sharing |
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339 |
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340 @desktop_sharing.setter |
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341 def desktop_sharing(self, active: bool) -> None: |
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342 if active != self._desktop_sharing: |
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343 self._desktop_sharing = active |
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344 self.on_desktop_switch(active) |
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345 try: |
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346 cb = self.bindings["desktop_sharing"] |
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347 except KeyError: |
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348 pass |
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349 else: |
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350 cb(active) |
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351 |
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352 @property |
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353 def sdp_set(self): |
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354 return self._sdp_set |
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355 |
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356 @sdp_set.setter |
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357 def sdp_set(self, is_set: bool): |
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358 self._sdp_set = is_set |
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359 if is_set: |
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360 self.on_ice_candidates_new(self.remote_candidates_buffer) |
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361 for data in self.remote_candidates_buffer.values(): |
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362 data["candidates"].clear() |
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363 |
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364 @property |
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365 def media_types(self): |
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366 if self._media_types is None: |
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367 raise Exception("self._media_types should not be None!") |
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368 return self._media_types |
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369 |
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370 @media_types.setter |
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371 def media_types(self, new_media_types: dict) -> None: |
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372 self._media_types = new_media_types |
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373 self._media_types_inv = {v: k for k, v in new_media_types.items()} |
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374 |
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375 @property |
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376 def media_types_inv(self) -> dict: |
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377 if self._media_types_inv is None: |
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378 raise Exception("self._media_types_inv should not be None!") |
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379 return self._media_types_inv |
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380 |
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381 def bind(self, **kwargs: Callable) -> None: |
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382 self.bindings.clear() |
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383 for key, cb in kwargs.items(): |
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384 if key not in ("desktop_sharing",): |
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385 raise ValueError( |
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386 'Only "desktop_sharing" is currently allowed for binding' |
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387 ) |
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388 self.bindings[key] = cb |
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389 |
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390 |
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391 def generate_dot_file( |
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392 self, |
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393 filename: str = "pipeline", |
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394 details: Gst.DebugGraphDetails = Gst.DebugGraphDetails.ALL, |
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395 with_timestamp: bool = True, |
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396 bin_: Gst.Bin|None = None, |
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397 ) -> None: |
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398 """Generate Dot File for debugging |
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399 |
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400 ``GST_DEBUG_DUMP_DOT_DIR`` environment variable must be set to destination dir. |
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401 ``dot -Tpng -o <filename>.png <filename>.dot`` can be use to convert to a PNG file. |
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402 See |
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403 https://gstreamer.freedesktop.org/documentation/gstreamer/debugutils.html?gi-language=python#GstDebugGraphDetails |
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404 for details. |
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405 |
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406 @param filename: name of the generated file |
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407 @param details: which details to print |
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408 @param with_timestamp: if True, add a timestamp to filename |
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409 @param bin_: which bin to output. By default, the whole pipeline |
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410 (``self.pipeline``) will be used. |
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411 """ |
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412 if bin_ is None: |
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413 bin_ = self.pipeline |
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414 if with_timestamp: |
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415 timestamp = datetime.now().isoformat(timespec='milliseconds') |
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416 filename = f"{timestamp}_filename" |
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417 |
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418 Gst.debug_bin_to_dot_file(bin_, details, filename) |
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419 |
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420 def get_sdp_mline_index(self, media_type: str) -> int: |
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421 """Gets the sdpMLineIndex for a given media type. |
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422 |
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423 @param media_type: The type of the media. |
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424 """ |
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425 for index, m_type in self.media_types.items(): |
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426 if m_type == media_type: |
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427 return index |
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428 raise ValueError(f"Media type '{media_type}' not found") |
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429 |
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430 def _set_media_types(self, offer_sdp: str) -> None: |
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431 """Sets media types from offer SDP |
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432 |
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433 @param offer: RTC session description containing the offer |
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434 """ |
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435 sdp_lines = offer_sdp.splitlines() |
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436 media_types = {} |
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437 mline_index = 0 |
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438 |
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439 for line in sdp_lines: |
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440 if line.startswith("m="): |
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441 media_types[mline_index] = line[2 : line.find(" ")] |
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442 mline_index += 1 |
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443 |
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444 self.media_types = media_types |
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445 |
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446 def _a_call(self, method, *args, **kwargs): |
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447 """Call an async method in main thread""" |
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448 aio.run_from_thread(method, *args, **kwargs, loop=self.main_loop) |
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449 |
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450 def get_payload_types( |
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451 self, sdpmsg, video_encoding: str, audio_encoding: str |
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452 ) -> dict[str, int | None]: |
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453 """Find the payload types for the specified video and audio encoding. |
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454 |
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455 Very simplistically finds the first payload type matching the encoding |
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456 name. More complex applications will want to match caps on |
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457 profile-level-id, packetization-mode, etc. |
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458 """ |
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459 # method coming from gstreamer example (Matthew Waters, Nirbheek Chauhan) at |
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460 # subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py |
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461 video_pt = None |
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462 audio_pt = None |
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463 for i in range(0, sdpmsg.medias_len()): |
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464 media = sdpmsg.get_media(i) |
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465 for j in range(0, media.formats_len()): |
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466 fmt = media.get_format(j) |
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467 if fmt == "webrtc-datachannel": |
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468 continue |
