Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc.py @ 4204:879bad48cc2d
frontends (tools/webrtc): implement X11 desktop sharing:
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author | Goffi <goffi@goffi.org> |
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date | Tue, 16 Jan 2024 10:42:00 +0100 |
parents | 60d107f2178a |
children | 17a8168966f9 |
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1 #!/usr/bin/env python3 |
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2 |
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3 # Libervia WebRTC implementation |
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4 # Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org) |
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5 |
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6 # This program is free software: you can redistribute it and/or modify |
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7 # it under the terms of the GNU Affero General Public License as published by |
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8 # the Free Software Foundation, either version 3 of the License, or |
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9 # (at your option) any later version. |
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10 |
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11 # This program is distributed in the hope that it will be useful, |
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12 # but WITHOUT ANY WARRANTY; without even the implied warranty of |
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13 # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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14 # GNU Affero General Public License for more details. |
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15 |
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16 # You should have received a copy of the GNU Affero General Public License |
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17 # along with this program. If not, see <http://www.gnu.org/licenses/>. |
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18 |
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19 import gi |
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20 gi.require_versions({ |
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21 "Gst": "1.0", |
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22 "GstWebRTC": "1.0" |
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23 }) |
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24 from gi.repository import Gst, GstWebRTC, GstSdp |
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25 |
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26 from libervia.backend.core import exceptions |
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27 |
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28 try: |
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29 from gi.overrides import Gst as _ |
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30 except ImportError: |
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31 raise exceptions.MissingModule( |
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32 "No GStreamer Python overrides available. Please install relevant packages on " |
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33 "your system (e.g., `python3-gst-1.0` on Debian and derivatives)." |
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34 ) |
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35 import asyncio |
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36 from dataclasses import dataclass |
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37 from datetime import datetime |
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38 import logging |
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39 import re |
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40 from typing import Callable |
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41 from urllib.parse import quote_plus |
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42 |
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43 from libervia.backend.tools.common import data_format |
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44 from libervia.frontends.tools import aio, display_servers |
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45 |
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46 current_server = display_servers.detect() |
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47 if current_server == display_servers.X11: |
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48 # GSTreamer's ximagesrc documentation asks to run this function |
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49 import ctypes |
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50 ctypes.CDLL('libX11.so.6').XInitThreads() |
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51 |
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52 |
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53 log = logging.getLogger(__name__) |
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54 |
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55 Gst.init(None) |
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56 |
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57 SOURCES_AUTO = "auto" |
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58 SOURCES_TEST = "test" |
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59 SINKS_APP = "app" |
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60 SINKS_AUTO = "auto" |
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61 SINKS_TEST = "test" |
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62 |
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63 |
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64 @dataclass |
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65 class AppSinkData: |
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66 local_video_cb: Callable |
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67 remote_video_cb: Callable |
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68 |
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69 |
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70 class WebRTC: |
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71 """GSTreamer based WebRTC implementation for audio and video communication. |
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72 |
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73 This class encapsulates the WebRTC functionalities required for initiating and |
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74 handling audio and video calls. |
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75 """ |
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76 |
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77 def __init__( |
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78 self, |
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79 bridge, |
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80 profile: str, |
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81 sources: str = SOURCES_AUTO, |
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82 sinks: str = SINKS_AUTO, |
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83 appsink_data: AppSinkData | None = None, |
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84 reset_cb: Callable | None = None, |
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85 ) -> None: |
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86 self.main_loop = asyncio.get_event_loop() |
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87 self.bridge = bridge |
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88 self.profile = profile |
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89 self.pipeline = None |
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90 self._audio_muted = False |
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91 self._video_muted = False |
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92 self._desktop_sharing = False |
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93 self.desktop_sharing_data = None |
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94 self.sources = sources |
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95 self.sinks = sinks |
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96 if sinks == SINKS_APP: |
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97 self.appsink_data = appsink_data |
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98 elif appsink_data is not None: |
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99 raise exceptions.InternalError( |
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100 "appsink_data can only be used for SINKS_APP sinks" |
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101 ) |
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102 self.reset_cb = reset_cb |
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103 self.reset_instance() |
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104 |
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105 @property |
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106 def audio_muted(self): |
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107 return self._audio_muted |
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108 |
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109 @audio_muted.setter |
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110 def audio_muted(self, muted: bool) -> None: |
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111 if muted != self._audio_muted: |
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112 self._audio_muted = muted |
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113 self.on_audio_mute(muted) |
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114 |
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115 @property |
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116 def video_muted(self): |
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117 return self._video_muted |
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118 |
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119 @video_muted.setter |
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120 def video_muted(self, muted: bool) -> None: |
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121 if muted != self._video_muted: |
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122 self._video_muted = muted |
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123 self.on_video_mute(muted) |
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124 |
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125 @property |
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126 def desktop_sharing(self): |
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127 return self._desktop_sharing |
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128 |
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129 @desktop_sharing.setter |
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130 def desktop_sharing(self, active: bool) -> None: |
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131 if active != self._desktop_sharing: |
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132 self._desktop_sharing = active |
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133 self.on_desktop_switch(active) |
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134 |
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135 @property |
