Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc.py @ 4207:2865e70b0b2c
tests (unit/cli/calls): unit tests for CLI call GUI:
rel 427
author | Goffi <goffi@goffi.org> |
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date | Sun, 11 Feb 2024 23:21:07 +0100 |
parents | 17a8168966f9 |
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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 |
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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 ) -> None: |
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259 self.main_loop = asyncio.get_event_loop() |
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260 self.bridge = bridge |
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261 self.profile = profile |
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262 self.pipeline = None |
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263 self._audio_muted = False |
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264 self._video_muted = False |
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265 self._desktop_sharing = False |
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266 self.desktop_sharing_data = None |
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267 self.sources = sources |
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268 self.sinks = sinks |
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269 if sinks == SINKS_APP: |
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270 self.appsink_data = appsink_data |
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271 elif appsink_data is not None: |
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272 raise exceptions.InternalError( |
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273 "appsink_data can only be used for SINKS_APP sinks" |
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274 ) |
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275 self.reset_cb = reset_cb |
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276 if current_server == display_servers.WAYLAND: |
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277 self.desktop_portal = DesktopPortal(self) |
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278 else: |
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279 self.desktop_portal = None |
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280 self.reset_instance() |
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281 |
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282 @property |
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283 def audio_muted(self): |
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284 return self._audio_muted |
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285 |
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286 @audio_muted.setter |
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287 def audio_muted(self, muted: bool) -> None: |
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288 if muted != self._audio_muted: |
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289 self._audio_muted = muted |
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290 self.on_audio_mute(muted) |
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291 |
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292 @property |
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293 def video_muted(self): |
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294 return self._video_muted |
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295 |
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296 @video_muted.setter |
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297 def video_muted(self, muted: bool) -> None: |
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298 if muted != self._video_muted: |
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299 self._video_muted = muted |
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300 self.on_video_mute(muted) |
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301 |
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302 @property |
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303 def desktop_sharing(self): |
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304 return self._desktop_sharing |
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305 |
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306 @desktop_sharing.setter |
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307 def desktop_sharing(self, active: bool) -> None: |
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308 if active != self._desktop_sharing: |
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309 self._desktop_sharing = active |
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310 self.on_desktop_switch(active) |
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311 try: |
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312 cb = self.bindings["desktop_sharing"] |
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313 except KeyError: |
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314 pass |
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315 else: |
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316 cb(active) |
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317 |
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318 @property |
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319 def sdp_set(self): |
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320 return self._sdp_set |
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321 |
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322 @sdp_set.setter |
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323 def sdp_set(self, is_set: bool): |
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324 self._sdp_set = is_set |
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325 if is_set: |
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326 self.on_ice_candidates_new(self.remote_candidates_buffer) |
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327 for data in self.remote_candidates_buffer.values(): |
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328 data["candidates"].clear() |
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329 |
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330 @property |
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331 def media_types(self): |
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332 if self._media_types is None: |
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333 raise Exception("self._media_types should not be None!") |
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334 return self._media_types |
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335 |
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336 @media_types.setter |
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337 def media_types(self, new_media_types: dict) -> None: |
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338 self._media_types = new_media_types |
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339 self._media_types_inv = {v: k for k, v in new_media_types.items()} |
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340 |
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341 @property |
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342 def media_types_inv(self) -> dict: |
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343 if self._media_types_inv is None: |
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344 raise Exception("self._media_types_inv should not be None!") |
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345 return self._media_types_inv |
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346 |
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347 def bind(self, **kwargs: Callable) -> None: |
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348 self.bindings.clear() |
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349 for key, cb in kwargs.items(): |
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350 if key not in ("desktop_sharing",): |
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351 raise ValueError( |
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352 'Only "desktop_sharing" is currently allowed for binding' |
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353 ) |
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354 self.bindings[key] = cb |
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355 |
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356 |
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357 def generate_dot_file( |
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358 self, |
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359 filename: str = "pipeline", |
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360 details: Gst.DebugGraphDetails = Gst.DebugGraphDetails.ALL, |
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361 with_timestamp: bool = True, |
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362 bin_: Gst.Bin|None = None, |
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363 ) -> None: |
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364 """Generate Dot File for debugging |
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365 |
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366 ``GST_DEBUG_DUMP_DOT_DIR`` environment variable must be set to destination dir. |
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367 ``dot -Tpng -o <filename>.png <filename>.dot`` can be use to convert to a PNG file. |
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368 See |
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369 https://gstreamer.freedesktop.org/documentation/gstreamer/debugutils.html?gi-language=python#GstDebugGraphDetails |
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370 for details. |
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371 |
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372 @param filename: name of the generated file |
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373 @param details: which details to print |
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374 @param with_timestamp: if True, add a timestamp to filename |
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375 @param bin_: which bin to output. By default, the whole pipeline |
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376 (``self.pipeline``) will be used. |
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377 """ |
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378 if bin_ is None: |
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379 bin_ = self.pipeline |
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380 if with_timestamp: |
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381 timestamp = datetime.now().isoformat(timespec='milliseconds') |
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382 filename = f"{timestamp}_filename" |
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383 |
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384 Gst.debug_bin_to_dot_file(bin_, details, filename) |
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385 |
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386 def get_sdp_mline_index(self, media_type: str) -> int: |
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387 """Gets the sdpMLineIndex for a given media type. |
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388 |
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389 @param media_type: The type of the media. |
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390 """ |
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391 for index, m_type in self.media_types.items(): |
