Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc.py @ 4194:3dbaf179c50d
core (launcher): new `wait_for_service` option:
if `wait_for_service` is used in `[DEFAULT]`, the service at the specified host/port is
waited for before starting the backend. This is useful in the situation where we need to
be sure that something is started before running, e.g. in a container.
author | Goffi <goffi@goffi.org> |
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date | Wed, 13 Dec 2023 22:00:25 +0100 |
parents | 60d107f2178a |
children | 879bad48cc2d |
rev | line source |
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1 #!/usr/bin/env python3 |
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2 |
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3 # Libervia WebRTC implementation |
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4 # Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org) |
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5 |
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6 # This program is free software: you can redistribute it and/or modify |
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7 # it under the terms of the GNU Affero General Public License as published by |
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8 # the Free Software Foundation, either version 3 of the License, or |
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9 # (at your option) any later version. |
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10 |
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11 # This program is distributed in the hope that it will be useful, |
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12 # but WITHOUT ANY WARRANTY; without even the implied warranty of |
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13 # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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14 # GNU Affero General Public License for more details. |
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15 |
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16 # You should have received a copy of the GNU Affero General Public License |
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17 # along with this program. If not, see <http://www.gnu.org/licenses/>. |
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18 |
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19 import gi |
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20 gi.require_versions({ |
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21 "Gst": "1.0", |
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22 "GstWebRTC": "1.0" |
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23 }) |
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24 from gi.repository import Gst, GstWebRTC, GstSdp |
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25 |
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26 from libervia.backend.core import exceptions |
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27 |
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28 try: |
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29 from gi.overrides import Gst as _ |
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30 except ImportError: |
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31 raise exceptions.MissingModule( |
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32 "No GStreamer Python overrides available. Please install relevant packages on " |
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33 "your system (e.g., `python3-gst-1.0` on Debian and derivatives)." |
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34 ) |
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35 import asyncio |
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36 from dataclasses import dataclass |
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37 from datetime import datetime |
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38 import logging |
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39 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 |
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45 |
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46 |
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47 log = logging.getLogger(__name__) |
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48 |
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49 Gst.init(None) |
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50 |
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51 SOURCES_AUTO = "auto" |
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52 SOURCES_TEST = "test" |
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53 SINKS_APP = "app" |
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54 SINKS_AUTO = "auto" |
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55 SINKS_TEST = "test" |
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56 |
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57 |
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58 @dataclass |
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59 class AppSinkData: |
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60 local_video_cb: Callable |
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61 remote_video_cb: Callable |
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62 |
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63 |
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64 class WebRTC: |
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65 """GSTreamer based WebRTC implementation for audio and video communication. |
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66 |
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67 This class encapsulates the WebRTC functionalities required for initiating and |
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68 handling audio and video calls. |
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69 """ |
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70 |
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71 def __init__( |
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72 self, |
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73 bridge, |
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74 profile: str, |
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75 sources: str = SOURCES_AUTO, |
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76 sinks: str = SINKS_AUTO, |
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77 appsink_data: AppSinkData | None = None, |
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78 reset_cb: Callable | None = None, |
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79 ) -> None: |
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80 self.main_loop = asyncio.get_event_loop() |
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81 self.bridge = bridge |
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82 self.profile = profile |
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83 self.pipeline = None |
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84 self._audio_muted = False |
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85 self._video_muted = False |
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86 self.sources = sources |
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87 self.sinks = sinks |
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88 if sinks == SINKS_APP: |
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89 self.appsink_data = appsink_data |
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90 elif appsink_data is not None: |
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91 raise exceptions.InternalError( |
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92 "appsink_data can only be used for SINKS_APP sinks" |
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93 ) |
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94 self.reset_cb = reset_cb |
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95 self.reset_instance() |
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96 |
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97 @property |
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98 def audio_muted(self): |
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99 return self._audio_muted |
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100 |
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101 @audio_muted.setter |
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102 def audio_muted(self, muted: bool) -> None: |
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103 if muted != self._audio_muted: |
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104 self._audio_muted = muted |
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105 self.on_audio_mute(muted) |
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106 |
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107 @property |
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108 def video_muted(self): |
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109 return self._video_muted |
