Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc.py @ 4294:a0ed5c976bf8 default tip @
component conferences, plugin XEP-0167, XEP-0298: add stream user metadata:
A/V conference now adds user metadata about the stream it is forwarding through XEP-0298.
This is parsed and added to metadata during confirmation on client side.
rel 448
author | Goffi <goffi@goffi.org> |
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date | Tue, 06 Aug 2024 23:43:11 +0200 |
parents | 2992f9d1e039 |
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1 #!/usr/bin/env python3 |
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2 |
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3 # Libervia WebRTC implementation |
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4 # Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org) |
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5 |
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6 # This program is free software: you can redistribute it and/or modify |
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7 # it under the terms of the GNU Affero General Public License as published by |
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8 # the Free Software Foundation, either version 3 of the License, or |
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9 # (at your option) any later version. |
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10 |
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11 # This program is distributed in the hope that it will be useful, |
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12 # but WITHOUT ANY WARRANTY; without even the implied warranty of |
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13 # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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14 # GNU Affero General Public License for more details. |
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15 |
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16 # You should have received a copy of the GNU Affero General Public License |
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17 # along with this program. If not, see <http://www.gnu.org/licenses/>. |
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18 |
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19 from collections.abc import Awaitable |
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20 import gi |
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21 |
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22 gi.require_versions({"Gst": "1.0", "GstWebRTC": "1.0"}) |
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23 from gi.repository import Gst, GstWebRTC, GstSdp |
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24 |
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25 from libervia.backend.core import exceptions |
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26 |
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27 try: |
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28 from gi.overrides import Gst as _ |
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29 except ImportError: |
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30 raise exceptions.MissingModule( |
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31 "No GStreamer Python overrides available. Please install relevant packages on " |
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32 "your system (e.g., `python3-gst-1.0` on Debian and derivatives)." |
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33 ) |
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34 import asyncio |
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35 from datetime import datetime |
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36 import logging |
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37 import re |
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38 from typing import Callable, Final |
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39 from urllib.parse import quote_plus |
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40 |
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41 from libervia.backend.tools.common import data_format |
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42 from libervia.frontends.tools import aio, display_servers, jid |
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43 from .webrtc_models import ( |
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44 CallData, |
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45 SinksApp, |
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46 SinksAuto, |
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47 SinksData, |
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48 SinksDataChannel, |
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49 SinksNone, |
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50 SourcesAuto, |
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51 SourcesData, |
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52 SourcesDataChannel, |
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53 SourcesNone, |
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54 SourcesPipeline, |
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55 SourcesTest, |
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56 ) |
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57 |
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58 current_server = display_servers.detect() |
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59 if current_server == display_servers.X11: |
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60 # GSTreamer's ximagesrc documentation asks to run this function |
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61 import ctypes |
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62 |
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63 ctypes.CDLL("libX11.so.6").XInitThreads() |
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64 |
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65 |
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66 log = logging.getLogger(__name__) |
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67 VIDEO_SOURCE_AUTO: Final = "v4l2src" |
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68 AUDIO_SOURCE_AUTO: Final = "pulsesrc" |
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69 NONE_NOT_IMPLEMENTED_MSG: Final = "None value is not handled yet." |
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70 |
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71 Gst.init(None) |
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72 |
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73 |
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74 class WebRTC: |
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75 """GSTreamer based WebRTC implementation for audio and video communication. |
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76 |
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77 This class encapsulates the WebRTC functionalities required for initiating and |
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78 handling audio and video calls, and data channels. |
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79 """ |
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80 |
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81 def __init__( |
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82 self, |
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83 bridge, |
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84 profile: str, |
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85 sources_data: SourcesData | None = None, |
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86 sinks_data: SinksData | None = None, |
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87 reset_cb: Callable | None = None, |
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88 merge_pip: bool | None = None, |
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89 target_size: tuple[int, int] | None = None, |
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90 call_start_cb: Callable[[str, dict, str], Awaitable[str]] | None = None, |
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91 dc_data_list: list[SourcesDataChannel | SinksDataChannel] | None = None, |
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92 ) -> None: |
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93 """Initializes a new WebRTC instance. |
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94 |
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95 @param bridge: An instance of backend bridge. |
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96 @param profile: Libervia profile. |
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97 @param sources_data: Data of the sources. |
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98 The model used will determine which sources to use. |
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99 SourcesDataChannel can be used here as a convenience. It will then be moved |
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100 to ``data_channels`` and ``SourcesNone`` will be used instead for |
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101 ``sources_data``. |
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102 If None, SourcesAuto will be used. |
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103 @param sinks_data: Data of the sinks. |
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104 The model used will determine which sinks to use. |
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105 SinksDataChannel can be used here as a convenience. It will then be moved |
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106 to ``data_channels`` and ``SinksNone`` will be used instead for |
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107 ``sinks_data``. |
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108 If None, SinksAuto will be used. |
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109 @param reset_cb: An optional Callable that is triggered on reset events. Can be |
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110 used to reset UI data on new calls. |
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111 @param merge_pip: A boolean flag indicating whether Picture-in-Picture mode is |
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112 enabled. When PiP is used, local feedback is merged to remote video stream. |
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113 Only one video stream is then produced (the local one). |
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114 If None, PiP mode is selected automatically according to selected sink (it's |
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115 used for SinksAuto only for now). |
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116 @param target_size: Expected size of the final sink stream. Mainly use by composer |
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117 when ``merge_pip`` is set. |
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118 None to autodetect (no real autodetection implemeted yet, default to |
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119 (1280,720)). |
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120 @param call_start_cb: Called when call is started. |
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121 @param dc_data_list: Data Channels to create. |
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122 If a SourcesDataChannel is used as ``sources_data``, or a SinksDataChannel is |
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123 used as ``sinks_data``, they will be automatically added to this list. |
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124 """ |
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125 self.main_loop = asyncio.get_event_loop() |
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126 self.bridge = bridge |
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127 self.profile = profile |
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128 self.pipeline = None |
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129 self._audio_muted = False |
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130 self._video_muted = False |
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131 self._desktop_sharing = False |
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132 self.desktop_sharing_data = None |
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133 if dc_data_list is None: |
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134 dc_data_list = [] |
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135 self.dc_data_list = dc_data_list |
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136 if sources_data is None: |
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137 sources_data = SourcesAuto() |
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138 elif isinstance(sources_data, SourcesDataChannel): |
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139 dc_data_list.append(sources_data) |
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140 sources_data = SourcesNone() |
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141 self.sources_data = sources_data |
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142 if sinks_data is None: |
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143 sinks_data = SinksAuto() |
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144 elif isinstance(sinks_data, SinksDataChannel): |
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145 dc_data_list.append(sinks_data) |
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146 sinks_data = SinksNone() |
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147 self.sinks_data = sinks_data |
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148 if target_size is None: |
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149 target_size = (1280, 720) |
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150 self.target_width, self.target_height = target_size |
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151 if merge_pip is None: |
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152 merge_pip = isinstance(sinks_data, SinksAuto) |
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153 self.merge_pip = merge_pip |
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154 if call_start_cb is None: |
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155 call_start_cb = self._call_start |
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156 self.call_start_cb = call_start_cb |
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157 if isinstance(sinks_data, SinksApp): |
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158 if merge_pip and sinks_data.remote_video_cb is not None: |
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159 raise ValueError("Remote_video_cb can't be used when merge_pip is used!") |
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160 self.reset_cb = reset_cb |
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161 if current_server == display_servers.WAYLAND: |
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162 from .portal_desktop import DesktopPortal |
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163 |
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164 self.desktop_portal = DesktopPortal( |
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165 on_session_closed_cb=self.on_portal_session_closed |
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166 ) |
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167 else: |
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168 self.desktop_portal = None |
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169 self.reset_instance() |
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170 |
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171 @property |
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172 def audio_muted(self): |
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173 return self._audio_muted |
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174 |
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175 @audio_muted.setter |
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176 def audio_muted(self, muted: bool) -> None: |
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177 if muted != self._audio_muted: |
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178 self._audio_muted = muted |
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179 self.on_audio_mute(muted) |
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180 |
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181 @property |
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182 def video_muted(self): |
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183 return self._video_muted |
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184 |
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185 @video_muted.setter |
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186 def video_muted(self, muted: bool) -> None: |
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187 if muted != self._video_muted: |
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188 self._video_muted = muted |
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189 self.on_video_mute(muted) |
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190 |
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191 @property |
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192 def desktop_sharing(self): |
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193 return self._desktop_sharing |
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194 |
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195 @desktop_sharing.setter |