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469 pt = int(fmt) |
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470 caps = media.get_caps_from_media(pt) |
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471 s = caps.get_structure(0) |
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472 encoding_name = s["encoding-name"] |
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473 if video_pt is None and encoding_name == video_encoding: |
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474 video_pt = pt |
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475 elif audio_pt is None and encoding_name == audio_encoding: |
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476 audio_pt = pt |
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477 return {video_encoding: video_pt, audio_encoding: audio_pt} |
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478 |
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479 def parse_ice_candidate(self, candidate_string): |
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480 """Parses the ice candidate string. |
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481 |
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482 @param candidate_string: The ice candidate string to be parsed. |
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483 """ |
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484 pattern = re.compile( |
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485 r"candidate:(?P<foundation>\S+) (?P<component_id>\d+) (?P<transport>\S+) " |
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486 r"(?P<priority>\d+) (?P<address>\S+) (?P<port>\d+) typ " |
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487 r"(?P<type>\S+)(?: raddr (?P<rel_addr>\S+) rport " |
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488 r"(?P<rel_port>\d+))?(?: generation (?P<generation>\d+))?" |
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489 ) |
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490 match = pattern.match(candidate_string) |
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491 if match: |
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492 candidate_dict = match.groupdict() |
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493 |
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494 # Apply the correct types to the dictionary values |
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495 candidate_dict["component_id"] = int(candidate_dict["component_id"]) |
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496 candidate_dict["priority"] = int(candidate_dict["priority"]) |
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497 candidate_dict["port"] = int(candidate_dict["port"]) |
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498 |
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499 if candidate_dict["rel_port"]: |
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500 candidate_dict["rel_port"] = int(candidate_dict["rel_port"]) |
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501 |
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502 if candidate_dict["generation"]: |
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503 candidate_dict["generation"] = candidate_dict["generation"] |
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504 |
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505 # Remove None values |
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506 return {k: v for k, v in candidate_dict.items() if v is not None} |
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507 else: |
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508 log.warning(f"can't parse candidate: {candidate_string!r}") |
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509 return None |
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510 |
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511 def build_ice_candidate(self, parsed_candidate): |
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512 """Builds ICE candidate |
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513 |
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514 @param parsed_candidate: Dictionary containing parsed ICE candidate |
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515 """ |
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516 base_format = ( |
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517 "candidate:{foundation} {component_id} {transport} {priority} " |
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518 "{address} {port} typ {type}" |
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519 ) |
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520 |
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521 if parsed_candidate.get("rel_addr") and parsed_candidate.get("rel_port"): |
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522 base_format += " raddr {rel_addr} rport {rel_port}" |
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523 |
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524 if parsed_candidate.get("generation"): |
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525 base_format += " generation {generation}" |
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526 |
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527 return base_format.format(**parsed_candidate) |
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528 |
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529 def extract_ufrag_pwd(self, sdp: str) -> tuple[str, str]: |
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530 """Retrieves ICE password and user fragment for SDP offer. |
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531 |
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532 @param sdp: The Session Description Protocol offer string. |
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533 @return: ufrag and pwd |
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534 @raise ValueError: Can't extract ufrag and password |
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535 """ |
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536 ufrag_line = re.search(r"ice-ufrag:(\S+)", sdp) |
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537 pwd_line = re.search(r"ice-pwd:(\S+)", sdp) |
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538 |
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539 if ufrag_line and pwd_line: |
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540 ufrag = self.ufrag = ufrag_line.group(1) |
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541 pwd = self.pwd = pwd_line.group(1) |
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542 return ufrag, pwd |
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543 else: |
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544 log.error(f"SDP with missing ice-ufrag or ice-pwd:\n{sdp}") |
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545 raise ValueError("Can't extract ice-ufrag and ice-pwd from SDP") |
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546 |
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547 def reset_instance(self): |
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548 """Inits or resets the instance variables to their default state.""" |
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549 self.role: str | None = None |
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550 if self.pipeline is not None: |
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551 self.pipeline.set_state(Gst.State.NULL) |
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552 self.pipeline = None |
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553 self._remote_video_pad = None |
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554 self.sid: str | None = None |
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555 self.offer: str | None = None |
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556 self.local_candidates_buffer = {} |
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557 self.ufrag: str | None = None |
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558 self.pwd: str | None = None |
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559 self.callee: str | None = None |
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560 self._media_types = None |
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561 self._media_types_inv = None |
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562 self._sdp_set: bool = False |
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563 self.remote_candidates_buffer: dict[str, dict[str, list]] = { |
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564 "audio": {"candidates": []}, |
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565 "video": {"candidates": []}, |
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566 } |
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567 self._media_types = None |
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568 self._media_types_inv = None |
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569 self.audio_valve = None |
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570 self.video_valve = None |
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571 if self.desktop_portal is not None: |
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572 self.desktop_portal.end_screenshare() |
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573 self.desktop_sharing = False |
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574 self.desktop_sink_pad = None |
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575 self.bindings = {} |
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576 if self.reset_cb is not None: |
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577 self.reset_cb() |
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578 |
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579 |
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580 async def setup_call( |
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581 self, |
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582 role: str, |
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583 audio_pt: int | None = 96, |
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584 video_pt: int | None = 97, |
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585 ) -> None: |
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586 """Sets up the call. |
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587 |
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588 This method establishes the Gstreamer pipeline for audio and video communication. |
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589 The method also manages STUN and TURN server configurations, signal watchers, and |
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590 various connection handlers for the webrtcbin. |
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591 |
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592 @param role: The role for the call, either 'initiator' or 'responder'. |
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593 @param audio_pt: The payload type for the audio stream. |
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594 @param video_pt: The payload type for the video stream |
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595 |