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110 |
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111 @video_muted.setter |
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112 def video_muted(self, muted: bool) -> None: |
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113 if muted != self._video_muted: |
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114 self._video_muted = muted |
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115 self.on_video_mute(muted) |
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116 |
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117 @property |
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118 def sdp_set(self): |
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119 return self._sdp_set |
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120 |
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121 @sdp_set.setter |
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122 def sdp_set(self, is_set: bool): |
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123 self._sdp_set = is_set |
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124 if is_set: |
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125 self.on_ice_candidates_new(self.remote_candidates_buffer) |
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126 for data in self.remote_candidates_buffer.values(): |
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127 data["candidates"].clear() |
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128 |
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129 @property |
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130 def media_types(self): |
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131 if self._media_types is None: |
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132 raise Exception("self._media_types should not be None!") |
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133 return self._media_types |
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134 |
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135 @media_types.setter |
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136 def media_types(self, new_media_types: dict) -> None: |
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137 self._media_types = new_media_types |
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138 self._media_types_inv = {v: k for k, v in new_media_types.items()} |
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139 |
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140 @property |
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141 def media_types_inv(self) -> dict: |
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142 if self._media_types_inv is None: |
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143 raise Exception("self._media_types_inv should not be None!") |
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144 return self._media_types_inv |
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145 |
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146 def generate_dot_file( |
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147 self, |
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148 filename: str = "pipeline", |
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149 details: Gst.DebugGraphDetails = Gst.DebugGraphDetails.ALL, |
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150 with_timestamp: bool = True, |
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151 bin_: Gst.Bin|None = None, |
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152 ) -> None: |
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153 """Generate Dot File for debugging |
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154 |
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155 ``GST_DEBUG_DUMP_DOT_DIR`` environment variable must be set to destination dir. |
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156 ``dot -Tpng -o <filename>.png <filename>.dot`` can be use to convert to a PNG file. |
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157 See |
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158 https://gstreamer.freedesktop.org/documentation/gstreamer/debugutils.html?gi-language=python#GstDebugGraphDetails |
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159 for details. |
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160 |
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161 @param filename: name of the generated file |
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162 @param details: which details to print |
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163 @param with_timestamp: if True, add a timestamp to filename |
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164 @param bin_: which bin to output. By default, the whole pipeline |
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165 (``self.pipeline``) will be used. |
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166 """ |
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167 if bin_ is None: |
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168 bin_ = self.pipeline |
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169 if with_timestamp: |
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170 timestamp = datetime.now().isoformat(timespec='milliseconds') |
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171 filename = f"{timestamp}_filename" |
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172 |
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173 Gst.debug_bin_to_dot_file(bin_, details, filename) |
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174 |
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175 def get_sdp_mline_index(self, media_type: str) -> int: |
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176 """Gets the sdpMLineIndex for a given media type. |
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177 |
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178 @param media_type: The type of the media. |
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179 """ |
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180 for index, m_type in self.media_types.items(): |
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181 if m_type == media_type: |
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182 return index |
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183 raise ValueError(f"Media type '{media_type}' not found") |
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184 |
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185 def _set_media_types(self, offer_sdp: str) -> None: |
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186 """Sets media types from offer SDP |
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187 |
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188 @param offer: RTC session description containing the offer |
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189 """ |
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190 sdp_lines = offer_sdp.splitlines() |
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191 media_types = {} |
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192 mline_index = 0 |
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193 |
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194 for line in sdp_lines: |
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195 if line.startswith("m="): |
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196 media_types[mline_index] = line[2 : line.find(" ")] |
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197 mline_index += 1 |
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198 |
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199 self.media_types = media_types |
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200 |
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201 def _a_call(self, method, *args, **kwargs): |
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202 """Call an async method in main thread""" |
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203 aio.run_from_thread(method, *args, **kwargs, loop=self.main_loop) |
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204 |
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205 def get_payload_types( |
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206 self, sdpmsg, video_encoding: str, audio_encoding: str |
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207 ) -> dict[str, int | None]: |
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208 """Find the payload types for the specified video and audio encoding. |
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209 |
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210 Very simplistically finds the first payload type matching the encoding |
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211 name. More complex applications will want to match caps on |
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212 profile-level-id, packetization-mode, etc. |