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136 def sdp_set(self): |
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137 return self._sdp_set |
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138 |
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139 @sdp_set.setter |
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140 def sdp_set(self, is_set: bool): |
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141 self._sdp_set = is_set |
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142 if is_set: |
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143 self.on_ice_candidates_new(self.remote_candidates_buffer) |
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144 for data in self.remote_candidates_buffer.values(): |
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145 data["candidates"].clear() |
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146 |
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147 @property |
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148 def media_types(self): |
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149 if self._media_types is None: |
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150 raise Exception("self._media_types should not be None!") |
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151 return self._media_types |
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152 |
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153 @media_types.setter |
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154 def media_types(self, new_media_types: dict) -> None: |
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155 self._media_types = new_media_types |
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156 self._media_types_inv = {v: k for k, v in new_media_types.items()} |
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157 |
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158 @property |
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159 def media_types_inv(self) -> dict: |
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160 if self._media_types_inv is None: |
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161 raise Exception("self._media_types_inv should not be None!") |
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162 return self._media_types_inv |
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163 |
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164 def generate_dot_file( |
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165 self, |
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166 filename: str = "pipeline", |
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167 details: Gst.DebugGraphDetails = Gst.DebugGraphDetails.ALL, |
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168 with_timestamp: bool = True, |
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169 bin_: Gst.Bin|None = None, |
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170 ) -> None: |
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171 """Generate Dot File for debugging |
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172 |
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173 ``GST_DEBUG_DUMP_DOT_DIR`` environment variable must be set to destination dir. |
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174 ``dot -Tpng -o <filename>.png <filename>.dot`` can be use to convert to a PNG file. |
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175 See |
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176 https://gstreamer.freedesktop.org/documentation/gstreamer/debugutils.html?gi-language=python#GstDebugGraphDetails |
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177 for details. |
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178 |
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179 @param filename: name of the generated file |
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180 @param details: which details to print |
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181 @param with_timestamp: if True, add a timestamp to filename |
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182 @param bin_: which bin to output. By default, the whole pipeline |
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183 (``self.pipeline``) will be used. |
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184 """ |
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185 if bin_ is None: |
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186 bin_ = self.pipeline |
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187 if with_timestamp: |
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188 timestamp = datetime.now().isoformat(timespec='milliseconds') |
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189 filename = f"{timestamp}_filename" |
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190 |
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191 Gst.debug_bin_to_dot_file(bin_, details, filename) |
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192 |
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193 def get_sdp_mline_index(self, media_type: str) -> int: |
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194 """Gets the sdpMLineIndex for a given media type. |
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195 |
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196 @param media_type: The type of the media. |
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197 """ |
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198 for index, m_type in self.media_types.items(): |
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199 if m_type == media_type: |
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200 return index |
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201 raise ValueError(f"Media type '{media_type}' not found") |
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202 |
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203 def _set_media_types(self, offer_sdp: str) -> None: |
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204 """Sets media types from offer SDP |
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205 |
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206 @param offer: RTC session description containing the offer |
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207 """ |
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208 sdp_lines = offer_sdp.splitlines() |
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209 media_types = {} |
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210 mline_index = 0 |
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211 |
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212 for line in sdp_lines: |
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213 if line.startswith("m="): |
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214 media_types[mline_index] = line[2 : line.find(" ")] |
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215 mline_index += 1 |
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216 |
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217 self.media_types = media_types |
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218 |
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219 def _a_call(self, method, *args, **kwargs): |
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220 """Call an async method in main thread""" |
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221 aio.run_from_thread(method, *args, **kwargs, loop=self.main_loop) |
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222 |
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223 def get_payload_types( |
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224 self, sdpmsg, video_encoding: str, audio_encoding: str |
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225 ) -> dict[str, int | None]: |
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226 """Find the payload types for the specified video and audio encoding. |
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227 |
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228 Very simplistically finds the first payload type matching the encoding |
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229 name. More complex applications will want to match caps on |
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230 profile-level-id, packetization-mode, etc. |
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231 """ |
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232 # method coming from gstreamer example (Matthew Waters, Nirbheek Chauhan) at |
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233 # subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py |
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234 video_pt = None |
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235 audio_pt = None |
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236 for i in range(0, sdpmsg.medias_len()): |
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237 media = sdpmsg.get_media(i) |
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238 for j in range(0, media.formats_len()): |
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239 fmt = media.get_format(j) |
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240 if fmt == "webrtc-datachannel": |
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241 continue |
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242 pt = int(fmt) |
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243 caps = media.get_caps_from_media(pt) |
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244 s = caps.get_structure(0) |
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245 encoding_name = s["encoding-name"] |
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246 if video_pt is None and encoding_name == video_encoding: |
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247 video_pt = pt |
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248 elif audio_pt is None and encoding_name == audio_encoding: |
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249 audio_pt = pt |
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250 return {video_encoding: video_pt, audio_encoding: audio_pt} |
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251 |
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252 def parse_ice_candidate(self, candidate_string): |
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253 """Parses the ice candidate string. |
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254 |
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255 @param candidate_string: The ice candidate string to be parsed. |
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256 """ |
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257 pattern = re.compile( |
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258 r"candidate:(?P<foundation>\S+) (?P<component_id>\d+) (?P<transport>\S+) " |
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259 r"(?P<priority>\d+) (?P<address>\S+) (?P<port>\d+) typ " |
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260 r"(?P<type>\S+)(?: raddr (?P<rel_addr>\S+) rport " |
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261 r"(?P<rel_port>\d+))?(?: generation (?P<generation>\d+))?" |
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262 ) |
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263 match = pattern.match(candidate_string) |
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264 if match: |
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265 candidate_dict = match.groupdict() |