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596 @raises NotImplementedError: If audio_pt or video_pt is set to None. |
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597 @raises AssertionError: If the role is not 'initiator' or 'responder'. |
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598 """ |
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599 assert role in ("initiator", "responder") |
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600 self.role = role |
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601 if audio_pt is None or video_pt is None: |
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602 raise NotImplementedError("None value is not handled yet") |
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603 |
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604 if self.sources == SOURCES_AUTO: |
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605 video_source_elt = "v4l2src" |
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606 audio_source_elt = "pulsesrc" |
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607 elif self.sources == SOURCES_TEST: |
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608 video_source_elt = "videotestsrc is-live=true pattern=ball" |
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609 audio_source_elt = "audiotestsrc" |
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610 else: |
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611 raise exceptions.InternalError(f'Unknown "sources" value: {self.sources!r}') |
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612 |
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613 |
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614 if self.sinks == SINKS_APP: |
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615 local_video_sink_elt = ( |
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616 "appsink name=local_video_sink emit-signals=true drop=true max-buffers=1 " |
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617 "sync=True" |
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618 ) |
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619 elif self.sinks == SINKS_AUTO: |
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620 local_video_sink_elt = "autovideosink" |
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621 else: |
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622 raise exceptions.InternalError(f"Unknown sinks value {self.sinks!r}") |
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623 |
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624 if self.merge_pip: |
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625 extra_elt = ( |
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626 "compositor name=compositor background=black " |
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627 f"! video/x-raw,width={self.target_width},height={self.target_height}," |
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628 "framerate=30/1 " |
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629 f"! {local_video_sink_elt}" |
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630 ) |
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631 local_video_sink_elt = "compositor.sink_1" |
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632 else: |
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633 extra_elt = "" |
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634 |
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635 self.gst_pipe_desc = f""" |
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636 webrtcbin latency=100 name=sendrecv bundle-policy=max-bundle |
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637 |
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638 input-selector name=video_selector |
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|
639 ! videorate |
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640 ! video/x-raw,framerate=30/1 |
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641 ! tee name=t |
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|
642 |
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|
643 {extra_elt} |
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|
644 |
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645 {video_source_elt} name=video_src ! queue leaky=downstream ! video_selector. |
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646 videotestsrc name=muted_src is-live=true pattern=black ! queue leaky=downstream ! video_selector. |
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647 |
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|
648 t. |
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649 ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream |
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650 ! videoconvert |
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651 ! vp8enc deadline=1 keyframe-max-dist=60 |
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652 ! rtpvp8pay picture-id-mode=15-bit |
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653 ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt} |
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654 ! sendrecv. |
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655 |
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|
656 t. |
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|
657 ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream |
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658 ! videoconvert |
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659 ! {local_video_sink_elt} |
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660 |
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661 {audio_source_elt} name=audio_src |
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662 ! valve |
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663 ! queue max-size-buffers=10 max-size-time=0 max-size-bytes=0 leaky=downstream |
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664 ! audioconvert |
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665 ! audioresample |
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666 ! opusenc audio-type=voice |
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667 ! rtpopuspay |
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668 ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} |
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669 ! sendrecv. |
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670 """ |
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671 |
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672 log.debug(f"Gstreamer pipeline: {self.gst_pipe_desc}") |
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673 |
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674 # Create the pipeline |
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675 try: |
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676 self.pipeline = Gst.parse_launch(self.gst_pipe_desc) |
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677 except Exception: |
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678 log.exception("Can't parse pipeline") |
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679 self.pipeline = None |
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680 if not self.pipeline: |
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681 raise exceptions.InternalError("Failed to create Gstreamer pipeline.") |
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682 |
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683 self.webrtcbin = self.pipeline.get_by_name("sendrecv") |
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684 self.video_src = self.pipeline.get_by_name("video_src") |
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685 self.muted_src = self.pipeline.get_by_name("muted_src") |
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686 self.video_selector = self.pipeline.get_by_name("video_selector") |
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687 self.audio_valve = self.pipeline.get_by_name("audio_valve") |
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688 |
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689 if self.video_muted: |
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690 self.on_video_mute(True) |
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691 if self.audio_muted: |
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692 self.on_audio_mute(True) |
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693 |
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694 # set STUN and TURN servers |
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|
695 external_disco = data_format.deserialise( |
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|
696 await self.bridge.external_disco_get("", self.profile), type_check=list |
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|
697 ) |
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|
698 |
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|
699 for server in external_disco: |
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700 if server["type"] == "stun": |
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|
701 if server["transport"] == "tcp": |
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702 log.info( |
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|
703 "ignoring TCP STUN server, GStreamer only support one STUN server" |
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|
704 ) |
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|
705 url = f"stun://{server['host']}:{server['port']}" |
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706 log.debug(f"adding stun server: {url}") |
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|
707 self.webrtcbin.set_property("stun-server", url) |
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|
708 elif server["type"] == "turn": |
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|
709 url = "{scheme}://{username}:{password}@{host}:{port}".format( |
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710 scheme="turns" if server["transport"] == "tcp" else "turn", |
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711 username=quote_plus(server["username"]), |
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|
712 password=quote_plus(server["password"]), |
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|
713 host=server["host"], |
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|
714 port=server["port"], |
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|
715 ) |
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|
716 log.debug(f"adding turn server: {url}") |
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|
717 |
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|
718 if not self.webrtcbin.emit("add-turn-server", url): |
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|
719 log.warning(f"Erreur while adding TURN server {url}") |
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|
720 |
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|
721 # local video feedback |
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|
722 if self.sinks == SINKS_APP: |
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|
723 assert self.appsink_data is not None |