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196 def desktop_sharing(self, active: bool) -> None: |
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197 if active != self._desktop_sharing: |
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198 self._desktop_sharing = active |
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199 self.on_desktop_switch(active) |
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200 try: |
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201 cb = self.bindings["desktop_sharing"] |
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202 except KeyError: |
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203 pass |
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204 else: |
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205 cb(active) |
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206 |
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207 @property |
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208 def sdp_set(self): |
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209 return self._sdp_set |
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210 |
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211 @sdp_set.setter |
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212 def sdp_set(self, is_set: bool): |
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213 self._sdp_set = is_set |
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214 if is_set: |
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215 self.on_ice_candidates_new(self.remote_candidates_buffer) |
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216 for data in self.remote_candidates_buffer.values(): |
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217 data["candidates"].clear() |
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218 |
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219 @property |
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220 def media_types(self): |
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221 if self._media_types is None: |
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222 raise Exception("self._media_types should not be None!") |
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223 return self._media_types |
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224 |
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225 @media_types.setter |
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226 def media_types(self, new_media_types: dict) -> None: |
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227 self._media_types = new_media_types |
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228 self._media_types_inv = {v: k for k, v in new_media_types.items()} |
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229 |
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230 @property |
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231 def media_types_inv(self) -> dict: |
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232 if self._media_types_inv is None: |
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233 raise Exception("self._media_types_inv should not be None!") |
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234 return self._media_types_inv |
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235 |
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236 def bind(self, **kwargs: Callable) -> None: |
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237 self.bindings.clear() |
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238 for key, cb in kwargs.items(): |
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239 if key not in ("desktop_sharing",): |
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240 raise ValueError( |
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241 'Only "desktop_sharing" is currently allowed for binding' |
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242 ) |
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243 self.bindings[key] = cb |
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244 |
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245 def generate_dot_file( |
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246 self, |
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247 filename: str = "pipeline", |
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248 details: Gst.DebugGraphDetails = Gst.DebugGraphDetails.ALL, |
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249 with_timestamp: bool = True, |
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250 bin_: Gst.Bin | None = None, |
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251 ) -> None: |
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252 """Generate Dot File for debugging |
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253 |
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254 ``GST_DEBUG_DUMP_DOT_DIR`` environment variable must be set to destination dir. |
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255 ``dot -Tpng -o <filename>.png <filename>.dot`` can be use to convert to a PNG file. |
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256 See |
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257 https://gstreamer.freedesktop.org/documentation/gstreamer/debugutils.html?gi-language=python#GstDebugGraphDetails |
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258 for details. |
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259 |
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260 @param filename: name of the generated file |
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261 @param details: which details to print |
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262 @param with_timestamp: if True, add a timestamp to filename |
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263 @param bin_: which bin to output. By default, the whole pipeline |
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264 (``self.pipeline``) will be used. |
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265 """ |
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266 if bin_ is None: |
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267 bin_ = self.pipeline |
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268 if with_timestamp: |
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269 timestamp = datetime.now().isoformat(timespec="milliseconds") |
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270 filename = f"{timestamp}_filename" |
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271 |
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272 Gst.debug_bin_to_dot_file(bin_, details, filename) |
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273 |
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274 def get_sdp_mline_index(self, media_type: str) -> int: |
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275 """Gets the sdpMLineIndex for a given media type. |
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276 |
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277 @param media_type: The type of the media. |
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278 """ |
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279 for index, m_type in self.media_types.items(): |
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280 if m_type == media_type: |
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281 return index |
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282 raise ValueError(f"Media type '{media_type}' not found") |
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283 |
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284 def _set_media_types(self, offer_sdp: str) -> None: |
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285 """Sets media types from offer SDP |
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286 |
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287 @param offer: RTC session description containing the offer |
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288 """ |
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289 sdp_lines = offer_sdp.splitlines() |
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290 media_types = {} |
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291 mline_index = 0 |
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292 |
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293 for line in sdp_lines: |
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294 if line.startswith("m="): |
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295 media_types[mline_index] = line[2 : line.find(" ")] |
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296 mline_index += 1 |
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297 |
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298 self.media_types = media_types |
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299 |
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300 def _a_call(self, method, *args, **kwargs): |
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301 """Call an async method in main thread""" |
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302 aio.run_from_thread(method, *args, **kwargs, loop=self.main_loop) |
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303 |
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304 def get_payload_types( |
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305 self, sdpmsg, video_encoding: str, audio_encoding: str |
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306 ) -> dict[str, int | None]: |
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307 """Find the payload types for the specified video and audio encoding. |
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308 |
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309 Very simplistically finds the first payload type matching the encoding |
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310 name. More complex applications will want to match caps on |
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311 profile-level-id, packetization-mode, etc. |
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312 """ |
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313 # method coming from gstreamer example (Matthew Waters, Nirbheek Chauhan) at |
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314 # subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py |
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315 video_pt = None |
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316 audio_pt = None |
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317 for i in range(0, sdpmsg.medias_len()): |
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318 media = sdpmsg.get_media(i) |
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319 for j in range(0, media.formats_len()): |
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320 fmt = media.get_format(j) |
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321 if fmt == "webrtc-datachannel": |
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322 continue |
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323 pt = int(fmt) |
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324 caps = media.get_caps_from_media(pt) |
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325 s = caps.get_structure(0) |
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326 encoding_name = s["encoding-name"] |
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327 if video_pt is None and encoding_name == video_encoding: |
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328 video_pt = pt |
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329 elif audio_pt is None and encoding_name == audio_encoding: |
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330 audio_pt = pt |
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331 return {video_encoding: video_pt, audio_encoding: audio_pt} |
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332 |
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333 def parse_ice_candidate(self, candidate_string): |
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334 """Parses the ice candidate string. |
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335 |
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336 @param candidate_string: The ice candidate string to be parsed. |
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337 """ |
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338 pattern = re.compile( |
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339 r"candidate:(?P<foundation>\S+) (?P<component_id>\d+) (?P<transport>\S+) " |
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340 r"(?P<priority>\d+) (?P<address>\S+) (?P<port>\d+) typ " |
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341 r"(?P<type>\S+)(?: raddr (?P<rel_addr>\S+) rport " |
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342 r"(?P<rel_port>\d+))?(?: generation (?P<generation>\d+))?" |
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343 ) |
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344 match = pattern.match(candidate_string) |
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345 if match: |
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346 candidate_dict = match.groupdict() |
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347 |
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348 # Apply the correct types to the dictionary values |
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349 candidate_dict["component_id"] = int(candidate_dict["component_id"]) |
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350 candidate_dict["priority"] = int(candidate_dict["priority"]) |
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351 candidate_dict["port"] = int(candidate_dict["port"]) |
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352 |
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353 if candidate_dict["rel_port"]: |
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354 candidate_dict["rel_port"] = int(candidate_dict["rel_port"]) |
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355 |
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356 if candidate_dict["generation"]: |
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357 candidate_dict["generation"] = candidate_dict["generation"] |
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358 |
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359 # Remove None values |
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360 return {k: v for k, v in candidate_dict.items() if v is not None} |
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361 else: |
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362 log.warning(f"can't parse candidate: {candidate_string!r}") |
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363 return None |
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364 |
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365 def build_ice_candidate(self, parsed_candidate): |
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366 """Builds ICE candidate |
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367 |
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368 @param parsed_candidate: Dictionary containing parsed ICE candidate |
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369 """ |
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370 base_format = ( |
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371 "candidate:{foundation} {component_id} {transport} {priority} " |
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372 "{address} {port} typ {type}" |
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373 ) |
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374 |
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375 if parsed_candidate.get("rel_addr") and parsed_candidate.get("rel_port"): |
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376 base_format += " raddr {rel_addr} rport {rel_port}" |
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377 |
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378 if parsed_candidate.get("generation"): |
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379 base_format += " generation {generation}" |
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380 |
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381 return base_format.format(**parsed_candidate) |
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382 |
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383 def extract_ufrag_pwd(self, sdp: str) -> None: |
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384 """Retrieves ICE password and user fragment for SDP offer. |
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385 |
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386 @param sdp: The Session Description Protocol offer string. |
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387 """ |
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388 lines = sdp.splitlines() |
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389 media = "" |
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390 mid_media_map = {} |
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391 bundle_media = set() |
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392 bundle_ufrag = "" |
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393 bundle_pwd = "" |
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394 in_bundle = False |
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395 |
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396 for line in lines: |
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397 if line.startswith("m="): |
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398 media = line.split("=")[1].split()[0] |
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399 elif line.startswith("a=mid:"): |