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392 if m_type == media_type: |
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393 return index |
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394 raise ValueError(f"Media type '{media_type}' not found") |
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395 |
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396 def _set_media_types(self, offer_sdp: str) -> None: |
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397 """Sets media types from offer SDP |
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398 |
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399 @param offer: RTC session description containing the offer |
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400 """ |
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401 sdp_lines = offer_sdp.splitlines() |
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402 media_types = {} |
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403 mline_index = 0 |
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404 |
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405 for line in sdp_lines: |
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406 if line.startswith("m="): |
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407 media_types[mline_index] = line[2 : line.find(" ")] |
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408 mline_index += 1 |
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409 |
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410 self.media_types = media_types |
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411 |
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412 def _a_call(self, method, *args, **kwargs): |
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413 """Call an async method in main thread""" |
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414 aio.run_from_thread(method, *args, **kwargs, loop=self.main_loop) |
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415 |
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416 def get_payload_types( |
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417 self, sdpmsg, video_encoding: str, audio_encoding: str |
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418 ) -> dict[str, int | None]: |
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419 """Find the payload types for the specified video and audio encoding. |
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420 |
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421 Very simplistically finds the first payload type matching the encoding |
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422 name. More complex applications will want to match caps on |
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423 profile-level-id, packetization-mode, etc. |
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424 """ |
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425 # method coming from gstreamer example (Matthew Waters, Nirbheek Chauhan) at |
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426 # subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py |
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427 video_pt = None |
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428 audio_pt = None |
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429 for i in range(0, sdpmsg.medias_len()): |
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430 media = sdpmsg.get_media(i) |
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431 for j in range(0, media.formats_len()): |
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432 fmt = media.get_format(j) |
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433 if fmt == "webrtc-datachannel": |
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434 continue |
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435 pt = int(fmt) |
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436 caps = media.get_caps_from_media(pt) |
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437 s = caps.get_structure(0) |
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438 encoding_name = s["encoding-name"] |
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439 if video_pt is None and encoding_name == video_encoding: |
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440 video_pt = pt |
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441 elif audio_pt is None and encoding_name == audio_encoding: |
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442 audio_pt = pt |
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443 return {video_encoding: video_pt, audio_encoding: audio_pt} |
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444 |
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445 def parse_ice_candidate(self, candidate_string): |
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446 """Parses the ice candidate string. |
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447 |
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448 @param candidate_string: The ice candidate string to be parsed. |
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449 """ |
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450 pattern = re.compile( |
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451 r"candidate:(?P<foundation>\S+) (?P<component_id>\d+) (?P<transport>\S+) " |
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452 r"(?P<priority>\d+) (?P<address>\S+) (?P<port>\d+) typ " |
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453 r"(?P<type>\S+)(?: raddr (?P<rel_addr>\S+) rport " |
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454 r"(?P<rel_port>\d+))?(?: generation (?P<generation>\d+))?" |
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455 ) |
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456 match = pattern.match(candidate_string) |
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457 if match: |
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458 candidate_dict = match.groupdict() |
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459 |
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460 # Apply the correct types to the dictionary values |
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461 candidate_dict["component_id"] = int(candidate_dict["component_id"]) |
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462 candidate_dict["priority"] = int(candidate_dict["priority"]) |
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463 candidate_dict["port"] = int(candidate_dict["port"]) |
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464 |
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465 if candidate_dict["rel_port"]: |
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466 candidate_dict["rel_port"] = int(candidate_dict["rel_port"]) |
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467 |
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468 if candidate_dict["generation"]: |
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469 candidate_dict["generation"] = candidate_dict["generation"] |
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470 |
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471 # Remove None values |
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472 return {k: v for k, v in candidate_dict.items() if v is not None} |
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473 else: |
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474 log.warning(f"can't parse candidate: {candidate_string!r}") |
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475 return None |
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476 |
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477 def build_ice_candidate(self, parsed_candidate): |
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478 """Builds ICE candidate |
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479 |
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480 @param parsed_candidate: Dictionary containing parsed ICE candidate |
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481 """ |
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482 base_format = ( |
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483 "candidate:{foundation} {component_id} {transport} {priority} " |
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484 "{address} {port} typ {type}" |
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485 ) |
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486 |
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487 if parsed_candidate.get("rel_addr") and parsed_candidate.get("rel_port"): |
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488 base_format += " raddr {rel_addr} rport {rel_port}" |
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489 |
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490 if parsed_candidate.get("generation"): |
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491 base_format += " generation {generation}" |
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492 |
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493 return base_format.format(**parsed_candidate) |
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494 |
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495 def extract_ufrag_pwd(self, sdp: str) -> tuple[str, str]: |
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496 """Retrieves ICE password and user fragment for SDP offer. |
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497 |
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498 @param sdp: The Session Description Protocol offer string. |
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499 @return: ufrag and pwd |
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500 @raise ValueError: Can't extract ufrag and password |
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501 """ |
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502 ufrag_line = re.search(r"ice-ufrag:(\S+)", sdp) |
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503 pwd_line = re.search(r"ice-pwd:(\S+)", sdp) |
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504 |
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505 if ufrag_line and pwd_line: |
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506 ufrag = self.ufrag = ufrag_line.group(1) |
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507 pwd = self.pwd = pwd_line.group(1) |
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508 return ufrag, pwd |
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509 else: |
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510 log.error(f"SDP with missing ice-ufrag or ice-pwd:\n{sdp}") |
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511 raise ValueError("Can't extract ice-ufrag and ice-pwd from SDP") |
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512 |
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513 def reset_instance(self): |
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514 """Inits or resets the instance variables to their default state.""" |
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515 self.role: str | None = None |
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516 if self.pipeline is not None: |
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517 self.pipeline.set_state(Gst.State.NULL) |
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518 self.pipeline = None |
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519 self._remote_video_pad = None |
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520 self.sid: str | None = None |
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521 self.offer: str | None = None |
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522 self.local_candidates_buffer = {} |