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|
724 local_video_sink = self.pipeline.get_by_name("local_video_sink") |
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|
725 local_video_sink.set_property("emit-signals", True) |
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|
726 local_video_sink.connect("new-sample", self.appsink_data.local_video_cb) |
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|
727 local_video_sink_caps = Gst.Caps.from_string(f"video/x-raw,format=RGB") |
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|
728 local_video_sink.set_property("caps", local_video_sink_caps) |
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|
729 |
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|
730 # Create bus and associate signal watchers |
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|
731 self.bus = self.pipeline.get_bus() |
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|
732 if not self.bus: |
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|
733 log.error("Failed to get bus from pipeline.") |
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|
734 return |
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|
735 |
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|
736 self.bus.add_signal_watch() |
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737 self.webrtcbin.connect("pad-added", self.on_pad_added) |
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|
738 self.bus.connect("message::error", self.on_bus_error) |
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|
739 self.bus.connect("message::eos", self.on_bus_eos) |
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740 self.webrtcbin.connect("on-negotiation-needed", self.on_negotiation_needed) |
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|
741 self.webrtcbin.connect("on-ice-candidate", self.on_ice_candidate) |
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|
742 self.webrtcbin.connect( |
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|
743 "notify::ice-gathering-state", self.on_ice_gathering_state_change |
6745c6bd4c7a
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|
744 ) |
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|
745 self.webrtcbin.connect( |
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|
746 "notify::ice-connection-state", self.on_ice_connection_state |
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|
747 ) |
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|
748 |
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|
749 def start_pipeline(self) -> None: |
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|
750 """Starts the GStreamer pipeline.""" |
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|
751 log.debug("starting the pipeline") |
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|
752 self.pipeline.set_state(Gst.State.PLAYING) |
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|
753 |
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|
754 def on_negotiation_needed(self, webrtc): |
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|
755 """Initiate SDP offer when negotiation is needed.""" |
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|
756 log.debug("Negotiation needed.") |
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|
757 if self.role == "initiator": |
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|
758 log.debug("Creating offer…") |
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|
759 promise = Gst.Promise.new_with_change_func(self.on_offer_created) |
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|
760 self.webrtcbin.emit("create-offer", None, promise) |
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|
761 |
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|
762 def on_offer_created(self, promise): |
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|
763 """Callback for when SDP offer is created.""" |
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|
764 log.info("on_offer_created called") |
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|
765 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
766 reply = promise.get_reply() |
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|
767 if reply is None: |
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|
768 log.error("Promise reply is None. Offer creation might have failed.") |
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|
769 return |
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|
770 offer = reply["offer"] |
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|
771 self.offer = offer.sdp.as_text() |
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|
772 log.info(f"SDP offer created: \n{self.offer}") |
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|
773 self._set_media_types(self.offer) |
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|
774 promise = Gst.Promise.new() |
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775 self.webrtcbin.emit("set-local-description", offer, promise) |
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|
776 promise.interrupt() |
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|
777 self._a_call(self._start_call) |
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|
778 |
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|
779 def on_answer_set(self, promise): |
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780 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
781 |
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|
782 def on_answer_created(self, promise, _, __): |
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|
783 """Callback for when SDP answer is created.""" |
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|
784 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
785 reply = promise.get_reply() |
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786 answer = reply["answer"] |
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|
787 promise = Gst.Promise.new() |
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788 self.webrtcbin.emit("set-local-description", answer, promise) |
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789 promise.interrupt() |
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|
790 answer_sdp = answer.sdp.as_text() |
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791 log.info(f"SDP answer set: \n{answer_sdp}") |
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792 self.sdp_set = True |
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793 self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile) |
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|
794 |
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|
795 def on_offer_set(self, promise): |
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796 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
797 promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None) |
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798 self.webrtcbin.emit("create-answer", None, promise) |
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|
799 |
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800 def link_element_or_pad( |
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|
801 self, source: Gst.Element, dest: Gst.Element | Gst.Pad |
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802 ) -> bool: |
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|
803 """Check if dest is a pad or an element, and link appropriately""" |
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|
804 src_pad = source.get_static_pad("src") |
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|
805 |
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|
806 if isinstance(dest, Gst.Pad): |
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|
807 # If the dest is a pad, link directly |
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|
808 if not src_pad.link(dest) == Gst.PadLinkReturn.OK: |
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|
809 log.error( |
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|
810 "Failed to link 'conv' to the compositor's newly requested pad!" |
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changeset
|
811 ) |
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|
812 return False |
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|
813 elif isinstance(dest, Gst.Element): |
6745c6bd4c7a
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|
814 return source.link(dest) |
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|
815 else: |
6745c6bd4c7a
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|
816 log.error(f"Unexpected type for dest: {type(sink)}") |
6745c6bd4c7a
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|
817 return False |
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|
818 |
6745c6bd4c7a
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|
819 return True |
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|
820 |
6745c6bd4c7a
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|
821 def scaled_dimensions( |
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|
822 self, original_width: int, original_height: int, max_width: int, max_height: int |
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|
823 ) -> tuple[int, int]: |
6745c6bd4c7a
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|
824 """Calculates the scaled dimensions preserving aspect ratio. |
6745c6bd4c7a
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|
825 |
6745c6bd4c7a
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|
826 @param original_width: Original width of the video stream. |
6745c6bd4c7a
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|
827 @param original_height: Original height of the video stream. |
6745c6bd4c7a
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|
828 @param max_width: Maximum desired width for the scaled video. |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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diff
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|
829 @param max_height: Maximum desired height for the scaled video. |
6745c6bd4c7a
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|
830 @return: The width and height of the scaled video. |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
831 """ |
6745c6bd4c7a
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|
832 aspect_ratio = original_width / original_height |
6745c6bd4c7a
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|
833 new_width = int(max_height * aspect_ratio) |
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|
834 |
6745c6bd4c7a
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|
835 if new_width <= max_width: |
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|
836 return new_width, max_height |
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|
837 |
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|
838 new_height = int(max_width / aspect_ratio) |
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|
839 return max_width, new_height |
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|
840 |
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|
841 def on_remote_decodebin_stream(self, _, pad: Gst.Pad) -> None: |
6745c6bd4c7a
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|
842 """Handle the stream from the remote decodebin. |