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266 |
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267 # Apply the correct types to the dictionary values |
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268 candidate_dict["component_id"] = int(candidate_dict["component_id"]) |
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269 candidate_dict["priority"] = int(candidate_dict["priority"]) |
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270 candidate_dict["port"] = int(candidate_dict["port"]) |
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271 |
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272 if candidate_dict["rel_port"]: |
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273 candidate_dict["rel_port"] = int(candidate_dict["rel_port"]) |
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274 |
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275 if candidate_dict["generation"]: |
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276 candidate_dict["generation"] = candidate_dict["generation"] |
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277 |
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278 # Remove None values |
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279 return {k: v for k, v in candidate_dict.items() if v is not None} |
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280 else: |
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281 log.warning(f"can't parse candidate: {candidate_string!r}") |
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282 return None |
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283 |
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284 def build_ice_candidate(self, parsed_candidate): |
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285 """Builds ICE candidate |
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286 |
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287 @param parsed_candidate: Dictionary containing parsed ICE candidate |
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288 """ |
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289 base_format = ( |
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290 "candidate:{foundation} {component_id} {transport} {priority} " |
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291 "{address} {port} typ {type}" |
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292 ) |
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293 |
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294 if parsed_candidate.get("rel_addr") and parsed_candidate.get("rel_port"): |
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295 base_format += " raddr {rel_addr} rport {rel_port}" |
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296 |
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297 if parsed_candidate.get("generation"): |
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298 base_format += " generation {generation}" |
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299 |
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300 return base_format.format(**parsed_candidate) |
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301 |
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302 def extract_ufrag_pwd(self, sdp: str) -> tuple[str, str]: |
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303 """Retrieves ICE password and user fragment for SDP offer. |
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304 |
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305 @param sdp: The Session Description Protocol offer string. |
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306 @return: ufrag and pwd |
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307 @raise ValueError: Can't extract ufrag and password |
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308 """ |
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309 ufrag_line = re.search(r"ice-ufrag:(\S+)", sdp) |
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310 pwd_line = re.search(r"ice-pwd:(\S+)", sdp) |
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311 |
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312 if ufrag_line and pwd_line: |
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313 ufrag = self.ufrag = ufrag_line.group(1) |
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314 pwd = self.pwd = pwd_line.group(1) |
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315 return ufrag, pwd |
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316 else: |
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317 log.error(f"SDP with missing ice-ufrag or ice-pwd:\n{sdp}") |
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318 raise ValueError("Can't extract ice-ufrag and ice-pwd from SDP") |
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319 |
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320 def reset_instance(self): |
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321 """Inits or resets the instance variables to their default state.""" |
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322 self.role: str | None = None |
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323 if self.pipeline is not None: |
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324 self.pipeline.set_state(Gst.State.NULL) |
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325 self.pipeline = None |
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326 self._remote_video_pad = None |
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327 self.sid: str | None = None |
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328 self.offer: str | None = None |
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329 self.local_candidates_buffer = {} |
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330 self.ufrag: str | None = None |
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331 self.pwd: str | None = None |
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332 self.callee: str | None = None |
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333 self._media_types = None |
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334 self._media_types_inv = None |
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335 self._sdp_set: bool = False |
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336 self.remote_candidates_buffer: dict[str, dict[str, list]] = { |
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337 "audio": {"candidates": []}, |
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338 "video": {"candidates": []}, |
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339 } |
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340 self._media_types = None |
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341 self._media_types_inv = None |
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342 self.audio_valve = None |
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343 self.video_valve = None |
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344 if self.reset_cb is not None: |
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345 self.reset_cb() |
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346 |
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347 |
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348 async def setup_call( |
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349 self, |
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350 role: str, |
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351 audio_pt: int | None = 96, |
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352 video_pt: int | None = 97, |
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353 ) -> None: |
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354 """Sets up the call. |
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355 |
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356 This method establishes the Gstreamer pipeline for audio and video communication. |
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357 The method also manages STUN and TURN server configurations, signal watchers, and |
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358 various connection handlers for the webrtcbin. |
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359 |
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360 @param role: The role for the call, either 'initiator' or 'responder'. |
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361 @param audio_pt: The payload type for the audio stream. |
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362 @param video_pt: The payload type for the video stream |
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363 |
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364 @raises NotImplementedError: If audio_pt or video_pt is set to None. |
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365 @raises AssertionError: If the role is not 'initiator' or 'responder'. |
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366 """ |
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367 assert role in ("initiator", "responder") |
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368 self.role = role |
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369 if audio_pt is None or video_pt is None: |
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370 raise NotImplementedError("None value is not handled yet") |
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371 |
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372 if self.sources == SOURCES_AUTO: |
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373 video_source_elt = "v4l2src" |
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374 audio_source_elt = "pulsesrc" |
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375 elif self.sources == SOURCES_TEST: |
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376 video_source_elt = "videotestsrc is-live=true pattern=ball" |
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377 audio_source_elt = "audiotestsrc" |
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378 else: |
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379 raise exceptions.InternalError(f'Unknown "sources" value: {self.sources!r}') |
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380 |
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381 extra_elt = "" |
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382 |
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383 if self.sinks == SINKS_APP: |
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384 local_video_sink_elt = ( |
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385 "appsink name=local_video_sink emit-signals=true drop=true max-buffers=1 " |
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386 "sync=True" |
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387 ) |
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388 elif self.sinks == SINKS_AUTO: |
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389 extra_elt = "compositor name=compositor ! autovideosink" |
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390 local_video_sink_elt = """compositor.sink_1""" |
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391 else: |
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392 raise exceptions.InternalError(f"Unknown sinks value {self.sinks!r}") |
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393 |
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394 self.gst_pipe_desc = f""" |