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523 self.ufrag: str | None = None |
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524 self.pwd: str | None = None |
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525 self.callee: str | None = None |
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526 self._media_types = None |
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527 self._media_types_inv = None |
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528 self._sdp_set: bool = False |
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529 self.remote_candidates_buffer: dict[str, dict[str, list]] = { |
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530 "audio": {"candidates": []}, |
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531 "video": {"candidates": []}, |
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532 } |
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533 self._media_types = None |
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534 self._media_types_inv = None |
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535 self.audio_valve = None |
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536 self.video_valve = None |
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537 if self.desktop_portal is not None: |
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538 self.desktop_portal.end_screenshare() |
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539 self.desktop_sharing = False |
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540 self.desktop_sink_pad = None |
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541 self.bindings = {} |
4139
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542 if self.reset_cb is not None: |
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543 self.reset_cb() |
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544 |
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545 |
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546 async def setup_call( |
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547 self, |
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548 role: str, |
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549 audio_pt: int | None = 96, |
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550 video_pt: int | None = 97, |
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551 ) -> None: |
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552 """Sets up the call. |
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553 |
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554 This method establishes the Gstreamer pipeline for audio and video communication. |
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555 The method also manages STUN and TURN server configurations, signal watchers, and |
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556 various connection handlers for the webrtcbin. |
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557 |
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558 @param role: The role for the call, either 'initiator' or 'responder'. |
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559 @param audio_pt: The payload type for the audio stream. |
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560 @param video_pt: The payload type for the video stream |
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561 |
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562 @raises NotImplementedError: If audio_pt or video_pt is set to None. |
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563 @raises AssertionError: If the role is not 'initiator' or 'responder'. |
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564 """ |
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565 assert role in ("initiator", "responder") |
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566 self.role = role |
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567 if audio_pt is None or video_pt is None: |
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568 raise NotImplementedError("None value is not handled yet") |
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569 |
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570 if self.sources == SOURCES_AUTO: |
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571 video_source_elt = "v4l2src" |
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572 audio_source_elt = "pulsesrc" |
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573 elif self.sources == SOURCES_TEST: |
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574 video_source_elt = "videotestsrc is-live=true pattern=ball" |
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575 audio_source_elt = "audiotestsrc" |
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576 else: |
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577 raise exceptions.InternalError(f'Unknown "sources" value: {self.sources!r}') |
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578 |
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579 extra_elt = "" |
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580 |
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581 if self.sinks == SINKS_APP: |
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582 local_video_sink_elt = ( |
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583 "appsink name=local_video_sink emit-signals=true drop=true max-buffers=1 " |
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584 "sync=True" |
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585 ) |
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586 elif self.sinks == SINKS_AUTO: |
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587 extra_elt = "compositor name=compositor ! autovideosink" |
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588 local_video_sink_elt = """compositor.sink_1""" |
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589 else: |
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590 raise exceptions.InternalError(f"Unknown sinks value {self.sinks!r}") |
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591 |
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592 self.gst_pipe_desc = f""" |
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593 webrtcbin latency=100 name=sendrecv bundle-policy=max-bundle |
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594 |
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595 input-selector name=video_selector |
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596 ! videorate |
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597 ! video/x-raw,framerate=30/1 |
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598 ! tee name=t |
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599 |
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600 {extra_elt} |
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601 |
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602 {video_source_elt} name=video_src ! queue leaky=downstream ! video_selector. |
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603 videotestsrc name=muted_src is-live=true pattern=black ! queue leaky=downstream ! video_selector. |
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604 |
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605 t. |
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606 ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream |
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607 ! videoconvert |
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608 ! vp8enc deadline=1 keyframe-max-dist=60 |
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609 ! rtpvp8pay picture-id-mode=15-bit |
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610 ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt} |
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611 ! sendrecv. |
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612 |
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613 t. |
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614 ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream |
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615 ! videoconvert |
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616 ! {local_video_sink_elt} |
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617 |
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618 {audio_source_elt} name=audio_src |
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619 ! valve |
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620 ! queue max-size-buffers=10 max-size-time=0 max-size-bytes=0 leaky=downstream |
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621 ! audioconvert |
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622 ! audioresample |
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623 ! opusenc audio-type=voice |
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624 ! rtpopuspay |
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625 ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} |
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626 ! sendrecv. |
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627 """ |
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628 |
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629 log.debug(f"Gstreamer pipeline: {self.gst_pipe_desc}") |
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630 |
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631 # Create the pipeline |
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632 self.pipeline = Gst.parse_launch(self.gst_pipe_desc) |
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633 if not self.pipeline: |
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634 raise exceptions.InternalError("Failed to create Gstreamer pipeline.") |
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635 |
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636 self.webrtcbin = self.pipeline.get_by_name("sendrecv") |
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637 self.video_src = self.pipeline.get_by_name("video_src") |
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638 self.muted_src = self.pipeline.get_by_name("muted_src") |
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639 self.video_selector = self.pipeline.get_by_name("video_selector") |
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640 self.audio_valve = self.pipeline.get_by_name("audio_valve") |
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641 |
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642 if self.video_muted: |
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643 self.on_video_mute(True) |
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644 if self.audio_muted: |
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645 self.on_audio_mute(True) |
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646 |
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647 # set STUN and TURN servers |
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648 external_disco = data_format.deserialise( |
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649 await self.bridge.external_disco_get("", self.profile), type_check=list |
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650 ) |
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651 |
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652 for server in external_disco: |
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653 if server["type"] == "stun": |
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654 if server["transport"] == "tcp": |
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655 log.info( |
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656 "ignoring TCP STUN server, GStreamer only support one STUN server" |
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657 ) |