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400 mid = line.split(":")[1].strip() |
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401 mid_media_map[mid] = media |
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402 elif line.startswith("a=group:BUNDLE"): |
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403 in_bundle = True |
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404 bundle_media = set(line.split(":")[1].strip().split()) |
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405 elif line.startswith("a=ice-ufrag:"): |
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406 if in_bundle: |
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407 bundle_ufrag = line.split(":")[1].strip() |
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408 else: |
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409 self.ufrag[media] = line.split(":")[1].strip() |
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410 elif line.startswith("a=ice-pwd:"): |
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411 if in_bundle: |
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412 bundle_pwd = line.split(":")[1].strip() |
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413 else: |
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414 self.pwd[media] = line.split(":")[1].strip() |
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415 else: |
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416 in_bundle = False |
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417 |
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418 if bundle_ufrag and bundle_pwd: |
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419 for mid in bundle_media: |
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420 media = mid_media_map[mid] |
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421 self.ufrag[media] = bundle_ufrag |
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422 self.pwd[media] = bundle_pwd |
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423 |
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424 def reset_instance(self): |
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425 """Inits or resets the instance variables to their default state.""" |
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426 self.role: str | None = None |
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427 if self.pipeline is not None: |
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428 self.pipeline.set_state(Gst.State.NULL) |
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429 self.pipeline = None |
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430 self._remote_video_pad = None |
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431 self.sid: str | None = None |
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432 self.offer: str | None = None |
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433 self.local_candidates_buffer = {} |
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434 self.ufrag: dict[str, str] = {} |
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435 self.pwd: dict[str, str] = {} |
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436 self.callee: jid.JID | None = None |
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437 self._media_types = None |
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438 self._media_types_inv = None |
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439 self._sdp_set: bool = False |
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440 self.remote_candidates_buffer: dict[str, dict[str, list]] = { |
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441 "audio": {"candidates": []}, |
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442 "video": {"candidates": []}, |
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443 } |
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444 self._media_types = None |
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445 self._media_types_inv = None |
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446 self.audio_valve = None |
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447 self.video_valve = None |
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448 self.video_selector = None |
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449 if self.desktop_portal is not None: |
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450 self.desktop_portal.end_session() |
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451 self.desktop_sharing = False |
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452 self.desktop_sink_pad = None |
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453 self.bindings = {} |
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454 if self.reset_cb is not None: |
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455 self.reset_cb() |
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456 self.data_channels: dict[str, GstWebRTC.WebRTCDataChannel] = {} |
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457 |
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458 @property |
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459 def data_channel(self) -> GstWebRTC.WebRTCDataChannel: |
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460 """Convenience method to get WebRTCDataChannel instance when there is only one.""" |
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461 if len(self.data_channels) != 1: |
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462 raise exceptions.InternalError( |
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463 "self.data_channel can only be used in a single Data Channel scenario. " |
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464 "Use self.data_channels dict instead." |
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465 ) |
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466 return next(iter(self.data_channels.values())) |
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467 |
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468 async def setup_call( |
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469 self, |
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470 role: str, |
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471 audio_pt: int | None = 96, |
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472 video_pt: int | None = 97, |
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473 ) -> None: |
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474 """Sets up the call. |
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475 |
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476 This method establishes the Gstreamer pipeline for audio and video communication. |
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477 The method also manages STUN and TURN server configurations, signal watchers, and |
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478 various connection handlers for the webrtcbin. |
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479 |
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480 @param role: The role for the call, either 'initiator' or 'responder'. |
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481 @param audio_pt: The payload type for the audio stream. |
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482 @param video_pt: The payload type for the video stream |
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483 |
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484 @raises NotImplementedError: If audio_pt or video_pt is set to None. |
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485 @raises AssertionError: If the role is not 'initiator' or 'responder'. |
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486 """ |
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487 assert role in ("initiator", "responder") |
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488 self.role = role |
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489 |
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490 if isinstance(self.sources_data, SourcesPipeline): |
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491 if self.sources_data.video_pipeline != "" and video_pt is None: |
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492 raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG) |
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493 if self.sources_data.audio_pipeline != "" and audio_pt is None: |
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494 raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG) |
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495 elif isinstance(self.sources_data, SourcesNone): |
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496 pass |
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497 else: |
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498 if audio_pt is None or video_pt is None: |
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499 raise NotImplementedError(NONE_NOT_IMPLEMENTED_MSG) |
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500 |
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501 match self.sources_data: |
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502 case SourcesAuto(): |
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503 video_source_elt = VIDEO_SOURCE_AUTO |
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504 audio_source_elt = AUDIO_SOURCE_AUTO |
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505 case SourcesNone(): |
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506 video_source_elt = "" |
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507 audio_source_elt = "" |
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508 case SourcesPipeline() as source: |
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509 if source.video_pipeline is None: |
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510 video_source_elt = VIDEO_SOURCE_AUTO |
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511 else: |
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512 video_source_elt = source.video_pipeline |
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513 if source.audio_pipeline is None: |
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514 audio_source_elt = AUDIO_SOURCE_AUTO |
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515 else: |
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516 audio_source_elt = source.audio_pipeline |
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517 case SourcesTest(): |
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518 video_source_elt = "videotestsrc is-live=true pattern=ball" |
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519 audio_source_elt = "audiotestsrc" |
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520 case _: |
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521 raise exceptions.InternalError( |
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522 f'Unexpected "sources_data" value: {self.sources_data!r}' |
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523 ) |
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524 |
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525 match self.sinks_data: |
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526 case SinksApp(): |
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527 local_video_sink_elt = ( |
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528 "appsink name=local_video_sink emit-signals=true drop=true " |
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529 "max-buffers=1 sync=True" |
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530 ) |
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531 case SinksAuto(): |
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532 if isinstance(self.sources_data, SourcesNone): |
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533 local_video_sink_elt = "" |
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534 else: |
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535 local_video_sink_elt = "autovideosink" |
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536 case SinksNone(): |
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537 local_video_sink_elt = "" |
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538 case _: |
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539 raise exceptions.InternalError( |
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540 f'Unexpected "sinks_data" value {self.sinks_data!r}' |
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541 ) |
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542 |
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543 gst_pipe_elements = [ |
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544 "webrtcbin latency=30 name=sendrecv bundle-policy=max-bundle" |
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545 ] |
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546 |
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547 if self.merge_pip and local_video_sink_elt: |
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548 # Compositor is used to merge local video feedback in video sink, useful when |
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549 # we have only a single video sink. |
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550 gst_pipe_elements.append( |
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551 "compositor name=compositor background=black " |
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552 f"! video/x-raw,width={self.target_width}," |
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553 f"height={self.target_height},framerate=30/1 " |
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554 f"! {local_video_sink_elt}" |
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555 ) |
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556 local_video_sink_elt = "compositor.sink_1" |
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557 |
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558 if video_source_elt: |
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559 # Video source with an input-selector to switch between normal and video mute |
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560 # (or desktop sharing). |
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561 gst_pipe_elements.append( |
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562 f""" |
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563 input-selector name=video_selector |
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564 ! videorate drop-only=1 max-rate=30 |
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565 ! video/x-raw,framerate=30/1 |
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566 ! tee name=t |
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567 |
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568 {video_source_elt} name=video_src |
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569 ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream |
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570 ! video_selector. |
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571 |
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572 videotestsrc name=muted_src is-live=true pattern=black |
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573 ! queue leaky=downstream |
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574 ! video_selector. |
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575 |
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576 t. |
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577 ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream |
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578 ! videoscale |
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579 ! videoconvert |
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580 ! vp8enc deadline=1 keyframe-max-dist=30 |
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581 ! rtpvp8pay picture-id-mode=15-bit |
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582 ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt} |
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583 ! sendrecv. |
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584 """ |
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585 ) |
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|
586 |
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587 if local_video_sink_elt: |
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588 # Local video feedback. |
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589 gst_pipe_elements.append( |
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590 f""" |
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591 t. |
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592 ! queue max-size-buffers=1 max-size-time=0 max-size-bytes=0 leaky=downstream |
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593 ! videoconvert |
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594 ! {local_video_sink_elt} |
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595 """ |
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596 ) |
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597 |
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598 if audio_source_elt: |
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599 # Audio with a valve for muting. |
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600 gst_pipe_elements.append( |
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601 f""" |
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602 {audio_source_elt} name=audio_src |
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603 ! valve |
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604 ! queue max-size-buffers=10 max-size-time=0 max-size-bytes=0 leaky=downstream |