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395 webrtcbin latency=100 name=sendrecv bundle-policy=max-bundle |
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396 |
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397 input-selector name=video_selector |
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398 ! videorate |
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399 ! video/x-raw,framerate=30/1 |
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400 ! tee name=t |
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401 |
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402 {extra_elt} |
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403 |
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404 {video_source_elt} name=video_src ! queue leaky=downstream ! video_selector. |
4204
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405 videotestsrc name=muted_src is-live=true pattern=black ! queue leaky=downstream ! video_selector. |
4139
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406 |
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407 t. |
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408 ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream |
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409 ! videoconvert |
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410 ! vp8enc deadline=1 keyframe-max-dist=60 |
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411 ! rtpvp8pay picture-id-mode=15-bit |
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412 ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt} |
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413 ! sendrecv. |
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414 |
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415 t. |
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416 ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream |
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417 ! videoconvert |
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418 ! {local_video_sink_elt} |
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419 |
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420 {audio_source_elt} name=audio_src |
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421 ! valve |
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422 ! queue max-size-buffers=10 max-size-time=0 max-size-bytes=0 leaky=downstream |
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423 ! audioconvert |
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424 ! audioresample |
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425 ! opusenc audio-type=voice |
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426 ! rtpopuspay |
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427 ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} |
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428 ! sendrecv. |
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429 """ |
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430 |
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431 log.debug(f"Gstreamer pipeline: {self.gst_pipe_desc}") |
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432 |
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433 # Create the pipeline |
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434 self.pipeline = Gst.parse_launch(self.gst_pipe_desc) |
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435 if not self.pipeline: |
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436 raise exceptions.InternalError("Failed to create Gstreamer pipeline.") |
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437 |
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438 self.webrtcbin = self.pipeline.get_by_name("sendrecv") |
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439 self.video_src = self.pipeline.get_by_name("video_src") |
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440 self.muted_src = self.pipeline.get_by_name("muted_src") |
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441 self.video_selector = self.pipeline.get_by_name("video_selector") |
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442 self.audio_valve = self.pipeline.get_by_name("audio_valve") |
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443 |
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444 if self.video_muted: |
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445 self.on_video_mute(True) |
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446 if self.audio_muted: |
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447 self.on_audio_mute(True) |
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448 |
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449 # set STUN and TURN servers |
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450 external_disco = data_format.deserialise( |
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451 await self.bridge.external_disco_get("", self.profile), type_check=list |
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452 ) |
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453 |
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454 for server in external_disco: |
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455 if server["type"] == "stun": |
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456 if server["transport"] == "tcp": |
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457 log.info( |
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458 "ignoring TCP STUN server, GStreamer only support one STUN server" |
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459 ) |
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460 url = f"stun://{server['host']}:{server['port']}" |
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461 log.debug(f"adding stun server: {url}") |
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462 self.webrtcbin.set_property("stun-server", url) |
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463 elif server["type"] == "turn": |
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464 url = "{scheme}://{username}:{password}@{host}:{port}".format( |
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465 scheme="turns" if server["transport"] == "tcp" else "turn", |
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466 username=quote_plus(server["username"]), |
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467 password=quote_plus(server["password"]), |
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468 host=server["host"], |
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469 port=server["port"], |
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470 ) |
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471 log.debug(f"adding turn server: {url}") |
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472 |
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473 if not self.webrtcbin.emit("add-turn-server", url): |
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474 log.warning(f"Erreur while adding TURN server {url}") |
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475 |
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476 # local video feedback |
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477 if self.sinks == SINKS_APP: |
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478 assert self.appsink_data is not None |
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479 local_video_sink = self.pipeline.get_by_name("local_video_sink") |
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480 local_video_sink.set_property("emit-signals", True) |
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481 local_video_sink.connect("new-sample", self.appsink_data.local_video_cb) |
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482 local_video_sink_caps = Gst.Caps.from_string(f"video/x-raw,format=RGB") |
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483 local_video_sink.set_property("caps", local_video_sink_caps) |
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484 |
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485 # Create bus and associate signal watchers |
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486 self.bus = self.pipeline.get_bus() |
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487 if not self.bus: |
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488 log.error("Failed to get bus from pipeline.") |
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489 return |
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490 |
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491 self.bus.add_signal_watch() |
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492 self.webrtcbin.connect("pad-added", self.on_pad_added) |
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493 self.bus.connect("message::error", self.on_bus_error) |
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494 self.bus.connect("message::eos", self.on_bus_eos) |
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495 self.webrtcbin.connect("on-negotiation-needed", self.on_negotiation_needed) |
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496 self.webrtcbin.connect("on-ice-candidate", self.on_ice_candidate) |
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497 self.webrtcbin.connect( |
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498 "notify::ice-gathering-state", self.on_ice_gathering_state_change |
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499 ) |
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500 self.webrtcbin.connect( |
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501 "notify::ice-connection-state", self.on_ice_connection_state |
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502 ) |
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503 |
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504 def start_pipeline(self) -> None: |
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505 """Starts the GStreamer pipeline.""" |
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506 log.debug("starting the pipeline") |
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507 self.pipeline.set_state(Gst.State.PLAYING) |
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508 |
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509 def on_negotiation_needed(self, webrtc): |
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510 """Initiate SDP offer when negotiation is needed.""" |
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511 log.debug("Negotiation needed.") |
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512 if self.role == "initiator": |
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513 log.debug("Creating offer…") |
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514 promise = Gst.Promise.new_with_change_func(self.on_offer_created) |
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515 self.webrtcbin.emit("create-offer", None, promise) |
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516 |
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517 def on_offer_created(self, promise): |
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518 """Callback for when SDP offer is created.""" |
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519 log.info("on_offer_created called") |
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520 assert promise.wait() == Gst.PromiseResult.REPLIED |
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521 reply = promise.get_reply() |
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522 if reply is None: |