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658 url = f"stun://{server['host']}:{server['port']}" |
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659 log.debug(f"adding stun server: {url}") |
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660 self.webrtcbin.set_property("stun-server", url) |
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661 elif server["type"] == "turn": |
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662 url = "{scheme}://{username}:{password}@{host}:{port}".format( |
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663 scheme="turns" if server["transport"] == "tcp" else "turn", |
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664 username=quote_plus(server["username"]), |
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665 password=quote_plus(server["password"]), |
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666 host=server["host"], |
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667 port=server["port"], |
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668 ) |
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669 log.debug(f"adding turn server: {url}") |
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670 |
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671 if not self.webrtcbin.emit("add-turn-server", url): |
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672 log.warning(f"Erreur while adding TURN server {url}") |
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673 |
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674 # local video feedback |
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675 if self.sinks == SINKS_APP: |
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676 assert self.appsink_data is not None |
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677 local_video_sink = self.pipeline.get_by_name("local_video_sink") |
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678 local_video_sink.set_property("emit-signals", True) |
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679 local_video_sink.connect("new-sample", self.appsink_data.local_video_cb) |
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680 local_video_sink_caps = Gst.Caps.from_string(f"video/x-raw,format=RGB") |
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681 local_video_sink.set_property("caps", local_video_sink_caps) |
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682 |
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683 # Create bus and associate signal watchers |
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684 self.bus = self.pipeline.get_bus() |
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685 if not self.bus: |
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686 log.error("Failed to get bus from pipeline.") |
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687 return |
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688 |
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689 self.bus.add_signal_watch() |
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690 self.webrtcbin.connect("pad-added", self.on_pad_added) |
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691 self.bus.connect("message::error", self.on_bus_error) |
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692 self.bus.connect("message::eos", self.on_bus_eos) |
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693 self.webrtcbin.connect("on-negotiation-needed", self.on_negotiation_needed) |
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694 self.webrtcbin.connect("on-ice-candidate", self.on_ice_candidate) |
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695 self.webrtcbin.connect( |
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696 "notify::ice-gathering-state", self.on_ice_gathering_state_change |
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697 ) |
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698 self.webrtcbin.connect( |
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699 "notify::ice-connection-state", self.on_ice_connection_state |
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700 ) |
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701 |
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702 def start_pipeline(self) -> None: |
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703 """Starts the GStreamer pipeline.""" |
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704 log.debug("starting the pipeline") |
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|
705 self.pipeline.set_state(Gst.State.PLAYING) |
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|
706 |
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|
707 def on_negotiation_needed(self, webrtc): |
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|
708 """Initiate SDP offer when negotiation is needed.""" |
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|
709 log.debug("Negotiation needed.") |
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710 if self.role == "initiator": |
6745c6bd4c7a
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711 log.debug("Creating offer…") |
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|
712 promise = Gst.Promise.new_with_change_func(self.on_offer_created) |
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|
713 self.webrtcbin.emit("create-offer", None, promise) |
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|
714 |
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|
715 def on_offer_created(self, promise): |
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716 """Callback for when SDP offer is created.""" |
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|
717 log.info("on_offer_created called") |
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|
718 assert promise.wait() == Gst.PromiseResult.REPLIED |
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719 reply = promise.get_reply() |
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|
720 if reply is None: |
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|
721 log.error("Promise reply is None. Offer creation might have failed.") |
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|
722 return |
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|
723 offer = reply["offer"] |
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|
724 self.offer = offer.sdp.as_text() |
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|
725 log.info(f"SDP offer created: \n{self.offer}") |
6745c6bd4c7a
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|
726 self._set_media_types(self.offer) |
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727 promise = Gst.Promise.new() |
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|
728 self.webrtcbin.emit("set-local-description", offer, promise) |
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|
729 promise.interrupt() |
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|
730 self._a_call(self._start_call) |
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|
731 |
6745c6bd4c7a
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|
732 def on_answer_set(self, promise): |
6745c6bd4c7a
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|
733 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
734 |
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|
735 def on_answer_created(self, promise, _, __): |
6745c6bd4c7a
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|
736 """Callback for when SDP answer is created.""" |
6745c6bd4c7a
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changeset
|
737 assert promise.wait() == Gst.PromiseResult.REPLIED |
6745c6bd4c7a
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|
738 reply = promise.get_reply() |
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|
739 answer = reply["answer"] |
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|
740 promise = Gst.Promise.new() |
6745c6bd4c7a
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|
741 self.webrtcbin.emit("set-local-description", answer, promise) |
6745c6bd4c7a
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parents:
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changeset
|
742 promise.interrupt() |
6745c6bd4c7a
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|
743 answer_sdp = answer.sdp.as_text() |
6745c6bd4c7a
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parents:
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|
744 log.info(f"SDP answer set: \n{answer_sdp}") |
6745c6bd4c7a
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|
745 self.sdp_set = True |
6745c6bd4c7a
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changeset
|
746 self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile) |
6745c6bd4c7a
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parents:
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|
747 |
6745c6bd4c7a
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|
748 def on_offer_set(self, promise): |
6745c6bd4c7a
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|
749 assert promise.wait() == Gst.PromiseResult.REPLIED |
6745c6bd4c7a
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parents:
diff
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|
750 promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None) |
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|
751 self.webrtcbin.emit("create-answer", None, promise) |
6745c6bd4c7a
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|
752 |
6745c6bd4c7a
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|
753 def link_element_or_pad( |
6745c6bd4c7a
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754 self, source: Gst.Element, dest: Gst.Element | Gst.Pad |
6745c6bd4c7a
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|
755 ) -> bool: |
6745c6bd4c7a
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|
756 """Check if dest is a pad or an element, and link appropriately""" |
6745c6bd4c7a
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|
757 src_pad = source.get_static_pad("src") |
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parents:
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|
758 |
6745c6bd4c7a
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|
759 if isinstance(dest, Gst.Pad): |
6745c6bd4c7a
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parents:
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changeset
|
760 # If the dest is a pad, link directly |
6745c6bd4c7a
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parents:
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|
761 if not src_pad.link(dest) == Gst.PadLinkReturn.OK: |
6745c6bd4c7a
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|
762 log.error( |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
763 "Failed to link 'conv' to the compositor's newly requested pad!" |
6745c6bd4c7a
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parents:
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|
764 ) |
6745c6bd4c7a
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|
765 return False |
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|
766 elif isinstance(dest, Gst.Element): |
6745c6bd4c7a
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parents:
diff
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|
767 return source.link(dest) |
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|
768 else: |
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|
769 log.error(f"Unexpected type for dest: {type(sink)}") |
6745c6bd4c7a
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|
770 return False |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
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|
771 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