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|
843 |
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|
844 This method processes the incoming stream from the remote decodebin, determining |
6745c6bd4c7a
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|
845 whether it's video or audio. It then sets up the appropriate GStreamer elements |
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|
846 for video/audio processing and adds them to the pipeline. |
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|
847 |
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|
848 @param pad: The Gst.Pad from the remote decodebin producing the stream. |
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|
849 """ |
6745c6bd4c7a
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|
850 assert self.pipeline is not None |
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|
851 if not pad.has_current_caps(): |
6745c6bd4c7a
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parents:
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|
852 log.error(f"{pad} has no caps, ignoring") |
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parents:
diff
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|
853 return |
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parents:
diff
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|
854 |
6745c6bd4c7a
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parents:
diff
changeset
|
855 caps = pad.get_current_caps() |
6745c6bd4c7a
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parents:
diff
changeset
|
856 assert len(caps) |
6745c6bd4c7a
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diff
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|
857 s = caps[0] |
6745c6bd4c7a
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diff
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|
858 name = s.get_name() |
6745c6bd4c7a
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parents:
diff
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|
859 log.debug(f"====> NAME START: {name}") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
860 |
6745c6bd4c7a
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|
861 q = Gst.ElementFactory.make("queue") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
862 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
863 if name.startswith("video"): |
6745c6bd4c7a
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|
864 log.debug("===> VIDEO OK") |
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frontends (tools): webrtc implementation:
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|
865 |
6745c6bd4c7a
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|
866 self._remote_video_pad = pad |
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|
867 |
6745c6bd4c7a
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|
868 # Check and log the original size of the video |
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869 width = self.target_width |
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870 height = self.target_height |
4139
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|
871 log.info(f"Original video size: {width}x{height}") |
6745c6bd4c7a
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parents:
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|
872 |
6745c6bd4c7a
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|
873 # This is a fix for an issue found with Movim on desktop: a non standard |
6745c6bd4c7a
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parents:
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|
874 # resolution is used (990x557) resulting in bad alignement and no color in |
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|
875 # rendered image |
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|
876 adjust_resolution = width % 4 != 0 or height % 4 != 0 |
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parents:
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|
877 if adjust_resolution: |
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|
878 log.info("non standard resolution, we need to adjust size") |
6745c6bd4c7a
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|
879 width = (width + 3) // 4 * 4 |
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|
880 height = (height + 3) // 4 * 4 |
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|
881 log.info(f"Adjusted video size: {width}x{height}") |
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parents:
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|
882 |
6745c6bd4c7a
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|
883 conv = Gst.ElementFactory.make("videoconvert") |
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|
884 if self.merge_pip: |
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|
885 # with ``merge_pip`` set, we plug the remote stream to the composer |
4139
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|
886 compositor = self.pipeline.get_by_name("compositor") |
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|
887 |
6745c6bd4c7a
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|
888 sink1_pad = compositor.get_static_pad("sink_1") |
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frontends (tools): webrtc implementation:
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parents:
diff
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|
889 |
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frontends (tools): webrtc implementation:
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diff
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|
890 local_width, local_height = self.scaled_dimensions( |
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frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
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|
891 sink1_pad.get_property("width"), |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
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|
892 sink1_pad.get_property("height"), |
6745c6bd4c7a
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|
893 width // 3, |
6745c6bd4c7a
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diff
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|
894 height // 3, |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
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|
895 ) |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
896 |
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|
897 sink1_pad.set_property("xpos", width - local_width) |
6745c6bd4c7a
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diff
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|
898 sink1_pad.set_property("ypos", height - local_height) |
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|
899 sink1_pad.set_property("width", local_width) |
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diff
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|
900 sink1_pad.set_property("height", local_height) |
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|
901 sink1_pad.set_property("sizing-policy", 1) |
4139
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|
902 sink1_pad.set_property("zorder", 1) |
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diff
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|
903 |
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|
904 # Request a new pad for the remote stream |
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|
905 sink_pad_template = compositor.get_pad_template("sink_%u") |
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|
906 remote_video_sink = compositor.request_pad(sink_pad_template, None, None) |
6745c6bd4c7a
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|
907 remote_video_sink.set_property("zorder", 0) |
4209
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|
908 remote_video_sink.set_property("width", width) |
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|
909 remote_video_sink.set_property("height", height) |
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|
910 remote_video_sink.set_property("sizing-policy", 1) |
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|
911 elif self.sinks == SINKS_APP: |
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|
912 # ``app`` sink without ``self.merge_pip`` set, be create the sink and |
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|
913 # connect it to the ``remote_video_cb``. |
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|
914 assert self.appsink_data is not None |
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|
915 remote_video_sink = Gst.ElementFactory.make("appsink") |
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|
916 |
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|
917 remote_video_caps = Gst.Caps.from_string("video/x-raw,format=RGB") |
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|
918 remote_video_sink.set_property("caps", remote_video_caps) |
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|
919 |
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|
920 remote_video_sink.set_property("emit-signals", True) |
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|
921 remote_video_sink.set_property("drop", True) |
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|
922 remote_video_sink.set_property("max-buffers", 1) |
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|
923 remote_video_sink.set_property("sync", True) |
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|
924 remote_video_sink.connect("new-sample", self.appsink_data.remote_video_cb) |
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|
925 self.pipeline.add(remote_video_sink) |
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|
926 elif self.sinks == SINKS_AUTO: |
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927 # if ``self.merge_pip`` is not set, we create a dedicated |
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928 # ``autovideosink`` for remote stream. |
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929 remote_video_sink = Gst.ElementFactory.make("autovideosink") |
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930 self.pipeline.add(remote_video_sink) |
4139
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931 else: |
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932 raise exceptions.InternalError(f'Unhandled "sinks" value: {self.sinks!r}') |
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933 |
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934 if adjust_resolution: |
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935 videoscale = Gst.ElementFactory.make("videoscale") |
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936 adjusted_caps = Gst.Caps.from_string( |
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937 f"video/x-raw,width={width},height={height}" |
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938 ) |
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939 capsfilter = Gst.ElementFactory.make("capsfilter") |
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940 capsfilter.set_property("caps", adjusted_caps) |
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941 |
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942 self.pipeline.add(q, conv, videoscale, capsfilter) |