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605 ! audioconvert |
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606 ! audioresample |
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607 ! opusenc audio-type=voice |
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608 ! rtpopuspay |
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609 ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} |
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610 ! sendrecv. |
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611 """ |
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612 ) |
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613 |
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614 self.gst_pipe_desc = "\n\n".join(gst_pipe_elements) |
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615 |
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616 log.debug(f"Gstreamer pipeline: {self.gst_pipe_desc}") |
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617 |
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618 # Create the pipeline |
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619 try: |
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620 self.pipeline = Gst.parse_launch(self.gst_pipe_desc) |
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621 except Exception: |
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622 log.exception("Can't parse pipeline") |
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623 self.pipeline = None |
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624 if not self.pipeline: |
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625 raise exceptions.InternalError("Failed to create Gstreamer pipeline.") |
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626 |
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627 if not isinstance(self.pipeline, Gst.Pipeline): |
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628 # in the case of Data Channel there is a single element, and Gst.parse_launch |
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629 # doesn't create a Pipeline in this case, so we do it manually. |
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630 pipeline = Gst.Pipeline() |
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631 pipeline.add(self.pipeline) |
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632 self.pipeline = pipeline |
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633 |
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634 self.webrtcbin = self.pipeline.get_by_name("sendrecv") |
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635 if self.webrtcbin is None: |
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636 raise exceptions.InternalError("Can't get the pipeline.") |
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637 |
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638 # If video or audio sources are not created, ``get_by_name`` will return None. |
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639 self.video_src = self.pipeline.get_by_name("video_src") |
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640 self.muted_src = self.pipeline.get_by_name("muted_src") |
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641 self.video_selector = self.pipeline.get_by_name("video_selector") |
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642 if self.video_src and isinstance(self.sources_data, SourcesPipeline): |
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643 for name, value in self.sources_data.video_properties.items(): |
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644 self.video_src.set_property(name, value) |
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645 |
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646 self.audio_src = self.pipeline.get_by_name("audio_src") |
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647 if self.audio_src and isinstance(self.sources_data, SourcesPipeline): |
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648 for name, value in self.sources_data.audio_properties.items(): |
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649 self.audio_src.set_property(name, value) |
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650 |
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651 self.audio_valve = self.pipeline.get_by_name("audio_valve") |
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652 |
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653 if self.video_muted: |
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654 self.on_video_mute(True) |
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655 if self.audio_muted: |
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656 self.on_audio_mute(True) |
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657 |
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658 # set STUN and TURN servers |
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659 external_disco = data_format.deserialise( |
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660 await self.bridge.external_disco_get("", self.profile), type_check=list |
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661 ) |
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662 |
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663 for server in external_disco: |
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664 if server["type"] == "stun": |
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665 if server["transport"] == "tcp": |
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666 log.info( |
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667 "ignoring TCP STUN server, GStreamer only support one STUN server" |
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668 ) |
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669 url = f"stun://{server['host']}:{server['port']}" |
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670 log.debug(f"adding stun server: {url}") |
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671 self.webrtcbin.set_property("stun-server", url) |
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672 elif server["type"] == "turn": |
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673 url = "{scheme}://{username}:{password}@{host}:{port}".format( |
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674 scheme="turns" if server["transport"] == "tcp" else "turn", |
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675 username=quote_plus(server["username"]), |
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676 password=quote_plus(server["password"]), |
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677 host=server["host"], |
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678 port=server["port"], |
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679 ) |
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680 log.debug(f"adding turn server: {url}") |
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681 |
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682 if not self.webrtcbin.emit("add-turn-server", url): |
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683 log.warning(f"Erreur while adding TURN server {url}") |
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684 |
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685 # local video feedback |
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686 if isinstance(self.sinks_data, SinksApp): |
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687 local_video_sink = self.pipeline.get_by_name("local_video_sink") |
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688 if local_video_sink is not None: |
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689 local_video_sink.set_property("emit-signals", True) |
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690 local_video_sink.connect("new-sample", self.sinks_data.local_video_cb) |
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691 local_video_sink_caps = Gst.Caps.from_string(f"video/x-raw,format=RGB") |
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692 local_video_sink.set_property("caps", local_video_sink_caps) |
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693 |
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694 # Create bus and associate signal watchers |
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695 self.bus = self.pipeline.get_bus() |
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696 if not self.bus: |
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697 log.error("Failed to get bus from pipeline.") |
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698 return |
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699 |
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700 self.bus.add_signal_watch() |
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701 self.webrtcbin.connect("pad-added", self.on_pad_added) |
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702 self.bus.connect("message::error", self.on_bus_error) |
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703 self.bus.connect("message::eos", self.on_bus_eos) |
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704 self.webrtcbin.connect("on-negotiation-needed", self.on_negotiation_needed) |
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|
705 self.webrtcbin.connect("on-ice-candidate", self.on_ice_candidate) |
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|
706 self.webrtcbin.connect( |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
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|
707 "notify::ice-gathering-state", self.on_ice_gathering_state_change |
6745c6bd4c7a
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parents:
diff
changeset
|
708 ) |
6745c6bd4c7a
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parents:
diff
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|
709 self.webrtcbin.connect( |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
710 "notify::ice-connection-state", self.on_ice_connection_state |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
parents:
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|
711 ) |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
parents:
diff
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|
712 |
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|
713 for dc_data in self.dc_data_list: |
79c8a70e1813
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diff
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|
714 self.create_data_channel(dc_data) |
79c8a70e1813
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4233
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|
715 |
4270
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Reformatted code base using black.
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4240
diff
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|
716 def create_data_channel(self, dc_data: SourcesDataChannel | SinksDataChannel) -> None: |
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|
717 """Create a Data Channel and connect relevant callbacks.""" |
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718 assert self.pipeline is not None |
79c8a70e1813
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diff
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|
719 if isinstance(dc_data, SourcesDataChannel): |
4233
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parents:
4209
diff
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|
720 # Data channel configuration for compatibility with browser defaults |
d01b8d002619
cli (call, file), frontends: implement webRTC data channel transfer:
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parents:
4209
diff
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|
721 data_channel_options = Gst.Structure.new_empty("data-channel-options") |
d01b8d002619
cli (call, file), frontends: implement webRTC data channel transfer:
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parents:
4209
diff
changeset
|
722 data_channel_options.set_value("ordered", True) |
d01b8d002619
cli (call, file), frontends: implement webRTC data channel transfer:
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parents:
4209
diff
changeset
|
723 data_channel_options.set_value("protocol", "") |
d01b8d002619
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Goffi <goffi@goffi.org>
parents:
4209
diff
changeset
|
724 |
d01b8d002619
cli (call, file), frontends: implement webRTC data channel transfer:
Goffi <goffi@goffi.org>
parents:
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diff
changeset
|
725 # Create the data channel |
d01b8d002619
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changeset
|
726 self.pipeline.set_state(Gst.State.READY) |
4240
79c8a70e1813
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|
727 self.data_channels[dc_data.name] = data_channel = self.webrtcbin.emit( |
79c8a70e1813
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parents:
4233
diff
changeset
|
728 "create-data-channel", dc_data.name, data_channel_options |
4233
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diff
changeset
|
729 ) |
4240
79c8a70e1813
backend, frontend: prepare remote control:
Goffi <goffi@goffi.org>
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|
730 if data_channel is None: |
4233
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parents:
4209
diff
changeset
|
731 log.error("Failed to create data channel") |
d01b8d002619
cli (call, file), frontends: implement webRTC data channel transfer:
Goffi <goffi@goffi.org>
parents:
4209
diff
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|
732 return |
4240
79c8a70e1813
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4233
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changeset
|
733 data_channel.connect("on-open", dc_data.dc_open_cb) |
79c8a70e1813
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Goffi <goffi@goffi.org>
parents:
4233
diff
changeset
|
734 elif isinstance(dc_data, SinksDataChannel): |
79c8a70e1813
backend, frontend: prepare remote control:
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parents:
4233
diff
changeset
|
735 self.webrtcbin.connect("on-data-channel", dc_data.dc_on_data_channel) |
79c8a70e1813
backend, frontend: prepare remote control:
Goffi <goffi@goffi.org>
parents:
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|
736 else: |
4270
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parents:
4240
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|
737 raise ValueError("Only SourcesDataChannel or SinksDataChannel are allowed.") |
4233
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|
738 |
4139
6745c6bd4c7a
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diff
changeset
|
739 def start_pipeline(self) -> None: |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
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|
740 """Starts the GStreamer pipeline.""" |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
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|
741 log.debug("starting the pipeline") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
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|
742 self.pipeline.set_state(Gst.State.PLAYING) |
6745c6bd4c7a
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|
743 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
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|
744 def on_negotiation_needed(self, webrtc): |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
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|
745 """Initiate SDP offer when negotiation is needed.""" |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
746 log.debug("Negotiation needed.") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
747 if self.role == "initiator": |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
748 log.debug("Creating offer…") |
6745c6bd4c7a
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parents:
diff
changeset
|
749 promise = Gst.Promise.new_with_change_func(self.on_offer_created) |
6745c6bd4c7a
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parents:
diff
changeset
|
750 self.webrtcbin.emit("create-offer", None, promise) |
6745c6bd4c7a
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diff
changeset
|
751 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
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diff
changeset
|
752 def on_offer_created(self, promise): |
6745c6bd4c7a
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diff
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|
753 """Callback for when SDP offer is created.""" |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
754 log.info("on_offer_created called") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
755 assert promise.wait() == Gst.PromiseResult.REPLIED |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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changeset
|
756 reply = promise.get_reply() |
6745c6bd4c7a
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diff
changeset
|
757 if reply is None: |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
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diff
changeset
|
758 log.error("Promise reply is None. Offer creation might have failed.") |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
759 return |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
760 offer = reply["offer"] |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
parents:
diff
changeset
|
761 self.offer = offer.sdp.as_text() |
6745c6bd4c7a
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parents:
diff
changeset
|
762 log.info(f"SDP offer created: \n{self.offer}") |
6745c6bd4c7a
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parents:
diff
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|
763 self._set_media_types(self.offer) |