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523 log.error("Promise reply is None. Offer creation might have failed.") |
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524 return |
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525 offer = reply["offer"] |
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526 self.offer = offer.sdp.as_text() |
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527 log.info(f"SDP offer created: \n{self.offer}") |
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528 self._set_media_types(self.offer) |
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529 promise = Gst.Promise.new() |
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530 self.webrtcbin.emit("set-local-description", offer, promise) |
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531 promise.interrupt() |
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532 self._a_call(self._start_call) |
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533 |
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534 def on_answer_set(self, promise): |
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535 assert promise.wait() == Gst.PromiseResult.REPLIED |
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536 |
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537 def on_answer_created(self, promise, _, __): |
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538 """Callback for when SDP answer is created.""" |
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539 assert promise.wait() == Gst.PromiseResult.REPLIED |
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540 reply = promise.get_reply() |
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541 answer = reply["answer"] |
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542 promise = Gst.Promise.new() |
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543 self.webrtcbin.emit("set-local-description", answer, promise) |
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544 promise.interrupt() |
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545 answer_sdp = answer.sdp.as_text() |
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546 log.info(f"SDP answer set: \n{answer_sdp}") |
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547 self.sdp_set = True |
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548 self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile) |
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549 |
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550 def on_offer_set(self, promise): |
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551 assert promise.wait() == Gst.PromiseResult.REPLIED |
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552 promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None) |
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553 self.webrtcbin.emit("create-answer", None, promise) |
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554 |
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555 def link_element_or_pad( |
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556 self, source: Gst.Element, dest: Gst.Element | Gst.Pad |
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557 ) -> bool: |
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558 """Check if dest is a pad or an element, and link appropriately""" |
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559 src_pad = source.get_static_pad("src") |
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560 |
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561 if isinstance(dest, Gst.Pad): |
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562 # If the dest is a pad, link directly |
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563 if not src_pad.link(dest) == Gst.PadLinkReturn.OK: |
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|
564 log.error( |
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|
565 "Failed to link 'conv' to the compositor's newly requested pad!" |
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|
566 ) |
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567 return False |
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568 elif isinstance(dest, Gst.Element): |
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|
569 return source.link(dest) |
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|
570 else: |
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571 log.error(f"Unexpected type for dest: {type(sink)}") |
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572 return False |
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|
573 |
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|
574 return True |
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575 |
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|
576 def scaled_dimensions( |
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577 self, original_width: int, original_height: int, max_width: int, max_height: int |
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578 ) -> tuple[int, int]: |
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579 """Calculates the scaled dimensions preserving aspect ratio. |
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580 |
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581 @param original_width: Original width of the video stream. |
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582 @param original_height: Original height of the video stream. |
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583 @param max_width: Maximum desired width for the scaled video. |
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584 @param max_height: Maximum desired height for the scaled video. |
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585 @return: The width and height of the scaled video. |
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586 """ |
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587 aspect_ratio = original_width / original_height |
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588 new_width = int(max_height * aspect_ratio) |
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589 |
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590 if new_width <= max_width: |
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591 return new_width, max_height |
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592 |
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593 new_height = int(max_width / aspect_ratio) |
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594 return max_width, new_height |
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595 |
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596 def on_remote_decodebin_stream(self, _, pad: Gst.Pad) -> None: |
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597 """Handle the stream from the remote decodebin. |
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598 |
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599 This method processes the incoming stream from the remote decodebin, determining |
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600 whether it's video or audio. It then sets up the appropriate GStreamer elements |
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|
601 for video/audio processing and adds them to the pipeline. |
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|
602 |
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|
603 @param pad: The Gst.Pad from the remote decodebin producing the stream. |
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|
604 """ |
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|
605 assert self.pipeline is not None |
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|
606 if not pad.has_current_caps(): |
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|
607 log.error(f"{pad} has no caps, ignoring") |
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|
608 return |
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|
609 |
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610 caps = pad.get_current_caps() |
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611 assert len(caps) |
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612 s = caps[0] |
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613 name = s.get_name() |
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614 log.debug(f"====> NAME START: {name}") |
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615 |
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616 q = Gst.ElementFactory.make("queue") |
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617 |
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618 if name.startswith("video"): |
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619 log.debug("===> VIDEO OK") |
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620 |
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621 self._remote_video_pad = pad |
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622 |
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623 # Check and log the original size of the video |
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624 width = s.get_int("width").value |
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625 height = s.get_int("height").value |
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626 log.info(f"Original video size: {width}x{height}") |
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627 |
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628 # This is a fix for an issue found with Movim on desktop: a non standard |
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629 # resolution is used (990x557) resulting in bad alignement and no color in |
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630 # rendered image |
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631 adjust_resolution = width % 4 != 0 or height % 4 != 0 |
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632 if adjust_resolution: |
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633 log.info("non standard resolution, we need to adjust size") |
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634 width = (width + 3) // 4 * 4 |
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635 height = (height + 3) // 4 * 4 |
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636 log.info(f"Adjusted video size: {width}x{height}") |
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637 |
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638 conv = Gst.ElementFactory.make("videoconvert") |
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639 if self.sinks == SINKS_APP: |
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640 assert self.appsink_data is not None |
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641 remote_video_sink = Gst.ElementFactory.make("appsink") |
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642 |
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643 appsink_caps = Gst.Caps.from_string("video/x-raw,format=RGB") |
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644 remote_video_sink.set_property("caps", appsink_caps) |
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645 |
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646 remote_video_sink.set_property("emit-signals", True) |
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647 remote_video_sink.set_property("drop", True) |
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648 remote_video_sink.set_property("max-buffers", 1) |