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|
772 return True |
6745c6bd4c7a
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|
773 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
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|
774 def scaled_dimensions( |
6745c6bd4c7a
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|
775 self, original_width: int, original_height: int, max_width: int, max_height: int |
6745c6bd4c7a
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|
776 ) -> tuple[int, int]: |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
777 """Calculates the scaled dimensions preserving aspect ratio. |
6745c6bd4c7a
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|
778 |
6745c6bd4c7a
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parents:
diff
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|
779 @param original_width: Original width of the video stream. |
6745c6bd4c7a
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parents:
diff
changeset
|
780 @param original_height: Original height of the video stream. |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
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|
781 @param max_width: Maximum desired width for the scaled video. |
6745c6bd4c7a
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changeset
|
782 @param max_height: Maximum desired height for the scaled video. |
6745c6bd4c7a
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|
783 @return: The width and height of the scaled video. |
6745c6bd4c7a
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parents:
diff
changeset
|
784 """ |
6745c6bd4c7a
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diff
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|
785 aspect_ratio = original_width / original_height |
6745c6bd4c7a
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|
786 new_width = int(max_height * aspect_ratio) |
6745c6bd4c7a
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|
787 |
6745c6bd4c7a
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|
788 if new_width <= max_width: |
6745c6bd4c7a
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|
789 return new_width, max_height |
6745c6bd4c7a
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|
790 |
6745c6bd4c7a
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|
791 new_height = int(max_width / aspect_ratio) |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
792 return max_width, new_height |
6745c6bd4c7a
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|
793 |
6745c6bd4c7a
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|
794 def on_remote_decodebin_stream(self, _, pad: Gst.Pad) -> None: |
6745c6bd4c7a
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|
795 """Handle the stream from the remote decodebin. |
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|
796 |
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diff
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|
797 This method processes the incoming stream from the remote decodebin, determining |
6745c6bd4c7a
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|
798 whether it's video or audio. It then sets up the appropriate GStreamer elements |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
799 for video/audio processing and adds them to the pipeline. |
6745c6bd4c7a
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|
800 |
6745c6bd4c7a
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|
801 @param pad: The Gst.Pad from the remote decodebin producing the stream. |
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|
802 """ |
6745c6bd4c7a
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|
803 assert self.pipeline is not None |
6745c6bd4c7a
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diff
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|
804 if not pad.has_current_caps(): |
6745c6bd4c7a
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parents:
diff
changeset
|
805 log.error(f"{pad} has no caps, ignoring") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
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diff
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|
806 return |
6745c6bd4c7a
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|
807 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
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diff
changeset
|
808 caps = pad.get_current_caps() |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
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|
809 assert len(caps) |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
810 s = caps[0] |
6745c6bd4c7a
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diff
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|
811 name = s.get_name() |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
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changeset
|
812 log.debug(f"====> NAME START: {name}") |
6745c6bd4c7a
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parents:
diff
changeset
|
813 |
6745c6bd4c7a
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parents:
diff
changeset
|
814 q = Gst.ElementFactory.make("queue") |
6745c6bd4c7a
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parents:
diff
changeset
|
815 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
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|
816 if name.startswith("video"): |
6745c6bd4c7a
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parents:
diff
changeset
|
817 log.debug("===> VIDEO OK") |
6745c6bd4c7a
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parents:
diff
changeset
|
818 |
6745c6bd4c7a
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diff
changeset
|
819 self._remote_video_pad = pad |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
820 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
821 # Check and log the original size of the video |
6745c6bd4c7a
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parents:
diff
changeset
|
822 width = s.get_int("width").value |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
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changeset
|
823 height = s.get_int("height").value |
6745c6bd4c7a
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diff
changeset
|
824 log.info(f"Original video size: {width}x{height}") |
6745c6bd4c7a
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parents:
diff
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|
825 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
826 # This is a fix for an issue found with Movim on desktop: a non standard |
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827 # resolution is used (990x557) resulting in bad alignement and no color in |
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828 # rendered image |
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829 adjust_resolution = width % 4 != 0 or height % 4 != 0 |
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830 if adjust_resolution: |
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831 log.info("non standard resolution, we need to adjust size") |
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832 width = (width + 3) // 4 * 4 |
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833 height = (height + 3) // 4 * 4 |
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834 log.info(f"Adjusted video size: {width}x{height}") |
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835 |
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836 conv = Gst.ElementFactory.make("videoconvert") |
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837 if self.sinks == SINKS_APP: |
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838 assert self.appsink_data is not None |
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839 remote_video_sink = Gst.ElementFactory.make("appsink") |
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840 |
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841 appsink_caps = Gst.Caps.from_string("video/x-raw,format=RGB") |
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842 remote_video_sink.set_property("caps", appsink_caps) |
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843 |
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844 remote_video_sink.set_property("emit-signals", True) |
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845 remote_video_sink.set_property("drop", True) |
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846 remote_video_sink.set_property("max-buffers", 1) |
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847 remote_video_sink.set_property("sync", True) |
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848 remote_video_sink.connect("new-sample", self.appsink_data.remote_video_cb) |
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849 self.pipeline.add(remote_video_sink) |
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850 elif self.sinks == SINKS_AUTO: |
4139
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851 compositor = self.pipeline.get_by_name("compositor") |
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852 |
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853 sink1_pad = compositor.get_static_pad("sink_1") |
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854 |
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855 local_width, local_height = self.scaled_dimensions( |
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856 sink1_pad.get_property("width"), |
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857 sink1_pad.get_property("height"), |
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858 width // 3, |
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|
859 height // 3, |
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|
860 ) |
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861 |
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|
862 sink1_pad.set_property("xpos", width - local_width) |
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863 sink1_pad.set_property("ypos", height - local_height) |
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864 sink1_pad.set_property("width", local_width) |
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865 sink1_pad.set_property("height", local_height) |
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866 sink1_pad.set_property("zorder", 1) |
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867 |
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868 # Request a new pad for the remote stream |
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869 sink_pad_template = compositor.get_pad_template("sink_%u") |
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870 remote_video_sink = compositor.request_pad(sink_pad_template, None, None) |
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|
871 remote_video_sink.set_property("zorder", 0) |
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872 |
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|
873 else: |
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874 raise exceptions.InternalError(f'Unhandled "sinks" value: {self.sinks!r}') |
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875 |
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|
876 if adjust_resolution: |
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|
877 videoscale = Gst.ElementFactory.make("videoscale") |
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878 adjusted_caps = Gst.Caps.from_string( |
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879 f"video/x-raw,width={width},height={height}" |
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880 ) |
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881 capsfilter = Gst.ElementFactory.make("capsfilter") |