6745c6bd4c7a
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diff
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|
764 promise = Gst.Promise.new() |
6745c6bd4c7a
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parents:
diff
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|
765 self.webrtcbin.emit("set-local-description", offer, promise) |
6745c6bd4c7a
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parents:
diff
changeset
|
766 promise.interrupt() |
6745c6bd4c7a
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parents:
diff
changeset
|
767 self._a_call(self._start_call) |
6745c6bd4c7a
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diff
changeset
|
768 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
Goffi <goffi@goffi.org>
parents:
diff
changeset
|
769 def on_answer_set(self, promise): |
6745c6bd4c7a
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parents:
diff
changeset
|
770 assert promise.wait() == Gst.PromiseResult.REPLIED |
6745c6bd4c7a
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|
771 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
772 def on_answer_created(self, promise, _, __): |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
773 """Callback for when SDP answer is created.""" |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
774 assert promise.wait() == Gst.PromiseResult.REPLIED |
6745c6bd4c7a
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parents:
diff
changeset
|
775 reply = promise.get_reply() |
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|
776 answer = reply["answer"] |
6745c6bd4c7a
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diff
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|
777 promise = Gst.Promise.new() |
6745c6bd4c7a
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|
778 self.webrtcbin.emit("set-local-description", answer, promise) |
6745c6bd4c7a
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changeset
|
779 promise.interrupt() |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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|
780 answer_sdp = answer.sdp.as_text() |
6745c6bd4c7a
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changeset
|
781 log.info(f"SDP answer set: \n{answer_sdp}") |
6745c6bd4c7a
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parents:
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|
782 self.sdp_set = True |
6745c6bd4c7a
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|
783 self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile) |
6745c6bd4c7a
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|
784 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
785 def on_offer_set(self, promise): |
6745c6bd4c7a
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parents:
diff
changeset
|
786 assert promise.wait() == Gst.PromiseResult.REPLIED |
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parents:
diff
changeset
|
787 promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None) |
6745c6bd4c7a
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diff
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|
788 self.webrtcbin.emit("create-answer", None, promise) |
6745c6bd4c7a
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|
789 |
6745c6bd4c7a
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diff
changeset
|
790 def link_element_or_pad( |
6745c6bd4c7a
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changeset
|
791 self, source: Gst.Element, dest: Gst.Element | Gst.Pad |
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diff
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|
792 ) -> bool: |
6745c6bd4c7a
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parents:
diff
changeset
|
793 """Check if dest is a pad or an element, and link appropriately""" |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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|
794 src_pad = source.get_static_pad("src") |
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changeset
|
795 |
6745c6bd4c7a
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changeset
|
796 if isinstance(dest, Gst.Pad): |
6745c6bd4c7a
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diff
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|
797 # If the dest is a pad, link directly |
6745c6bd4c7a
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parents:
diff
changeset
|
798 if not src_pad.link(dest) == Gst.PadLinkReturn.OK: |
6745c6bd4c7a
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parents:
diff
changeset
|
799 log.error( |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
800 "Failed to link 'conv' to the compositor's newly requested pad!" |
6745c6bd4c7a
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parents:
diff
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|
801 ) |
6745c6bd4c7a
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diff
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|
802 return False |
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
803 elif isinstance(dest, Gst.Element): |
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frontends (tools): webrtc implementation:
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parents:
diff
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|
804 return source.link(dest) |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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diff
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|
805 else: |
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4209
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changeset
|
806 log.error(f"Unexpected type for dest: {type(dest)}") |
4139
6745c6bd4c7a
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parents:
diff
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|
807 return False |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
808 |
6745c6bd4c7a
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parents:
diff
changeset
|
809 return True |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
810 |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
811 def scaled_dimensions( |
6745c6bd4c7a
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parents:
diff
changeset
|
812 self, original_width: int, original_height: int, max_width: int, max_height: int |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
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|
813 ) -> tuple[int, int]: |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
814 """Calculates the scaled dimensions preserving aspect ratio. |
6745c6bd4c7a
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changeset
|
815 |
6745c6bd4c7a
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parents:
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changeset
|
816 @param original_width: Original width of the video stream. |
6745c6bd4c7a
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parents:
diff
changeset
|
817 @param original_height: Original height of the video stream. |
6745c6bd4c7a
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parents:
diff
changeset
|
818 @param max_width: Maximum desired width for the scaled video. |
6745c6bd4c7a
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parents:
diff
changeset
|
819 @param max_height: Maximum desired height for the scaled video. |
6745c6bd4c7a
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parents:
diff
changeset
|
820 @return: The width and height of the scaled video. |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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diff
changeset
|
821 """ |
6745c6bd4c7a
frontends (tools): webrtc implementation:
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parents:
diff
changeset
|
822 aspect_ratio = original_width / original_height |
6745c6bd4c7a
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823 new_width = int(max_height * aspect_ratio) |
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824 |
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825 if new_width <= max_width: |
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826 return new_width, max_height |
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827 |
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828 new_height = int(max_width / aspect_ratio) |
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829 return max_width, new_height |
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830 |
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831 def on_remote_decodebin_stream(self, _, pad: Gst.Pad) -> None: |
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832 """Handle the stream from the remote decodebin. |
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833 |
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834 This method processes the incoming stream from the remote decodebin, determining |
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835 whether it's video or audio. It then sets up the appropriate GStreamer elements |
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836 for video/audio processing and adds them to the pipeline. |
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837 |
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838 @param pad: The Gst.Pad from the remote decodebin producing the stream. |
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839 """ |
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840 assert self.pipeline is not None |
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841 if not pad.has_current_caps(): |
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842 log.error(f"{pad} has no caps, ignoring") |
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843 return |
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844 |
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845 caps = pad.get_current_caps() |
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846 assert len(caps) |
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847 s = caps[0] |
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848 name = s.get_name() |
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|
849 log.debug(f"====> NAME START: {name}") |
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850 |
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851 q = Gst.ElementFactory.make("queue") |
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852 |
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|
853 if name.startswith("video"): |
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854 log.debug("===> VIDEO OK") |
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855 |
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|
856 self._remote_video_pad = pad |
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|
857 |
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858 # Check and log the original size of the video |
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859 width = self.target_width |
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860 height = self.target_height |
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861 log.info(f"Original video size: {width}x{height}") |
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862 |
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863 # This is a fix for an issue found with Movim on desktop: a non standard |
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864 # resolution is used (990x557) resulting in bad alignement and no color in |
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865 # rendered image |
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866 adjust_resolution = width % 4 != 0 or height % 4 != 0 |
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|
867 if adjust_resolution: |
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868 log.info("non standard resolution, we need to adjust size") |
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869 width = (width + 3) // 4 * 4 |
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870 height = (height + 3) // 4 * 4 |
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871 log.info(f"Adjusted video size: {width}x{height}") |
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|
872 |
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|
873 conv = Gst.ElementFactory.make("videoconvert") |
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874 if self.merge_pip: |
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875 # with ``merge_pip`` set, we plug the remote stream to the composer |
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876 compositor = self.pipeline.get_by_name("compositor") |
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877 |
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|
878 sink1_pad = compositor.get_static_pad("sink_1") |
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|
879 |
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|
880 local_width, local_height = self.scaled_dimensions( |
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|
881 sink1_pad.get_property("width"), |
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|
882 sink1_pad.get_property("height"), |
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|
883 width // 3, |
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|
884 height // 3, |
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|
885 ) |
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|
886 |
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|
887 sink1_pad.set_property("xpos", width - local_width) |
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|
888 sink1_pad.set_property("ypos", height - local_height) |
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|
889 sink1_pad.set_property("width", local_width) |
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|
890 sink1_pad.set_property("height", local_height) |
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|
891 sink1_pad.set_property("sizing-policy", 1) |
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|
892 sink1_pad.set_property("zorder", 1) |
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|
893 |
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|
894 # Request a new pad for the remote stream |
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|
895 sink_pad_template = compositor.get_pad_template("sink_%u") |
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896 remote_video_sink = compositor.request_pad(sink_pad_template, None, None) |
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|
897 remote_video_sink.set_property("zorder", 0) |
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898 remote_video_sink.set_property("width", width) |
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|
899 remote_video_sink.set_property("height", height) |
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900 remote_video_sink.set_property("sizing-policy", 1) |
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|
901 elif isinstance(self.sinks_data, SinksApp): |
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902 # ``app`` sink without ``self.merge_pip`` set, be create the sink and |
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|
903 # connect it to the ``remote_video_cb``. |
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|
904 remote_video_sink = Gst.ElementFactory.make("appsink") |
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|
905 |
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|
906 remote_video_caps = Gst.Caps.from_string("video/x-raw,format=RGB") |
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|
907 remote_video_sink.set_property("caps", remote_video_caps) |
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908 |
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|
909 remote_video_sink.set_property("emit-signals", True) |
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|
910 remote_video_sink.set_property("drop", True) |
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|
911 remote_video_sink.set_property("max-buffers", 1) |
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|
912 remote_video_sink.set_property("sync", True) |
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|
913 remote_video_sink.connect("new-sample", self.sinks_data.remote_video_cb) |
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|
914 self.pipeline.add(remote_video_sink) |
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|
915 elif isinstance(self.sinks_data, SinksAuto): |
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|
916 # if ``self.merge_pip`` is not set, we create a dedicated |
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|
917 # ``autovideosink`` for remote stream. |
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|
918 remote_video_sink = Gst.ElementFactory.make("autovideosink") |
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|
919 self.pipeline.add(remote_video_sink) |
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|
920 else: |
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|
921 raise exceptions.InternalError( |
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|
922 f'Unhandled "sinks_data" value: {self.sinks_data!r}' |
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|
923 ) |
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|
924 |
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|
925 if adjust_resolution: |
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|
926 videoscale = Gst.ElementFactory.make("videoscale") |
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|
927 adjusted_caps = Gst.Caps.from_string( |
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|
928 f"video/x-raw,width={width},height={height}" |
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|
929 ) |
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|