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649 remote_video_sink.set_property("sync", True) |
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650 remote_video_sink.connect("new-sample", self.appsink_data.remote_video_cb) |
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651 self.pipeline.add(remote_video_sink) |
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652 elif self.sinks == SINKS_AUTO: |
4139
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653 compositor = self.pipeline.get_by_name("compositor") |
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654 |
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655 sink1_pad = compositor.get_static_pad("sink_1") |
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656 |
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657 local_width, local_height = self.scaled_dimensions( |
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658 sink1_pad.get_property("width"), |
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659 sink1_pad.get_property("height"), |
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660 width // 3, |
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661 height // 3, |
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662 ) |
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663 |
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664 sink1_pad.set_property("xpos", width - local_width) |
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665 sink1_pad.set_property("ypos", height - local_height) |
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666 sink1_pad.set_property("width", local_width) |
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667 sink1_pad.set_property("height", local_height) |
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668 sink1_pad.set_property("zorder", 1) |
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669 |
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670 # Request a new pad for the remote stream |
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671 sink_pad_template = compositor.get_pad_template("sink_%u") |
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672 remote_video_sink = compositor.request_pad(sink_pad_template, None, None) |
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673 remote_video_sink.set_property("zorder", 0) |
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674 |
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675 else: |
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676 raise exceptions.InternalError(f'Unhandled "sinks" value: {self.sinks!r}') |
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677 |
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678 if adjust_resolution: |
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679 videoscale = Gst.ElementFactory.make("videoscale") |
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680 adjusted_caps = Gst.Caps.from_string( |
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681 f"video/x-raw,width={width},height={height}" |
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682 ) |
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683 capsfilter = Gst.ElementFactory.make("capsfilter") |
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684 capsfilter.set_property("caps", adjusted_caps) |
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685 |
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686 self.pipeline.add(q, conv, videoscale, capsfilter) |
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687 |
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688 |
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689 self.pipeline.sync_children_states() |
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690 ret = pad.link(q.get_static_pad("sink")) |
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691 if ret != Gst.PadLinkReturn.OK: |
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692 log.error(f"Error linking pad: {ret}") |
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693 q.link(conv) |
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694 conv.link(videoscale) |
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695 videoscale.link(capsfilter) |
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696 self.link_element_or_pad(capsfilter.link, remote_video_sink) |
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697 |
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698 else: |
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699 self.pipeline.add(q, conv) |
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700 |
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701 self.pipeline.sync_children_states() |
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702 ret = pad.link(q.get_static_pad("sink")) |
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703 if ret != Gst.PadLinkReturn.OK: |
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704 log.error(f"Error linking pad: {ret}") |
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705 q.link(conv) |
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706 self.link_element_or_pad(conv, remote_video_sink) |
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707 |
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708 elif name.startswith("audio"): |
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709 log.debug("===> Audio OK") |
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710 conv = Gst.ElementFactory.make("audioconvert") |
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711 resample = Gst.ElementFactory.make("audioresample") |
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712 remote_audio_sink = Gst.ElementFactory.make("autoaudiosink") |
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713 self.pipeline.add(q, conv, resample, remote_audio_sink) |
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714 self.pipeline.sync_children_states() |
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715 ret = pad.link(q.get_static_pad("sink")) |
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716 if ret != Gst.PadLinkReturn.OK: |
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717 log.error(f"Error linking pad: {ret}") |
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718 q.link(conv) |
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719 conv.link(resample) |
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720 resample.link(remote_audio_sink) |
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721 |
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722 else: |
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723 log.warning(f"unmanaged name: {name!r}") |
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724 |
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725 def on_pad_added(self, __, pad: Gst.Pad) -> None: |
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726 """Handle the addition of a new pad to the element. |
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727 |
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728 When a new source pad is added to the element, this method creates a decodebin, |
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729 connects it to handle the stream, and links the pad to the decodebin. |
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730 |
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731 @param __: Placeholder for the signal source. Not used in this method. |
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732 @param pad: The newly added pad. |
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733 """ |
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|
734 log.debug("on_pad_added") |
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735 if pad.direction != Gst.PadDirection.SRC: |
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|
736 return |
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737 |
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738 decodebin = Gst.ElementFactory.make("decodebin") |
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739 decodebin.connect("pad-added", self.on_remote_decodebin_stream) |
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740 self.pipeline.add(decodebin) |
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741 decodebin.sync_state_with_parent() |
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742 pad.link(decodebin.get_static_pad("sink")) |
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743 |
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744 async def _start_call(self) -> None: |
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745 """Initiate the call. |
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746 |
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747 Initiates a call with the callee using the stored offer. If there are any buffered |
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748 local ICE candidates, they are sent as part of the initiation. |
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749 """ |
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750 assert self.callee |
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751 self.sid = await self.bridge.call_start( |
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752 str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile |
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753 ) |
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754 if self.local_candidates_buffer: |
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755 log.debug( |
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756 f"sending buffered local ICE candidates: {self.local_candidates_buffer}" |
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757 ) |
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758 if self.pwd is None: |
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759 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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760 self.extract_ufrag_pwd(sdp) |
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761 ice_data = {} |
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762 for media_type, candidates in self.local_candidates_buffer.items(): |
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763 ice_data[media_type] = { |
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764 "ufrag": self.ufrag, |
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765 "pwd": self.pwd, |
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766 "candidates": candidates, |
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767 } |
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768 await self.bridge.ice_candidates_add( |
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|
769 self.sid, data_format.serialise(ice_data), self.profile |
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|
770 ) |
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771 self.local_candidates_buffer.clear() |
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772 |
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773 def _remote_sdp_set(self, promise) -> None: |
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774 assert promise.wait() == Gst.PromiseResult.REPLIED |
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775 self.sdp_set = True |
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776 |
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777 def on_accepted_call(self, sdp: str, profile: str) -> None: |
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778 """Outgoing call has been accepted. |