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882 capsfilter.set_property("caps", adjusted_caps) |
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883 |
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884 self.pipeline.add(q, conv, videoscale, capsfilter) |
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885 |
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886 |
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887 self.pipeline.sync_children_states() |
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888 ret = pad.link(q.get_static_pad("sink")) |
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889 if ret != Gst.PadLinkReturn.OK: |
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|
890 log.error(f"Error linking pad: {ret}") |
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891 q.link(conv) |
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|
892 conv.link(videoscale) |
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893 videoscale.link(capsfilter) |
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894 self.link_element_or_pad(capsfilter.link, remote_video_sink) |
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|
895 |
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|
896 else: |
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|
897 self.pipeline.add(q, conv) |
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898 |
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899 self.pipeline.sync_children_states() |
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|
900 ret = pad.link(q.get_static_pad("sink")) |
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|
901 if ret != Gst.PadLinkReturn.OK: |
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|
902 log.error(f"Error linking pad: {ret}") |
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|
903 q.link(conv) |
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|
904 self.link_element_or_pad(conv, remote_video_sink) |
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|
905 |
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|
906 elif name.startswith("audio"): |
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|
907 log.debug("===> Audio OK") |
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|
908 conv = Gst.ElementFactory.make("audioconvert") |
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|
909 resample = Gst.ElementFactory.make("audioresample") |
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910 remote_audio_sink = Gst.ElementFactory.make("autoaudiosink") |
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|
911 self.pipeline.add(q, conv, resample, remote_audio_sink) |
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|
912 self.pipeline.sync_children_states() |
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|
913 ret = pad.link(q.get_static_pad("sink")) |
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914 if ret != Gst.PadLinkReturn.OK: |
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915 log.error(f"Error linking pad: {ret}") |
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|
916 q.link(conv) |
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|
917 conv.link(resample) |
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|
918 resample.link(remote_audio_sink) |
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|
919 |
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|
920 else: |
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|
921 log.warning(f"unmanaged name: {name!r}") |
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|
922 |
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|
923 def on_pad_added(self, __, pad: Gst.Pad) -> None: |
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924 """Handle the addition of a new pad to the element. |
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|
925 |
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|
926 When a new source pad is added to the element, this method creates a decodebin, |
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|
927 connects it to handle the stream, and links the pad to the decodebin. |
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|
928 |
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diff
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|
929 @param __: Placeholder for the signal source. Not used in this method. |
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930 @param pad: The newly added pad. |
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|
931 """ |
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|
932 log.debug("on_pad_added") |
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|
933 if pad.direction != Gst.PadDirection.SRC: |
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|
934 return |
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|
935 |
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|
936 decodebin = Gst.ElementFactory.make("decodebin") |
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|
937 decodebin.connect("pad-added", self.on_remote_decodebin_stream) |
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938 self.pipeline.add(decodebin) |
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939 decodebin.sync_state_with_parent() |
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|
940 pad.link(decodebin.get_static_pad("sink")) |
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941 |
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|
942 async def _start_call(self) -> None: |
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943 """Initiate the call. |
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944 |
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945 Initiates a call with the callee using the stored offer. If there are any buffered |
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946 local ICE candidates, they are sent as part of the initiation. |
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947 """ |
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948 assert self.callee |
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949 self.sid = await self.bridge.call_start( |
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950 str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile |
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951 ) |
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952 if self.local_candidates_buffer: |
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953 log.debug( |
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954 f"sending buffered local ICE candidates: {self.local_candidates_buffer}" |
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955 ) |
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956 if self.pwd is None: |
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957 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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958 self.extract_ufrag_pwd(sdp) |
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959 ice_data = {} |
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960 for media_type, candidates in self.local_candidates_buffer.items(): |
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961 ice_data[media_type] = { |
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962 "ufrag": self.ufrag, |
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963 "pwd": self.pwd, |
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964 "candidates": candidates, |
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965 } |
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966 await self.bridge.ice_candidates_add( |
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967 self.sid, data_format.serialise(ice_data), self.profile |
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968 ) |
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969 self.local_candidates_buffer.clear() |
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970 |
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971 def _remote_sdp_set(self, promise) -> None: |
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972 assert promise.wait() == Gst.PromiseResult.REPLIED |
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973 self.sdp_set = True |
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974 |
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975 def on_accepted_call(self, sdp: str, profile: str) -> None: |
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976 """Outgoing call has been accepted. |
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977 |
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978 @param sdp: The SDP answer string received from the other party. |
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979 @param profile: Profile used for the call. |
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980 """ |
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981 log.debug(f"SDP answer received: \n{sdp}") |
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982 |
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983 __, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp) |
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984 answer = GstWebRTC.WebRTCSessionDescription.new( |
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985 GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg |
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986 ) |
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987 promise = Gst.Promise.new_with_change_func(self._remote_sdp_set) |
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988 self.webrtcbin.emit("set-remote-description", answer, promise) |
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989 |
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990 async def answer_call(self, sdp: str, profile: str) -> None: |
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991 """Answer an incoming call |
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992 |
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993 @param sdp: The SDP offer string received from the initiator. |
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994 @param profile: Profile used for the call. |
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995 |
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996 @raise AssertionError: Raised when either "VP8" or "OPUS" is not present in |
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997 payload types. |
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998 """ |
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999 log.debug(f"SDP offer received: \n{sdp}") |
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1000 self._set_media_types(sdp) |
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1001 __, offer_sdp_msg = GstSdp.SDPMessage.new_from_text(sdp) |
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1002 payload_types = self.get_payload_types( |
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1003 offer_sdp_msg, video_encoding="VP8", audio_encoding="OPUS" |
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1004 ) |
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1005 assert "VP8" in payload_types |
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1006 assert "OPUS" in payload_types |
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1007 await self.setup_call( |
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1008 "responder", audio_pt=payload_types["OPUS"], video_pt=payload_types["VP8"] |
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1009 ) |
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1010 self.start_pipeline() |
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1011 offer = GstWebRTC.WebRTCSessionDescription.new( |