930 capsfilter = Gst.ElementFactory.make("capsfilter") |
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|
931 capsfilter.set_property("caps", adjusted_caps) |
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|
932 |
6745c6bd4c7a
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|
933 self.pipeline.add(q, conv, videoscale, capsfilter) |
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|
934 |
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|
935 self.pipeline.sync_children_states() |
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|
936 ret = pad.link(q.get_static_pad("sink")) |
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937 if ret != Gst.PadLinkReturn.OK: |
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|
938 log.error(f"Error linking pad: {ret}") |
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|
939 q.link(conv) |
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|
940 conv.link(videoscale) |
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|
941 videoscale.link(capsfilter) |
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|
942 self.link_element_or_pad(capsfilter.link, remote_video_sink) |
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|
943 |
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|
944 else: |
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|
945 self.pipeline.add(q, conv) |
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|
946 |
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|
947 self.pipeline.sync_children_states() |
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|
948 ret = pad.link(q.get_static_pad("sink")) |
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|
949 if ret != Gst.PadLinkReturn.OK: |
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|
950 log.error(f"Error linking pad: {ret}") |
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|
951 q.link(conv) |
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|
952 self.link_element_or_pad(conv, remote_video_sink) |
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|
953 |
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|
954 elif name.startswith("audio"): |
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|
955 log.debug("===> Audio OK") |
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956 conv = Gst.ElementFactory.make("audioconvert") |
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957 resample = Gst.ElementFactory.make("audioresample") |
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958 remote_audio_sink = Gst.ElementFactory.make("autoaudiosink") |
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|
959 self.pipeline.add(q, conv, resample, remote_audio_sink) |
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|
960 self.pipeline.sync_children_states() |
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|
961 ret = pad.link(q.get_static_pad("sink")) |
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|
962 if ret != Gst.PadLinkReturn.OK: |
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|
963 log.error(f"Error linking pad: {ret}") |
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|
964 q.link(conv) |
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|
965 conv.link(resample) |
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|
966 resample.link(remote_audio_sink) |
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|
967 |
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|
968 else: |
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|
969 log.warning(f"unmanaged name: {name!r}") |
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|
970 |
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|
971 def on_pad_added(self, __, pad: Gst.Pad) -> None: |
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972 """Handle the addition of a new pad to the element. |
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|
973 |
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diff
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|
974 When a new source pad is added to the element, this method creates a decodebin, |
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|
975 connects it to handle the stream, and links the pad to the decodebin. |
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|
976 |
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|
977 @param __: Placeholder for the signal source. Not used in this method. |
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|
978 @param pad: The newly added pad. |
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|
979 """ |
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|
980 log.debug("on_pad_added") |
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|
981 if pad.direction != Gst.PadDirection.SRC: |
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diff
changeset
|
982 return |
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|
983 |
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|
984 decodebin = Gst.ElementFactory.make("decodebin") |
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|
985 decodebin.connect("pad-added", self.on_remote_decodebin_stream) |
6745c6bd4c7a
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diff
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|
986 self.pipeline.add(decodebin) |
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|
987 decodebin.sync_state_with_parent() |
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|
988 pad.link(decodebin.get_static_pad("sink")) |
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|
989 |
4233
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diff
changeset
|
990 async def _call_start(self, callee: jid.JID, call_data: dict, profile: str) -> str: |
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4209
diff
changeset
|
991 return await self.bridge.call_start( |
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parents:
4209
diff
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|
992 str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile |
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diff
changeset
|
993 ) |
d01b8d002619
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parents:
4209
diff
changeset
|
994 |
4139
6745c6bd4c7a
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|
995 async def _start_call(self) -> None: |
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diff
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|
996 """Initiate the call. |
6745c6bd4c7a
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diff
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|
997 |
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parents:
diff
changeset
|
998 Initiates a call with the callee using the stored offer. If there are any buffered |
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changeset
|
999 local ICE candidates, they are sent as part of the initiation. |
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|
1000 """ |
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|
1001 assert self.callee |
4233
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|
1002 assert self.call_start_cb is not None |
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|
1003 self.sid = await self.call_start_cb( |
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|
1004 self.callee, {"sdp": self.offer}, self.profile |
4139
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|
1005 ) |
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|
1006 if self.local_candidates_buffer: |
6745c6bd4c7a
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|
1007 log.debug( |
6745c6bd4c7a
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diff
changeset
|
1008 f"sending buffered local ICE candidates: {self.local_candidates_buffer}" |
6745c6bd4c7a
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|
1009 ) |
4240
79c8a70e1813
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4233
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|
1010 if not self.pwd: |
4139
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|
1011 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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|
1012 self.extract_ufrag_pwd(sdp) |
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|
1013 ice_data = {} |
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|
1014 for media_type, candidates in self.local_candidates_buffer.items(): |
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|
1015 ice_data[media_type] = { |
4240
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|
1016 "ufrag": self.ufrag[media_type], |
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|
1017 "pwd": self.pwd[media_type], |
4139
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|
1018 "candidates": candidates, |
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|
1019 } |
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|
1020 await self.bridge.ice_candidates_add( |
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|
1021 self.sid, data_format.serialise(ice_data), self.profile |
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|
1022 ) |
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|
1023 self.local_candidates_buffer.clear() |
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|
1024 |
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|
1025 def _remote_sdp_set(self, promise) -> None: |
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|
1026 assert promise.wait() == Gst.PromiseResult.REPLIED |
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|
1027 self.sdp_set = True |
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|
1028 |
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|
1029 def on_accepted_call(self, sdp: str, profile: str) -> None: |
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|
1030 """Outgoing call has been accepted. |
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|
1031 |
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|
1032 @param sdp: The SDP answer string received from the other party. |
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|
1033 @param profile: Profile used for the call. |
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|
1034 """ |
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|
1035 log.debug(f"SDP answer received: \n{sdp}") |
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|
1036 |
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|
1037 __, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp) |
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|
1038 answer = GstWebRTC.WebRTCSessionDescription.new( |
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|
1039 GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg |
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|
1040 ) |
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|
1041 promise = Gst.Promise.new_with_change_func(self._remote_sdp_set) |
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|
1042 self.webrtcbin.emit("set-remote-description", answer, promise) |
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|
1043 |
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|
1044 async def answer_call(self, sdp: str, profile: str) -> None: |
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|
1045 """Answer an incoming call |
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|
1046 |
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|
1047 @param sdp: The SDP offer string received from the initiator. |
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|
1048 @param profile: Profile used for the call. |
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|
1049 |
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|
1050 @raise AssertionError: Raised when either "VP8" or "OPUS" is not present in |
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|
1051 payload types. |
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|
1052 """ |
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|
1053 log.debug(f"SDP offer received: \n{sdp}") |
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|
1054 self._set_media_types(sdp) |
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|
1055 __, offer_sdp_msg = GstSdp.SDPMessage.new_from_text(sdp) |
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|
1056 payload_types = self.get_payload_types( |
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|
1057 offer_sdp_msg, video_encoding="VP8", audio_encoding="OPUS" |
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|
1058 ) |
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|
1059 assert "VP8" in payload_types |
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|
1060 assert "OPUS" in payload_types |
4240
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1061 try: |
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|
1062 await self.setup_call( |
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1063 "responder", audio_pt=payload_types["OPUS"], video_pt=payload_types["VP8"] |
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1064 ) |
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1065 except Exception: |
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1066 log.exception("Can't setup call") |
79c8a70e1813
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1067 raise |
4139
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|
1068 self.start_pipeline() |
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Goffi <goffi@goffi.org>
parents:
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|
1069 offer = GstWebRTC.WebRTCSessionDescription.new( |
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parents:
diff
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|
1070 GstWebRTC.WebRTCSDPType.OFFER, offer_sdp_msg |
6745c6bd4c7a
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|
1071 ) |
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|
1072 promise = Gst.Promise.new_with_change_func(self.on_offer_set) |
6745c6bd4c7a
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diff
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|
1073 self.webrtcbin.emit("set-remote-description", offer, promise) |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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|
1074 |
6745c6bd4c7a
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diff
changeset
|
1075 def on_ice_candidate(self, webrtc, mline_index, candidate_sdp): |
6745c6bd4c7a
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|
1076 """Handles the on-ice-candidate signal of webrtcbin. |
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|
1077 |
6745c6bd4c7a
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1078 @param webrtc: The webrtcbin element. |
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|
1079 @param mlineindex: The mline index. |
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|
1080 @param candidate: The ICE candidate. |
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|
1081 """ |
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1082 log.debug( |
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Goffi <goffi@goffi.org>
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diff
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|
1083 f"Local ICE candidate. MLine Index: {mline_index}, Candidate: {candidate_sdp}" |
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1084 ) |
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|
1085 parsed_candidate = self.parse_ice_candidate(candidate_sdp) |
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|
1086 if parsed_candidate is None: |
d01b8d002619
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1087 log.warning(f"Can't parse candidate: {candidate_sdp}") |
d01b8d002619
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diff
changeset
|
1088 return |
4139
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changeset
|
1089 try: |
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diff
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|
1090 media_type = self.media_types[mline_index] |
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
1091 except KeyError: |
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Goffi <goffi@goffi.org>
parents:
diff
changeset
|
1092 raise exceptions.InternalError("can't find media type") |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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|
1093 |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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|
1094 if self.sid is None: |
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Goffi <goffi@goffi.org>
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diff
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|
1095 log.debug("buffering local ICE candidate") |
6745c6bd4c7a
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|
1096 self.local_candidates_buffer.setdefault(media_type, []).append( |
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diff
changeset
|
1097 parsed_candidate |
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|
1098 ) |
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parents:
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|
1099 else: |
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|
1100 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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1101 assert sdp is not None |
4240
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1102 self.extract_ufrag_pwd(sdp) |
79c8a70e1813
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1103 ice_data = { |
79c8a70e1813
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|
1104 "ufrag": self.ufrag[media_type], |
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|
1105 "pwd": self.pwd[media_type], |
4270
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Reformatted code base using black.