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|
779 |
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780 @param sdp: The SDP answer string received from the other party. |
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781 @param profile: Profile used for the call. |
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782 """ |
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783 log.debug(f"SDP answer received: \n{sdp}") |
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784 |
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785 __, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp) |
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786 answer = GstWebRTC.WebRTCSessionDescription.new( |
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787 GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg |
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788 ) |
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789 promise = Gst.Promise.new_with_change_func(self._remote_sdp_set) |
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790 self.webrtcbin.emit("set-remote-description", answer, promise) |
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791 |
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792 async def answer_call(self, sdp: str, profile: str) -> None: |
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793 """Answer an incoming call |
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794 |
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795 @param sdp: The SDP offer string received from the initiator. |
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796 @param profile: Profile used for the call. |
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797 |
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798 @raise AssertionError: Raised when either "VP8" or "OPUS" is not present in |
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799 payload types. |
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800 """ |
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|
801 log.debug(f"SDP offer received: \n{sdp}") |
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802 self._set_media_types(sdp) |
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803 __, offer_sdp_msg = GstSdp.SDPMessage.new_from_text(sdp) |
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804 payload_types = self.get_payload_types( |
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805 offer_sdp_msg, video_encoding="VP8", audio_encoding="OPUS" |
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806 ) |
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807 assert "VP8" in payload_types |
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808 assert "OPUS" in payload_types |
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809 await self.setup_call( |
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810 "responder", audio_pt=payload_types["OPUS"], video_pt=payload_types["VP8"] |
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811 ) |
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812 self.start_pipeline() |
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813 offer = GstWebRTC.WebRTCSessionDescription.new( |
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814 GstWebRTC.WebRTCSDPType.OFFER, offer_sdp_msg |
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815 ) |
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816 promise = Gst.Promise.new_with_change_func(self.on_offer_set) |
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817 self.webrtcbin.emit("set-remote-description", offer, promise) |
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818 |
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819 def on_ice_candidate(self, webrtc, mline_index, candidate_sdp): |
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820 """Handles the on-ice-candidate signal of webrtcbin. |
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821 |
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822 @param webrtc: The webrtcbin element. |
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823 @param mlineindex: The mline index. |
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824 @param candidate: The ICE candidate. |
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825 """ |
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826 log.debug( |
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827 f"Local ICE candidate. MLine Index: {mline_index}, Candidate: {candidate_sdp}" |
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828 ) |
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829 parsed_candidate = self.parse_ice_candidate(candidate_sdp) |
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830 try: |
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831 media_type = self.media_types[mline_index] |
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832 except KeyError: |
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833 raise exceptions.InternalError("can't find media type") |
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834 |
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835 if self.sid is None: |
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836 log.debug("buffering local ICE candidate") |
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837 self.local_candidates_buffer.setdefault(media_type, []).append( |
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838 parsed_candidate |
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839 ) |
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840 else: |
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841 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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842 assert sdp is not None |
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843 ufrag, pwd = self.extract_ufrag_pwd(sdp) |
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844 ice_data = {"ufrag": ufrag, "pwd": pwd, "candidates": [parsed_candidate]} |
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845 self._a_call( |
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846 self.bridge.ice_candidates_add, |
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847 self.sid, |
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848 data_format.serialise({media_type: ice_data}), |
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849 self.profile, |
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850 ) |
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851 |
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852 def on_ice_candidates_new(self, candidates: dict) -> None: |
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853 """Handle new ICE candidates. |
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854 |
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855 @param candidates: A dictionary containing media types ("audio" or "video") as |
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856 keys and corresponding ICE data as values. |
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857 |
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858 @raise exceptions.InternalError: Raised when sdp mline index is not found. |
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859 """ |
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860 if not self.sdp_set: |
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861 log.debug("buffering remote ICE candidate") |
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862 for media_type in ("audio", "video"): |
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863 media_candidates = candidates.get(media_type) |
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864 if media_candidates: |
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865 buffer = self.remote_candidates_buffer[media_type] |
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866 buffer["candidates"].extend(media_candidates["candidates"]) |
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867 return |
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868 for media_type, ice_data in candidates.items(): |
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869 for candidate in ice_data["candidates"]: |
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870 candidate_sdp = self.build_ice_candidate(candidate) |
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871 try: |
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872 mline_index = self.get_sdp_mline_index(media_type) |
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873 except Exception as e: |
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874 raise exceptions.InternalError(f"Can't find sdp mline index: {e}") |
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875 self.webrtcbin.emit("add-ice-candidate", mline_index, candidate_sdp) |
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876 log.debug( |
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877 f"Remote ICE candidate added. MLine Index: {mline_index}, " |
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878 f"{candidate_sdp}" |
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|
879 ) |
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880 |
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881 def on_ice_gathering_state_change(self, pspec, __): |
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882 state = self.webrtcbin.get_property("ice-gathering-state") |
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883 log.debug(f"ICE gathering state changed to {state}") |
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884 |
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885 def on_ice_connection_state(self, pspec, __): |
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886 state = self.webrtcbin.props.ice_connection_state |
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887 if state == GstWebRTC.WebRTCICEConnectionState.FAILED: |
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888 log.error("ICE connection failed") |
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889 log.info(f"ICE connection state changed to {state}") |
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890 |
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891 def on_bus_error(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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892 """Handles the GStreamer bus error messages. |
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893 |
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894 @param bus: The GStreamer bus. |
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895 @param message: The error message. |
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896 """ |
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897 err, debug = message.parse_error() |
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898 log.error(f"Error from {message.src.get_name()}: {err.message}") |
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899 log.error(f"Debugging info: {debug}") |
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900 |
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|
901 def on_bus_eos(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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902 """Handles the GStreamer bus eos messages. |
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|
903 |
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|