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943 |
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944 |
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945 self.pipeline.sync_children_states() |
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946 ret = pad.link(q.get_static_pad("sink")) |
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947 if ret != Gst.PadLinkReturn.OK: |
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948 log.error(f"Error linking pad: {ret}") |
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949 q.link(conv) |
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950 conv.link(videoscale) |
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951 videoscale.link(capsfilter) |
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952 self.link_element_or_pad(capsfilter.link, remote_video_sink) |
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953 |
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954 else: |
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955 self.pipeline.add(q, conv) |
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956 |
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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 self.link_element_or_pad(conv, remote_video_sink) |
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963 |
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964 elif name.startswith("audio"): |
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965 log.debug("===> Audio OK") |
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966 conv = Gst.ElementFactory.make("audioconvert") |
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967 resample = Gst.ElementFactory.make("audioresample") |
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968 remote_audio_sink = Gst.ElementFactory.make("autoaudiosink") |
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969 self.pipeline.add(q, conv, resample, remote_audio_sink) |
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970 self.pipeline.sync_children_states() |
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971 ret = pad.link(q.get_static_pad("sink")) |
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972 if ret != Gst.PadLinkReturn.OK: |
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973 log.error(f"Error linking pad: {ret}") |
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974 q.link(conv) |
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975 conv.link(resample) |
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976 resample.link(remote_audio_sink) |
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977 |
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978 else: |
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979 log.warning(f"unmanaged name: {name!r}") |
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980 |
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981 def on_pad_added(self, __, pad: Gst.Pad) -> None: |
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982 """Handle the addition of a new pad to the element. |
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983 |
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984 When a new source pad is added to the element, this method creates a decodebin, |
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985 connects it to handle the stream, and links the pad to the decodebin. |
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986 |
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987 @param __: Placeholder for the signal source. Not used in this method. |
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988 @param pad: The newly added pad. |
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989 """ |
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990 log.debug("on_pad_added") |
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991 if pad.direction != Gst.PadDirection.SRC: |
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992 return |
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993 |
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994 decodebin = Gst.ElementFactory.make("decodebin") |
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995 decodebin.connect("pad-added", self.on_remote_decodebin_stream) |
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996 self.pipeline.add(decodebin) |
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997 decodebin.sync_state_with_parent() |
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998 pad.link(decodebin.get_static_pad("sink")) |
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999 |
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1000 async def _start_call(self) -> None: |
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1001 """Initiate the call. |
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1002 |
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1003 Initiates a call with the callee using the stored offer. If there are any buffered |
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1004 local ICE candidates, they are sent as part of the initiation. |
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1005 """ |
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1006 assert self.callee |
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1007 self.sid = await self.bridge.call_start( |
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1008 str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile |
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1009 ) |
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1010 if self.local_candidates_buffer: |
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1011 log.debug( |
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1012 f"sending buffered local ICE candidates: {self.local_candidates_buffer}" |
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1013 ) |
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1014 if self.pwd is None: |
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1015 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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1016 self.extract_ufrag_pwd(sdp) |
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1017 ice_data = {} |
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1018 for media_type, candidates in self.local_candidates_buffer.items(): |
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1019 ice_data[media_type] = { |
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1020 "ufrag": self.ufrag, |
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1021 "pwd": self.pwd, |
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1022 "candidates": candidates, |
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1023 } |
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1024 await self.bridge.ice_candidates_add( |
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1025 self.sid, data_format.serialise(ice_data), self.profile |
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1026 ) |
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1027 self.local_candidates_buffer.clear() |
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1028 |
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1029 def _remote_sdp_set(self, promise) -> None: |
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1030 assert promise.wait() == Gst.PromiseResult.REPLIED |
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1031 self.sdp_set = True |
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1032 |
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1033 def on_accepted_call(self, sdp: str, profile: str) -> None: |
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1034 """Outgoing call has been accepted. |
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1035 |
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1036 @param sdp: The SDP answer string received from the other party. |
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1037 @param profile: Profile used for the call. |
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1038 """ |
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1039 log.debug(f"SDP answer received: \n{sdp}") |
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1040 |
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1041 __, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp) |
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1042 answer = GstWebRTC.WebRTCSessionDescription.new( |
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1043 GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg |
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1044 ) |
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1045 promise = Gst.Promise.new_with_change_func(self._remote_sdp_set) |
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1046 self.webrtcbin.emit("set-remote-description", answer, promise) |
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1047 |
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1048 async def answer_call(self, sdp: str, profile: str) -> None: |
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1049 """Answer an incoming call |
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1050 |
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1051 @param sdp: The SDP offer string received from the initiator. |
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1052 @param profile: Profile used for the call. |
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1053 |
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1054 @raise AssertionError: Raised when either "VP8" or "OPUS" is not present in |
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1055 payload types. |
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1056 """ |
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1057 log.debug(f"SDP offer received: \n{sdp}") |
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1058 self._set_media_types(sdp) |
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1059 __, offer_sdp_msg = GstSdp.SDPMessage.new_from_text(sdp) |
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1060 payload_types = self.get_payload_types( |
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1061 offer_sdp_msg, video_encoding="VP8", audio_encoding="OPUS" |
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1062 ) |
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1063 assert "VP8" in payload_types |
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1064 assert "OPUS" in payload_types |
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1065 await self.setup_call( |
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1066 "responder", audio_pt=payload_types["OPUS"], video_pt=payload_types["VP8"] |
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1067 ) |
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1068 self.start_pipeline() |
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1069 offer = GstWebRTC.WebRTCSessionDescription.new( |
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1070 GstWebRTC.WebRTCSDPType.OFFER, offer_sdp_msg |
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1071 ) |
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1072 promise = Gst.Promise.new_with_change_func(self.on_offer_set) |