Goffi <goffi@goffi.org>
parents:
4240
diff
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|
1106 "candidates": [parsed_candidate], |
4240
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1107 } |
4139
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|
1108 self._a_call( |
6745c6bd4c7a
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diff
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|
1109 self.bridge.ice_candidates_add, |
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changeset
|
1110 self.sid, |
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|
1111 data_format.serialise({media_type: ice_data}), |
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Goffi <goffi@goffi.org>
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|
1112 self.profile, |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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diff
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|
1113 ) |
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Goffi <goffi@goffi.org>
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diff
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|
1114 |
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|
1115 def on_ice_candidates_new(self, candidates: dict) -> None: |
6745c6bd4c7a
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Goffi <goffi@goffi.org>
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|
1116 """Handle new ICE candidates. |
6745c6bd4c7a
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|
1117 |
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diff
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|
1118 @param candidates: A dictionary containing media types ("audio" or "video") as |
6745c6bd4c7a
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|
1119 keys and corresponding ICE data as values. |
6745c6bd4c7a
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|
1120 |
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|
1121 @raise exceptions.InternalError: Raised when sdp mline index is not found. |
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|
1122 """ |
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|
1123 if not self.sdp_set: |
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|
1124 log.debug("buffering remote ICE candidate") |
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|
1125 for media_type in ("audio", "video"): |
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|
1126 media_candidates = candidates.get(media_type) |
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|
1127 if media_candidates: |
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|
1128 buffer = self.remote_candidates_buffer[media_type] |
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|
1129 buffer["candidates"].extend(media_candidates["candidates"]) |
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|
1130 return |
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|
1131 for media_type, ice_data in candidates.items(): |
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|
1132 for candidate in ice_data["candidates"]: |
6745c6bd4c7a
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|
1133 candidate_sdp = self.build_ice_candidate(candidate) |
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|
1134 try: |
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|
1135 mline_index = self.get_sdp_mline_index(media_type) |
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|
1136 except Exception as e: |
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|
1137 raise exceptions.InternalError(f"Can't find sdp mline index: {e}") |
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|
1138 self.webrtcbin.emit("add-ice-candidate", mline_index, candidate_sdp) |
4233
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1139 log.warning( |
4139
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|
1140 f"Remote ICE candidate added. MLine Index: {mline_index}, " |
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|
1141 f"{candidate_sdp}" |
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|
1142 ) |
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|
1143 |
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|
1144 def on_ice_gathering_state_change(self, pspec, __): |
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1145 state = self.webrtcbin.get_property("ice-gathering-state") |
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|
1146 log.debug(f"ICE gathering state changed to {state}") |
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|
1147 |
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|
1148 def on_ice_connection_state(self, pspec, __): |
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|
1149 state = self.webrtcbin.props.ice_connection_state |
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|
1150 if state == GstWebRTC.WebRTCICEConnectionState.FAILED: |
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|
1151 log.error("ICE connection failed") |
6745c6bd4c7a
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|
1152 log.info(f"ICE connection state changed to {state}") |
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|
1153 |
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|
1154 def on_bus_error(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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|
1155 """Handles the GStreamer bus error messages. |
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|
1156 |
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|
1157 @param bus: The GStreamer bus. |
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|
1158 @param message: The error message. |
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|
1159 """ |
6745c6bd4c7a
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|
1160 err, debug = message.parse_error() |
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|
1161 log.error(f"Error from {message.src.get_name()}: {err.message}") |
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|
1162 log.error(f"Debugging info: {debug}") |
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|
1163 |
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|
1164 def on_bus_eos(self, bus: Gst.Bus, message: Gst.Message) -> None: |
6745c6bd4c7a
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|
1165 """Handles the GStreamer bus eos messages. |
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|
1166 |
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|
1167 @param bus: The GStreamer bus. |
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|
1168 @param message: The eos message. |
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|
1169 """ |
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|
1170 log.info("End of stream") |
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|
1171 |
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|
1172 def on_audio_mute(self, muted: bool) -> None: |
4204
879bad48cc2d
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parents:
4146
diff
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|
1173 """Handles (un)muting of audio. |
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1174 |
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1175 @param muted: True if audio is muted. |
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1176 """ |
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1177 if self.audio_valve is not None: |
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1178 self.audio_valve.set_property("drop", muted) |
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1179 state = "muted" if muted else "unmuted" |
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1180 log.info(f"audio is now {state}") |
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1181 |
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1182 def on_video_mute(self, muted: bool) -> None: |
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1183 """Handles (un)muting of video. |
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1184 |
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1185 @param muted: True if video is muted. |
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1186 """ |
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1187 if self.video_selector is not None: |
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1188 current_source = ( |
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1189 None if muted else "desktop" if self.desktop_sharing else "video" |
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1190 ) |
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1191 self.switch_video_source(current_source) |
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1192 state = "muted" if muted else "unmuted" |
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1193 log.info(f"Video is now {state}") |
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1194 |
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1195 def on_desktop_switch(self, desktop_active: bool) -> None: |
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1196 """Switches the video source between desktop and video. |
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1197 |
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1198 @param desktop_active: True if desktop must be active. False for video. |
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1199 """ |
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1200 if desktop_active and self.desktop_portal is not None: |
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1201 aio.run_async(self.on_desktop_switch_portal(desktop_active)) |
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1202 else: |
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1203 self.do_desktop_switch(desktop_active) |
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1204 |
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1205 async def on_desktop_switch_portal(self, desktop_active: bool) -> None: |
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1206 """Call freedesktop screenshare portal and the activate the shared stream""" |
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1207 assert self.desktop_portal is not None |
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1208 try: |
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1209 screenshare_data = await self.desktop_portal.request_screenshare() |
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1210 except exceptions.CancelError: |
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1211 self.desktop_sharing = False |
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1212 return |
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1213 self.desktop_sharing_data = {"path": str(screenshare_data["node_id"])} |
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1214 self.do_desktop_switch(desktop_active) |
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1215 |
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1216 def on_portal_session_closed(self) -> None: |
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1217 self.desktop_sharing = False |
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1218 |
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1219 def do_desktop_switch(self, desktop_active: bool) -> None: |
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1220 if self.video_muted: |
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1221 # Update the active source state but do not switch |
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1222 self.desktop_sharing = desktop_active |
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1223 return |
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1224 |
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1225 source = "desktop" if desktop_active else "video" |
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1226 self.switch_video_source(source) |
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1227 self.desktop_sharing = desktop_active |
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1228 |
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1229 def switch_video_source(self, source: str | None) -> None: |
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1230 """Activates the specified source while deactivating the others. |
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1231 |
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1232 @param source: 'desktop', 'video', 'muted' or None for muted source. |
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1233 """ |
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1234 if source is None: |
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1235 source = "muted" |
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1236 if source not in ["video", "muted", "desktop"]: |
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1237 raise ValueError( |
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1238 f"Invalid source: {source!r}, use one of {'video', 'muted', 'desktop'}" |
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1239 ) |
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1240 |
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1241 self.pipeline.set_state(Gst.State.PAUSED) |
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1242 |
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1243 # Create a new desktop source if necessary |
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1244 if source == "desktop": |
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1245 self._setup_desktop_source(self.desktop_sharing_data) |
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1246 |
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1247 # Activate the chosen source and deactivate the others |
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1248 for src_name in ["video", "muted", "desktop"]: |
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1249 src_element = self.pipeline.get_by_name(f"{src_name}_src") |
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1250 if src_name == source: |
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1251 if src_element: |
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1252 src_element.set_state(Gst.State.PLAYING) |
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1253 else: |
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1254 if src_element: |
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1255 if src_name == "desktop": |
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1256 self._remove_desktop_source(src_element) |
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1257 else: |
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1258 src_element.set_state(Gst.State.NULL) |
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1259 |
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1260 # Set the video_selector active pad |
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1261 if source == "desktop": |
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1262 if self.desktop_sink_pad: |
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1263 pad = self.desktop_sink_pad |
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1264 else: |
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1265 log.error(f"No desktop pad available") |
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1266 pad = None |
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1267 else: |
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1268 pad_name = f"sink_{['video', 'muted'].index(source)}" |
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1269 pad = self.video_selector.get_static_pad(pad_name) |
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1270 |
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1271 if pad is not None: |
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1272 self.video_selector.props.active_pad = pad |