904 @param bus: The GStreamer bus. |
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905 @param message: The eos message. |
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|
906 """ |
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|
907 log.info("End of stream") |
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|
908 |
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|
909 def on_audio_mute(self, muted: bool) -> None: |
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910 """Handles (un)muting of audio. |
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|
911 |
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|
912 @param muted: True if audio is muted. |
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|
913 """ |
4139
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914 if self.audio_valve is not None: |
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|
915 self.audio_valve.set_property("drop", muted) |
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916 state = "muted" if muted else "unmuted" |
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917 log.info(f"audio is now {state}") |
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|
918 |
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|
919 def on_video_mute(self, muted: bool) -> None: |
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920 """Handles (un)muting of video. |
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|
921 |
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|
922 @param muted: True if video is muted. |
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|
923 """ |
4139
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|
924 if self.video_selector is not None: |
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|
925 current_source = None if muted else "desktop" if self.desktop_sharing else "camera" |
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|
926 self.switch_video_source(current_source) |
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|
927 state = "muted" if muted else "unmuted" |
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|
928 log.info(f"Video is now {state}") |
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|
929 |
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|
930 def on_desktop_switch(self, desktop_active: bool) -> None: |
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|
931 """Switches the video source between desktop and camera. |
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|
932 |
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|
933 @param desktop_active: True if desktop must be active. False for camera. |
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|
934 """ |
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|
935 if self.video_muted: |
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|
936 # Update the active source state but do not switch |
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|
937 self.desktop_sharing = desktop_active |
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|
938 return |
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|
939 |
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|
940 source = "desktop" if desktop_active else "camera" |
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|
941 self.switch_video_source(source) |
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|
942 self.desktop_sharing = desktop_active |
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|
943 |
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|
944 def switch_video_source(self, source: str|None) -> None: |
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|
945 """Activates the specified source while deactivating the others. |
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|
946 |
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|
947 @param source: 'desktop', 'camera', 'muted' or None for muted source. |
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|
948 """ |
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|
949 if source is None: |
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|
950 source = "muted" |
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|
951 if source not in ["camera", "muted", "desktop"]: |
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|
952 raise ValueError( |
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|
953 f"Invalid source: {source!r}, use one of {'camera', 'muted', 'desktop'}" |
879bad48cc2d
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|
954 ) |
879bad48cc2d
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|
955 |
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|
956 self.pipeline.set_state(Gst.State.PAUSED) |
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|
957 |
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|
958 # Create a new desktop source if necessary |
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|
959 if source == "desktop": |
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|
960 self._setup_desktop_source(self.desktop_sharing_data) |
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|
961 |
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|
962 # Activate the chosen source and deactivate the others |
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|
963 for src_name in ["camera", "muted", "desktop"]: |
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|
964 src_element = self.pipeline.get_by_name(f"{src_name}_src") |
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|
965 if src_name == source: |
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|
966 if src_element: |
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|
967 src_element.set_state(Gst.State.PLAYING) |
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968 else: |
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969 if src_element: |
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970 src_element.set_state(Gst.State.NULL) |
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971 if src_name == "desktop": |
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972 self._remove_desktop_source(src_element) |
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973 |
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974 # Set the video_selector active pad |
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975 pad_name = f"sink_{['camera', 'muted', 'desktop'].index(source)}" |
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976 pad = self.video_selector.get_static_pad(pad_name) |
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977 self.video_selector.props.active_pad = pad |
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978 self.pipeline.set_state(Gst.State.PLAYING) |
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979 |
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980 def _setup_desktop_source(self, properties: dict[str, object]|None) -> None: |
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981 """Set up a new desktop source. |
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982 |
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983 @param properties: The properties to set on the desktop source. |
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984 """ |
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985 desktop_src = Gst.ElementFactory.make("ximagesrc", "desktop_src") |
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986 if properties is None: |
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987 properties = {} |
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988 for key, value in properties.items(): |
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989 log.debug(f"setting ximagesrc property: {key!r}={value!r}") |
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990 desktop_src.set_property(key, value) |
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991 video_convert = Gst.ElementFactory.make("videoconvert", "desktop_videoconvert") |
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992 queue = Gst.ElementFactory.make("queue", "desktop_queue") |
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993 queue.set_property("leaky", "downstream") |
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994 |
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995 self.pipeline.add(desktop_src) |
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996 self.pipeline.add(video_convert) |
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997 self.pipeline.add(queue) |
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998 |
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999 desktop_src.link(video_convert) |
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1000 video_convert.link(queue) |
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1001 |
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1002 sink_pad_template = self.video_selector.get_pad_template("sink_%u") |
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1003 sink_pad = self.video_selector.request_pad(sink_pad_template, None, None) |
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1004 queue_src_pad = queue.get_static_pad("src") |
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1005 queue_src_pad.link(sink_pad) |
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1006 |
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1007 desktop_src.sync_state_with_parent() |
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1008 video_convert.sync_state_with_parent() |
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1009 queue.sync_state_with_parent() |
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1010 |
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1011 def _remove_desktop_source(self, desktop_src: Gst.Element) -> None: |
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1012 """Remove the desktop source from the pipeline. |
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1013 |
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1014 @param desktop_src: The desktop source to remove. |
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1015 """ |
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1016 # Remove elements for the desktop source |
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1017 video_convert = self.pipeline.get_by_name("desktop_videoconvert") |
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1018 queue = self.pipeline.get_by_name("desktop_queue") |
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1019 if video_convert: |
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1020 video_convert.set_state(Gst.State.NULL) |
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1021 desktop_src.unlink(video_convert) |
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1022 self.pipeline.remove(video_convert) |
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1023 if queue: |
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1024 queue.set_state(Gst.State.NULL) |
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1025 self.pipeline.remove(queue) |
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1026 self.pipeline.remove(desktop_src) |
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1027 |
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1028 async def end_call(self) -> None: |
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1029 """Stop streaming and clean instance""" |
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1030 self.reset_instance() |