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1012 GstWebRTC.WebRTCSDPType.OFFER, offer_sdp_msg |
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1013 ) |
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1014 promise = Gst.Promise.new_with_change_func(self.on_offer_set) |
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1015 self.webrtcbin.emit("set-remote-description", offer, promise) |
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1016 |
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1017 def on_ice_candidate(self, webrtc, mline_index, candidate_sdp): |
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1018 """Handles the on-ice-candidate signal of webrtcbin. |
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1019 |
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1020 @param webrtc: The webrtcbin element. |
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1021 @param mlineindex: The mline index. |
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1022 @param candidate: The ICE candidate. |
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1023 """ |
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1024 log.debug( |
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1025 f"Local ICE candidate. MLine Index: {mline_index}, Candidate: {candidate_sdp}" |
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1026 ) |
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1027 parsed_candidate = self.parse_ice_candidate(candidate_sdp) |
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1028 try: |
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1029 media_type = self.media_types[mline_index] |
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1030 except KeyError: |
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1031 raise exceptions.InternalError("can't find media type") |
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1032 |
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1033 if self.sid is None: |
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1034 log.debug("buffering local ICE candidate") |
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1035 self.local_candidates_buffer.setdefault(media_type, []).append( |
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1036 parsed_candidate |
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1037 ) |
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1038 else: |
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1039 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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1040 assert sdp is not None |
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1041 ufrag, pwd = self.extract_ufrag_pwd(sdp) |
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1042 ice_data = {"ufrag": ufrag, "pwd": pwd, "candidates": [parsed_candidate]} |
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1043 self._a_call( |
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1044 self.bridge.ice_candidates_add, |
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1045 self.sid, |
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1046 data_format.serialise({media_type: ice_data}), |
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1047 self.profile, |
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1048 ) |
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1049 |
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1050 def on_ice_candidates_new(self, candidates: dict) -> None: |
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1051 """Handle new ICE candidates. |
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1052 |
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1053 @param candidates: A dictionary containing media types ("audio" or "video") as |
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1054 keys and corresponding ICE data as values. |
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1055 |
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1056 @raise exceptions.InternalError: Raised when sdp mline index is not found. |
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1057 """ |
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1058 if not self.sdp_set: |
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1059 log.debug("buffering remote ICE candidate") |
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1060 for media_type in ("audio", "video"): |
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1061 media_candidates = candidates.get(media_type) |
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1062 if media_candidates: |
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1063 buffer = self.remote_candidates_buffer[media_type] |
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1064 buffer["candidates"].extend(media_candidates["candidates"]) |
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1065 return |
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1066 for media_type, ice_data in candidates.items(): |
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1067 for candidate in ice_data["candidates"]: |
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1068 candidate_sdp = self.build_ice_candidate(candidate) |
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1069 try: |
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1070 mline_index = self.get_sdp_mline_index(media_type) |
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1071 except Exception as e: |
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1072 raise exceptions.InternalError(f"Can't find sdp mline index: {e}") |
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1073 self.webrtcbin.emit("add-ice-candidate", mline_index, candidate_sdp) |
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1074 log.debug( |
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1075 f"Remote ICE candidate added. MLine Index: {mline_index}, " |
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1076 f"{candidate_sdp}" |
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1077 ) |
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1078 |
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1079 def on_ice_gathering_state_change(self, pspec, __): |
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1080 state = self.webrtcbin.get_property("ice-gathering-state") |
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1081 log.debug(f"ICE gathering state changed to {state}") |
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1082 |
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1083 def on_ice_connection_state(self, pspec, __): |
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1084 state = self.webrtcbin.props.ice_connection_state |
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1085 if state == GstWebRTC.WebRTCICEConnectionState.FAILED: |
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1086 log.error("ICE connection failed") |
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1087 log.info(f"ICE connection state changed to {state}") |
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1088 |
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1089 def on_bus_error(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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1090 """Handles the GStreamer bus error messages. |
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1091 |
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1092 @param bus: The GStreamer bus. |
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1093 @param message: The error message. |
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1094 """ |
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1095 err, debug = message.parse_error() |
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1096 log.error(f"Error from {message.src.get_name()}: {err.message}") |
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1097 log.error(f"Debugging info: {debug}") |
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1098 |
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1099 def on_bus_eos(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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1100 """Handles the GStreamer bus eos messages. |
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1101 |
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1102 @param bus: The GStreamer bus. |
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1103 @param message: The eos message. |
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1104 """ |
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1105 log.info("End of stream") |
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1106 |
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1107 def on_audio_mute(self, muted: bool) -> None: |
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1108 """Handles (un)muting of audio. |
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1109 |
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1110 @param muted: True if audio is muted. |
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1111 """ |
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1112 if self.audio_valve is not None: |
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1113 self.audio_valve.set_property("drop", muted) |
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1114 state = "muted" if muted else "unmuted" |
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1115 log.info(f"audio is now {state}") |
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1116 |
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1117 def on_video_mute(self, muted: bool) -> None: |
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1118 """Handles (un)muting of video. |
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1119 |
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1120 @param muted: True if video is muted. |
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1121 """ |
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1122 if self.video_selector is not None: |
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1123 current_source = None if muted else "desktop" if self.desktop_sharing else "video" |
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1124 self.switch_video_source(current_source) |
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1125 state = "muted" if muted else "unmuted" |
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1126 log.info(f"Video is now {state}") |
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1127 |
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1128 def on_desktop_switch(self, desktop_active: bool) -> None: |
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1129 """Switches the video source between desktop and video. |
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1130 |
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1131 @param desktop_active: True if desktop must be active. False for video. |
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1132 """ |
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1133 if desktop_active and self.desktop_portal is not None: |
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1134 aio.run_async(self.on_desktop_switch_portal(desktop_active)) |
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1135 else: |
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1136 self.do_desktop_switch(desktop_active) |
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1137 |
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1138 async def on_desktop_switch_portal(self, desktop_active: bool) -> None: |
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1139 """Call freedesktop screenshare portal and the activate the shared stream""" |
