Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc.py @ 4219:1b5cf2ee1d86
plugin XEP-0384, XEP-0391: download missing devices list:
when a peer jid was not in our roster, devices list was not retrieved, resulting in failed
en/decryption. This patch does check it and download missing devices list in necessary.
There is no subscription managed yet, so the list won't be updated in case of new devices,
this should be addressed at some point.
author | Goffi <goffi@goffi.org> |
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date | Tue, 05 Mar 2024 17:31:36 +0100 |
parents | fe29fbdabce6 |
children | d01b8d002619 |
rev | line source |
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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 ) |
6745c6bd4c7a
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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 |
6745c6bd4c7a
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|
724 local_video_sink = self.pipeline.get_by_name("local_video_sink") |
6745c6bd4c7a
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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) |
6745c6bd4c7a
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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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changeset
|
729 |
6745c6bd4c7a
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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.") |
6745c6bd4c7a
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|
734 return |
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|
735 |
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|
736 self.bus.add_signal_watch() |
6745c6bd4c7a
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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( |
6745c6bd4c7a
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|
743 "notify::ice-gathering-state", self.on_ice_gathering_state_change |
6745c6bd4c7a
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|
744 ) |
6745c6bd4c7a
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|
745 self.webrtcbin.connect( |
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diff
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|
746 "notify::ice-connection-state", self.on_ice_connection_state |
6745c6bd4c7a
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|
747 ) |
6745c6bd4c7a
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|
748 |
6745c6bd4c7a
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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 |
6745c6bd4c7a
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|
754 def on_negotiation_needed(self, webrtc): |
6745c6bd4c7a
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|
755 """Initiate SDP offer when negotiation is needed.""" |
6745c6bd4c7a
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|
756 log.debug("Negotiation needed.") |
6745c6bd4c7a
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|
757 if self.role == "initiator": |
6745c6bd4c7a
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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) |
6745c6bd4c7a
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|
760 self.webrtcbin.emit("create-offer", None, promise) |
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changeset
|
761 |
6745c6bd4c7a
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diff
changeset
|
762 def on_offer_created(self, promise): |
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|
763 """Callback for when SDP offer is created.""" |
6745c6bd4c7a
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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.") |
6745c6bd4c7a
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|
769 return |
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|
770 offer = reply["offer"] |
6745c6bd4c7a
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|
771 self.offer = offer.sdp.as_text() |
6745c6bd4c7a
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|
772 log.info(f"SDP offer created: \n{self.offer}") |
6745c6bd4c7a
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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() |
6745c6bd4c7a
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|
791 log.info(f"SDP answer set: \n{answer_sdp}") |
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|
792 self.sdp_set = True |
6745c6bd4c7a
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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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diff
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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") |
6745c6bd4c7a
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|
805 |
6745c6bd4c7a
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|
806 if isinstance(dest, Gst.Pad): |
6745c6bd4c7a
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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!" |
6745c6bd4c7a
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|
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)}") |
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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>
parents:
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|
829 @param max_height: Maximum desired height for the scaled video. |
6745c6bd4c7a
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parents:
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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 |
6745c6bd4c7a
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|
840 |
6745c6bd4c7a
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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
frontends (tools): webrtc implementation:
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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. |
6745c6bd4c7a
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diff
changeset
|
847 |
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diff
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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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Goffi <goffi@goffi.org>
parents:
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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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frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
853 return |
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parents:
diff
changeset
|
854 |
6745c6bd4c7a
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parents:
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|
855 caps = pad.get_current_caps() |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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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
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
859 log.debug(f"====> NAME START: {name}") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
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|
860 |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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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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diff
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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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diff
changeset
|
865 |
6745c6bd4c7a
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|
866 self._remote_video_pad = pad |
6745c6bd4c7a
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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 |
6745c6bd4c7a
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|
875 # rendered image |
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frontends (tools): webrtc implementation:
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diff
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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: |
6745c6bd4c7a
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diff
changeset
|
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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frontends (tools): webrtc implementation:
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parents:
diff
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|
881 log.info(f"Adjusted video size: {width}x{height}") |
6745c6bd4c7a
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parents:
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|
882 |
6745c6bd4c7a
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diff
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|
883 conv = Gst.ElementFactory.make("videoconvert") |
4209
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|
884 if self.merge_pip: |
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4205
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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") |
6745c6bd4c7a
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|
887 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
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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 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
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|
890 local_width, local_height = self.scaled_dimensions( |
6745c6bd4c7a
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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parents:
diff
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|
892 sink1_pad.get_property("height"), |
6745c6bd4c7a
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diff
changeset
|
893 width // 3, |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
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 |
6745c6bd4c7a
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|
897 sink1_pad.set_property("xpos", width - local_width) |
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diff
changeset
|
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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frontends (tools): webrtc implementation:
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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) |
6745c6bd4c7a
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diff
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|
903 |
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diff
changeset
|
904 # Request a new pad for the remote stream |
6745c6bd4c7a
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diff
changeset
|
905 sink_pad_template = compositor.get_pad_template("sink_%u") |
6745c6bd4c7a
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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") |
4139
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|
916 |
4209
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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() |