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1273 |
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1274 self.pipeline.set_state(Gst.State.PLAYING) |
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1275 |
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1276 def _setup_desktop_source(self, properties: dict[str, object] | None) -> None: |
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1277 """Set up a new desktop source. |
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1278 |
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1279 @param properties: The properties to set on the desktop source. |
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1280 """ |
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1281 source_elt = "ximagesrc" if self.desktop_portal is None else "pipewiresrc" |
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1282 desktop_src = Gst.ElementFactory.make(source_elt, "desktop_src") |
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1283 if properties is None: |
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1284 properties = {} |
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1285 for key, value in properties.items(): |
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1286 log.debug(f"setting {source_elt} property: {key!r}={value!r}") |
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1287 desktop_src.set_property(key, value) |
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1288 video_convert = Gst.ElementFactory.make("videoconvert", "desktop_videoconvert") |
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1289 queue = Gst.ElementFactory.make("queue", "desktop_queue") |
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1290 queue.set_property("leaky", "downstream") |
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1291 |
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1292 self.pipeline.add(desktop_src) |
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1293 self.pipeline.add(video_convert) |
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1294 self.pipeline.add(queue) |
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1295 |
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1296 desktop_src.link(video_convert) |
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1297 video_convert.link(queue) |
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1298 |
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1299 sink_pad_template = self.video_selector.get_pad_template("sink_%u") |
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1300 self.desktop_sink_pad = self.video_selector.request_pad( |
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1301 sink_pad_template, None, None |
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1302 ) |
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1303 queue_src_pad = queue.get_static_pad("src") |
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1304 queue_src_pad.link(self.desktop_sink_pad) |
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1305 |
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1306 desktop_src.sync_state_with_parent() |
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1307 video_convert.sync_state_with_parent() |
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1308 queue.sync_state_with_parent() |
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1309 |
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1310 def _remove_desktop_source(self, desktop_src: Gst.Element) -> None: |
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1311 """Remove the desktop source from the pipeline. |
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1312 |
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1313 @param desktop_src: The desktop source to remove. |
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1314 """ |
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1315 # Remove elements for the desktop source |
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1316 video_convert = self.pipeline.get_by_name("desktop_videoconvert") |
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1317 queue = self.pipeline.get_by_name("desktop_queue") |
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1318 |
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1319 if video_convert: |
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1320 video_convert.set_state(Gst.State.NULL) |
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1321 desktop_src.unlink(video_convert) |
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1322 self.pipeline.remove(video_convert) |
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1323 |
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1324 if queue: |
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1325 queue.set_state(Gst.State.NULL) |
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1326 self.pipeline.remove(queue) |
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1327 |
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1328 desktop_src.set_state(Gst.State.NULL) |
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1329 self.pipeline.remove(desktop_src) |
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1330 |
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1331 # Release the pad associated with the desktop source |
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1332 if self.desktop_sink_pad: |
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1333 self.video_selector.release_request_pad(self.desktop_sink_pad) |
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1334 self.desktop_sink_pad = None |
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1335 |
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1336 if self.desktop_portal is not None: |
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1337 self.desktop_portal.end_session() |
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1338 |
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1339 async def end_call(self) -> None: |
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1340 """Stop streaming and clean instance""" |
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1341 self.reset_instance() |
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1342 |
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1343 |
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1344 class WebRTCCall: |
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1345 """Helper class to create and handle WebRTC. |
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1346 |
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1347 This class handles signals and communication of connection data with backend. |
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1348 |
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1349 """ |
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1350 |
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1351 def __init__( |
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1352 self, |
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1353 bridge, |
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1354 profile: str, |
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1355 callee: jid.JID, |
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1356 on_call_setup_cb: Callable | None = None, |
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1357 on_call_ended_cb: Callable | None = None, |
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1358 **kwargs, |
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1359 ): |
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1360 """Create and setup a webRTC instance |
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1361 |
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1362 @param bridge: async Bridge. |
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1363 @param profile: profile making or receiving the call |
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1364 @param callee: peer jid |
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1365 @param kwargs: extra kw args to use when instantiating WebRTC |
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1366 """ |
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1367 self.profile = profile |
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1368 self.webrtc = WebRTC(bridge, profile, **kwargs) |
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1369 self.webrtc.callee = callee |
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1370 self.on_call_setup_cb = on_call_setup_cb |
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1371 self.on_call_ended_cb = on_call_ended_cb |
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1372 bridge.register_signal("ice_candidates_new", self.on_ice_candidates_new, "plugin") |
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1373 bridge.register_signal("call_setup", self.on_call_setup, "plugin") |
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1374 bridge.register_signal("call_ended", self.on_call_ended, "plugin") |
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1375 |
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1376 @classmethod |
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1377 async def make_webrtc_call( |
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1378 cls, bridge, profile: str, call_data: CallData, **kwargs |
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1379 ) -> "WebRTCCall": |
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1380 """Create the webrtc_call instance |
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1381 |
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1382 @param call_data: Call data of the command |
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1383 @param kwargs: extra args used to instanciate WebRTCCall |
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1384 |
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1385 """ |
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1386 webrtc_call = cls(bridge, profile, call_data.callee, **call_data.kwargs, **kwargs) |
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1387 if call_data.sid is None: |
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1388 # we are making the call |
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1389 await webrtc_call.start() |
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1390 else: |
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1391 # we are receiving the call |
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1392 webrtc_call.sid = call_data.sid |
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1393 if call_data.action_id is not None: |
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1394 await bridge.action_launch( |
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1395 call_data.action_id, |
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1396 data_format.serialise({"cancelled": False}), |
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1397 profile, |
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1398 ) |
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1399 return webrtc_call |
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1400 |
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1401 @property |
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1402 def sid(self) -> str | None: |
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1403 return self.webrtc.sid |
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1404 |
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1405 @sid.setter |
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1406 def sid(self, new_sid: str | None) -> None: |
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1407 self.webrtc.sid = new_sid |
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1408 |
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1409 async def on_ice_candidates_new( |
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1410 self, sid: str, candidates_s: str, profile: str |
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1411 ) -> None: |
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1412 if sid != self.webrtc.sid or profile != self.profile: |
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1413 return |
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1414 self.webrtc.on_ice_candidates_new( |
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1415 data_format.deserialise(candidates_s), |
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1416 ) |
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1417 |
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1418 async def on_call_setup(self, sid: str, setup_data_s: str, profile: str) -> None: |
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1419 if sid != self.webrtc.sid or profile != self.profile: |
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1420 return |
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1421 setup_data = data_format.deserialise(setup_data_s) |
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1422 try: |
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1423 role = setup_data["role"] |
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1424 sdp = setup_data["sdp"] |
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1425 except KeyError: |
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1426 log.error(f"Invalid setup data received: {setup_data}") |
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1427 return |
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1428 if role == "initiator": |
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1429 self.webrtc.on_accepted_call(sdp, profile) |
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1430 elif role == "responder": |
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1431 await self.webrtc.answer_call(sdp, profile) |
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1432 else: |
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1433 log.error(f"Invalid role received during setup: {setup_data}") |
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1434 if self.on_call_setup_cb is not None: |
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1435 await aio.maybe_async(self.on_call_setup_cb(sid, profile)) |
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1436 |
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1437 async def on_call_ended(self, sid: str, data_s: str, profile: str) -> None: |
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1438 if sid != self.webrtc.sid or profile != self.profile: |
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1439 return |
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1440 await self.webrtc.end_call() |
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1441 if self.on_call_ended_cb is not None: |
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1442 await aio.maybe_async(self.on_call_ended_cb(sid, profile)) |
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1443 |
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1444 async def start(self): |
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1445 """Start a call. |
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1446 |
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1447 To be used only if we are initiator |
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1448 """ |
4240
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1449 try: |
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1450 await self.webrtc.setup_call("initiator") |
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1451 except Exception: |
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|
1452 log.exception("Can't setup call") |
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|
1453 raise |
4233
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|
1454 self.webrtc.start_pipeline() |