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1073 self.webrtcbin.emit("set-remote-description", offer, promise) |
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1074 |
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1075 def on_ice_candidate(self, webrtc, mline_index, candidate_sdp): |
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1076 """Handles the on-ice-candidate signal of webrtcbin. |
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1077 |
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1078 @param webrtc: The webrtcbin element. |
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1079 @param mlineindex: The mline index. |
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1080 @param candidate: The ICE candidate. |
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1081 """ |
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1082 log.debug( |
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1083 f"Local ICE candidate. MLine Index: {mline_index}, Candidate: {candidate_sdp}" |
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1084 ) |
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1085 parsed_candidate = self.parse_ice_candidate(candidate_sdp) |
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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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1092 log.debug("buffering local ICE candidate") |
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1093 self.local_candidates_buffer.setdefault(media_type, []).append( |
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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 |
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1099 ufrag, pwd = self.extract_ufrag_pwd(sdp) |
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1100 ice_data = {"ufrag": ufrag, "pwd": pwd, "candidates": [parsed_candidate]} |
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1101 self._a_call( |
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1102 self.bridge.ice_candidates_add, |
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1103 self.sid, |
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1104 data_format.serialise({media_type: ice_data}), |
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1105 self.profile, |
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1106 ) |
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1107 |
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1108 def on_ice_candidates_new(self, candidates: dict) -> None: |
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1109 """Handle new ICE candidates. |
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1110 |
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1111 @param candidates: A dictionary containing media types ("audio" or "video") as |
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1112 keys and corresponding ICE data as values. |
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1113 |
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1114 @raise exceptions.InternalError: Raised when sdp mline index is not found. |
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1115 """ |
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1116 if not self.sdp_set: |
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1117 log.debug("buffering remote ICE candidate") |
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1118 for media_type in ("audio", "video"): |
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1119 media_candidates = candidates.get(media_type) |
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1120 if media_candidates: |
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1121 buffer = self.remote_candidates_buffer[media_type] |
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1122 buffer["candidates"].extend(media_candidates["candidates"]) |
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1123 return |
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1124 for media_type, ice_data in candidates.items(): |
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1125 for candidate in ice_data["candidates"]: |
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1126 candidate_sdp = self.build_ice_candidate(candidate) |
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1127 try: |
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1128 mline_index = self.get_sdp_mline_index(media_type) |
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1129 except Exception as e: |
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1130 raise exceptions.InternalError(f"Can't find sdp mline index: {e}") |
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1131 self.webrtcbin.emit("add-ice-candidate", mline_index, candidate_sdp) |
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1132 log.debug( |
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1133 f"Remote ICE candidate added. MLine Index: {mline_index}, " |
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1134 f"{candidate_sdp}" |
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1135 ) |
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1136 |
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1137 def on_ice_gathering_state_change(self, pspec, __): |
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1138 state = self.webrtcbin.get_property("ice-gathering-state") |
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1139 log.debug(f"ICE gathering state changed to {state}") |
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1140 |
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1141 def on_ice_connection_state(self, pspec, __): |
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1142 state = self.webrtcbin.props.ice_connection_state |
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1143 if state == GstWebRTC.WebRTCICEConnectionState.FAILED: |
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1144 log.error("ICE connection failed") |
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1145 log.info(f"ICE connection state changed to {state}") |
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1146 |
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1147 def on_bus_error(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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1148 """Handles the GStreamer bus error messages. |
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1149 |
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1150 @param bus: The GStreamer bus. |
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1151 @param message: The error message. |
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1152 """ |
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1153 err, debug = message.parse_error() |
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1154 log.error(f"Error from {message.src.get_name()}: {err.message}") |
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1155 log.error(f"Debugging info: {debug}") |
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1156 |
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1157 def on_bus_eos(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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1158 """Handles the GStreamer bus eos messages. |
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1159 |
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1160 @param bus: The GStreamer bus. |
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1161 @param message: The eos message. |
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1162 """ |
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1163 log.info("End of stream") |
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1164 |
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1165 def on_audio_mute(self, muted: bool) -> None: |
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1166 """Handles (un)muting of audio. |
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1167 |
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1168 @param muted: True if audio is muted. |
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1169 """ |
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1170 if self.audio_valve is not None: |
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1171 self.audio_valve.set_property("drop", muted) |
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1172 state = "muted" if muted else "unmuted" |
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1173 log.info(f"audio is now {state}") |
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1174 |
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1175 def on_video_mute(self, muted: bool) -> None: |
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1176 """Handles (un)muting of video. |
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1177 |
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1178 @param muted: True if video is muted. |
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1179 """ |
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1180 if self.video_selector is not None: |
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1181 current_source = None if muted else "desktop" if self.desktop_sharing else "video" |
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1182 self.switch_video_source(current_source) |
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1183 state = "muted" if muted else "unmuted" |
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1184 log.info(f"Video is now {state}") |
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1185 |
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1186 def on_desktop_switch(self, desktop_active: bool) -> None: |
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1187 """Switches the video source between desktop and video. |
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1188 |
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1189 @param desktop_active: True if desktop must be active. False for video. |
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1190 """ |
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1191 if desktop_active and self.desktop_portal is not None: |
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1192 aio.run_async(self.on_desktop_switch_portal(desktop_active)) |
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1193 else: |
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1194 self.do_desktop_switch(desktop_active) |
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1195 |
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1196 async def on_desktop_switch_portal(self, desktop_active: bool) -> None: |
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1197 """Call freedesktop screenshare portal and the activate the shared stream""" |
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1198 assert self.desktop_portal is not None |
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1199 try: |
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1200 screenshare_data = await self.desktop_portal.request_screenshare() |
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1201 except exceptions.CancelError: |
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1202 self.desktop_sharing = False |