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1140 assert self.desktop_portal is not None |
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1141 try: |
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1142 screenshare_data = await self.desktop_portal.request_screenshare() |
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1143 except exceptions.CancelError: |
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1144 self.desktop_sharing = False |
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1145 return |
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1146 self.desktop_sharing_data = { |
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1147 "path": str(screenshare_data["node_id"]) |
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1148 } |
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1149 self.do_desktop_switch(desktop_active) |
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1150 |
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1151 def do_desktop_switch(self, desktop_active: bool) -> None: |
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1152 if self.video_muted: |
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1153 # Update the active source state but do not switch |
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1154 self.desktop_sharing = desktop_active |
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1155 return |
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1156 |
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1157 source = "desktop" if desktop_active else "video" |
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1158 self.switch_video_source(source) |
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1159 self.desktop_sharing = desktop_active |
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1160 |
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1161 def switch_video_source(self, source: str|None) -> None: |
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1162 """Activates the specified source while deactivating the others. |
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1163 |
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1164 @param source: 'desktop', 'video', 'muted' or None for muted source. |
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1165 """ |
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1166 if source is None: |
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1167 source = "muted" |
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1168 if source not in ["video", "muted", "desktop"]: |
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1169 raise ValueError( |
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1170 f"Invalid source: {source!r}, use one of {'video', 'muted', 'desktop'}" |
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1171 ) |
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1172 |
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1173 self.pipeline.set_state(Gst.State.PAUSED) |
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1174 |
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1175 # Create a new desktop source if necessary |
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1176 if source == "desktop": |
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1177 self._setup_desktop_source(self.desktop_sharing_data) |
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1178 |
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1179 # Activate the chosen source and deactivate the others |
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1180 for src_name in ["video", "muted", "desktop"]: |
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1181 src_element = self.pipeline.get_by_name(f"{src_name}_src") |
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1182 if src_name == source: |
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1183 if src_element: |
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1184 src_element.set_state(Gst.State.PLAYING) |
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1185 else: |
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1186 if src_element: |
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1187 if src_name == "desktop": |
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1188 self._remove_desktop_source(src_element) |
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1189 else: |
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1190 src_element.set_state(Gst.State.NULL) |
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1191 |
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1192 # Set the video_selector active pad |
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1193 if source == "desktop": |
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1194 if self.desktop_sink_pad: |
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1195 pad = self.desktop_sink_pad |
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1196 else: |
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1197 log.error(f"No desktop pad available") |
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1198 pad = None |
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1199 else: |
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1200 pad_name = f"sink_{['video', 'muted'].index(source)}" |
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1201 pad = self.video_selector.get_static_pad(pad_name) |
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1202 |
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1203 if pad is not None: |
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1204 self.video_selector.props.active_pad = pad |
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1205 |
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1206 self.pipeline.set_state(Gst.State.PLAYING) |
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1207 |
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1208 def _setup_desktop_source(self, properties: dict[str, object]|None) -> None: |
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1209 """Set up a new desktop source. |
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1210 |
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1211 @param properties: The properties to set on the desktop source. |
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1212 """ |
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1213 source_elt = "ximagesrc" if self.desktop_portal is None else "pipewiresrc" |
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1214 desktop_src = Gst.ElementFactory.make(source_elt, "desktop_src") |
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1215 if properties is None: |
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1216 properties = {} |
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1217 for key, value in properties.items(): |
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1218 log.debug(f"setting {source_elt} property: {key!r}={value!r}") |
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1219 desktop_src.set_property(key, value) |
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1220 video_convert = Gst.ElementFactory.make("videoconvert", "desktop_videoconvert") |
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1221 queue = Gst.ElementFactory.make("queue", "desktop_queue") |
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1222 queue.set_property("leaky", "downstream") |
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1223 |
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1224 self.pipeline.add(desktop_src) |
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1225 self.pipeline.add(video_convert) |
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1226 self.pipeline.add(queue) |
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1227 |
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1228 desktop_src.link(video_convert) |
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1229 video_convert.link(queue) |
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1230 |
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1231 sink_pad_template = self.video_selector.get_pad_template("sink_%u") |
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1232 self.desktop_sink_pad = self.video_selector.request_pad(sink_pad_template, None, None) |
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1233 queue_src_pad = queue.get_static_pad("src") |
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1234 queue_src_pad.link(self.desktop_sink_pad) |
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1235 |
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1236 desktop_src.sync_state_with_parent() |
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1237 video_convert.sync_state_with_parent() |
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1238 queue.sync_state_with_parent() |
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1239 |
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1240 def _remove_desktop_source(self, desktop_src: Gst.Element) -> None: |
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1241 """Remove the desktop source from the pipeline. |
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1242 |
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1243 @param desktop_src: The desktop source to remove. |
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1244 """ |
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1245 # Remove elements for the desktop source |
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1246 video_convert = self.pipeline.get_by_name("desktop_videoconvert") |
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1247 queue = self.pipeline.get_by_name("desktop_queue") |
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1248 |
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1249 if video_convert: |
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1250 video_convert.set_state(Gst.State.NULL) |
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1251 desktop_src.unlink(video_convert) |
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1252 self.pipeline.remove(video_convert) |
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1253 |
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1254 if queue: |
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1255 queue.set_state(Gst.State.NULL) |
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1256 self.pipeline.remove(queue) |
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1257 |
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1258 desktop_src.set_state(Gst.State.NULL) |
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1259 self.pipeline.remove(desktop_src) |
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1260 |
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1261 # Release the pad associated with the desktop source |
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1262 if self.desktop_sink_pad: |
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1263 self.video_selector.release_request_pad(self.desktop_sink_pad) |
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1264 self.desktop_sink_pad = None |
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1265 |
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1266 if self.desktop_portal is not None: |
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1267 self.desktop_portal.end_screenshare() |
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1268 |
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1269 async def end_call(self) -> None: |
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1270 """Stop streaming and clean instance""" |
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1271 self.reset_instance() |