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1203 return |
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1204 self.desktop_sharing_data = { |
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1205 "path": str(screenshare_data["node_id"]) |
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1206 } |
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1207 self.do_desktop_switch(desktop_active) |
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1208 |
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1209 def do_desktop_switch(self, desktop_active: bool) -> None: |
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1210 if self.video_muted: |
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1211 # Update the active source state but do not switch |
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1212 self.desktop_sharing = desktop_active |
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1213 return |
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1214 |
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1215 source = "desktop" if desktop_active else "video" |
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1216 self.switch_video_source(source) |
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1217 self.desktop_sharing = desktop_active |
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1218 |
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1219 def switch_video_source(self, source: str|None) -> None: |
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1220 """Activates the specified source while deactivating the others. |
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1221 |
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1222 @param source: 'desktop', 'video', 'muted' or None for muted source. |
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1223 """ |
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1224 if source is None: |
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1225 source = "muted" |
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1226 if source not in ["video", "muted", "desktop"]: |
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1227 raise ValueError( |
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1228 f"Invalid source: {source!r}, use one of {'video', 'muted', 'desktop'}" |
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1229 ) |
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1230 |
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1231 self.pipeline.set_state(Gst.State.PAUSED) |
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1232 |
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1233 # Create a new desktop source if necessary |
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1234 if source == "desktop": |
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1235 self._setup_desktop_source(self.desktop_sharing_data) |
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1236 |
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1237 # Activate the chosen source and deactivate the others |
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1238 for src_name in ["video", "muted", "desktop"]: |
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1239 src_element = self.pipeline.get_by_name(f"{src_name}_src") |
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1240 if src_name == source: |
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1241 if src_element: |
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1242 src_element.set_state(Gst.State.PLAYING) |
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1243 else: |
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1244 if src_element: |
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1245 if src_name == "desktop": |
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1246 self._remove_desktop_source(src_element) |
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1247 else: |
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1248 src_element.set_state(Gst.State.NULL) |
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1249 |
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1250 # Set the video_selector active pad |
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1251 if source == "desktop": |
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1252 if self.desktop_sink_pad: |
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1253 pad = self.desktop_sink_pad |
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1254 else: |
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1255 log.error(f"No desktop pad available") |
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1256 pad = None |
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1257 else: |
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1258 pad_name = f"sink_{['video', 'muted'].index(source)}" |
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1259 pad = self.video_selector.get_static_pad(pad_name) |
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1260 |
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1261 if pad is not None: |
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1262 self.video_selector.props.active_pad = pad |
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1263 |
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1264 self.pipeline.set_state(Gst.State.PLAYING) |
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1265 |
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1266 def _setup_desktop_source(self, properties: dict[str, object]|None) -> None: |
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1267 """Set up a new desktop source. |
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1268 |
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1269 @param properties: The properties to set on the desktop source. |
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1270 """ |
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1271 source_elt = "ximagesrc" if self.desktop_portal is None else "pipewiresrc" |
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1272 desktop_src = Gst.ElementFactory.make(source_elt, "desktop_src") |
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1273 if properties is None: |
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1274 properties = {} |
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1275 for key, value in properties.items(): |
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1276 log.debug(f"setting {source_elt} property: {key!r}={value!r}") |
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1277 desktop_src.set_property(key, value) |
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1278 video_convert = Gst.ElementFactory.make("videoconvert", "desktop_videoconvert") |
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1279 queue = Gst.ElementFactory.make("queue", "desktop_queue") |
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1280 queue.set_property("leaky", "downstream") |
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1281 |
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1282 self.pipeline.add(desktop_src) |
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1283 self.pipeline.add(video_convert) |
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1284 self.pipeline.add(queue) |
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1285 |
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1286 desktop_src.link(video_convert) |
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1287 video_convert.link(queue) |
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1288 |
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1289 sink_pad_template = self.video_selector.get_pad_template("sink_%u") |
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1290 self.desktop_sink_pad = self.video_selector.request_pad(sink_pad_template, None, None) |
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1291 queue_src_pad = queue.get_static_pad("src") |
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1292 queue_src_pad.link(self.desktop_sink_pad) |
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1293 |
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1294 desktop_src.sync_state_with_parent() |
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1295 video_convert.sync_state_with_parent() |
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1296 queue.sync_state_with_parent() |
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1297 |
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1298 def _remove_desktop_source(self, desktop_src: Gst.Element) -> None: |
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1299 """Remove the desktop source from the pipeline. |
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1300 |
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1301 @param desktop_src: The desktop source to remove. |
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1302 """ |
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1303 # Remove elements for the desktop source |
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1304 video_convert = self.pipeline.get_by_name("desktop_videoconvert") |
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1305 queue = self.pipeline.get_by_name("desktop_queue") |
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1306 |
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1307 if video_convert: |
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1308 video_convert.set_state(Gst.State.NULL) |
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1309 desktop_src.unlink(video_convert) |
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1310 self.pipeline.remove(video_convert) |
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1311 |
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1312 if queue: |
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1313 queue.set_state(Gst.State.NULL) |
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1314 self.pipeline.remove(queue) |
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1315 |
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1316 desktop_src.set_state(Gst.State.NULL) |
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1317 self.pipeline.remove(desktop_src) |
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1318 |
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1319 # Release the pad associated with the desktop source |
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1320 if self.desktop_sink_pad: |
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1321 self.video_selector.release_request_pad(self.desktop_sink_pad) |
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1322 self.desktop_sink_pad = None |
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1323 |
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1324 if self.desktop_portal is not None: |
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1325 self.desktop_portal.end_screenshare() |
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1326 |
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1327 async def end_call(self) -> None: |
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1328 """Stop streaming and clean instance""" |
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1329 self.reset_instance() |