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213 """ |
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214 # method coming from gstreamer example (Matthew Waters, Nirbheek Chauhan) at |
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215 # subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py |
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216 video_pt = None |
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217 audio_pt = None |
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218 for i in range(0, sdpmsg.medias_len()): |
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219 media = sdpmsg.get_media(i) |
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220 for j in range(0, media.formats_len()): |
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221 fmt = media.get_format(j) |
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222 if fmt == "webrtc-datachannel": |
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223 continue |
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224 pt = int(fmt) |
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225 caps = media.get_caps_from_media(pt) |
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226 s = caps.get_structure(0) |
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227 encoding_name = s["encoding-name"] |
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228 if video_pt is None and encoding_name == video_encoding: |
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229 video_pt = pt |
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230 elif audio_pt is None and encoding_name == audio_encoding: |
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231 audio_pt = pt |
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232 return {video_encoding: video_pt, audio_encoding: audio_pt} |
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233 |
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234 def parse_ice_candidate(self, candidate_string): |
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235 """Parses the ice candidate string. |
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236 |
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237 @param candidate_string: The ice candidate string to be parsed. |
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238 """ |
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239 pattern = re.compile( |
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240 r"candidate:(?P<foundation>\S+) (?P<component_id>\d+) (?P<transport>\S+) " |
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241 r"(?P<priority>\d+) (?P<address>\S+) (?P<port>\d+) typ " |
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242 r"(?P<type>\S+)(?: raddr (?P<rel_addr>\S+) rport " |
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243 r"(?P<rel_port>\d+))?(?: generation (?P<generation>\d+))?" |
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244 ) |
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245 match = pattern.match(candidate_string) |
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246 if match: |
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247 candidate_dict = match.groupdict() |
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248 |
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249 # Apply the correct types to the dictionary values |
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250 candidate_dict["component_id"] = int(candidate_dict["component_id"]) |
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251 candidate_dict["priority"] = int(candidate_dict["priority"]) |
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252 candidate_dict["port"] = int(candidate_dict["port"]) |
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253 |
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254 if candidate_dict["rel_port"]: |
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255 candidate_dict["rel_port"] = int(candidate_dict["rel_port"]) |
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256 |
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257 if candidate_dict["generation"]: |
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258 candidate_dict["generation"] = candidate_dict["generation"] |
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259 |
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260 # Remove None values |
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261 return {k: v for k, v in candidate_dict.items() if v is not None} |
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262 else: |
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263 log.warning(f"can't parse candidate: {candidate_string!r}") |
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264 return None |
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265 |
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266 def build_ice_candidate(self, parsed_candidate): |
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267 """Builds ICE candidate |
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268 |
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269 @param parsed_candidate: Dictionary containing parsed ICE candidate |
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270 """ |
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271 base_format = ( |
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272 "candidate:{foundation} {component_id} {transport} {priority} " |
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273 "{address} {port} typ {type}" |
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274 ) |
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275 |
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276 if parsed_candidate.get("rel_addr") and parsed_candidate.get("rel_port"): |
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277 base_format += " raddr {rel_addr} rport {rel_port}" |
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278 |
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279 if parsed_candidate.get("generation"): |
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280 base_format += " generation {generation}" |
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281 |
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282 return base_format.format(**parsed_candidate) |
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283 |
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284 def extract_ufrag_pwd(self, sdp: str) -> tuple[str, str]: |
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285 """Retrieves ICE password and user fragment for SDP offer. |
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286 |
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287 @param sdp: The Session Description Protocol offer string. |
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288 @return: ufrag and pwd |
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289 @raise ValueError: Can't extract ufrag and password |
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290 """ |
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291 ufrag_line = re.search(r"ice-ufrag:(\S+)", sdp) |
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292 pwd_line = re.search(r"ice-pwd:(\S+)", sdp) |
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293 |
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294 if ufrag_line and pwd_line: |
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295 ufrag = self.ufrag = ufrag_line.group(1) |
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296 pwd = self.pwd = pwd_line.group(1) |
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297 return ufrag, pwd |
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298 else: |
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299 log.error(f"SDP with missing ice-ufrag or ice-pwd:\n{sdp}") |
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300 raise ValueError("Can't extract ice-ufrag and ice-pwd from SDP") |
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301 |
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302 def reset_instance(self): |
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303 """Inits or resets the instance variables to their default state.""" |
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304 self.role: str | None = None |
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305 if self.pipeline is not None: |
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306 self.pipeline.set_state(Gst.State.NULL) |
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307 self.pipeline = None |
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308 self._remote_video_pad = None |
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309 self.sid: str | None = None |
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310 self.offer: str | None = None |
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311 self.local_candidates_buffer = {} |
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312 self.ufrag: str | None = None |
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313 self.pwd: str | None = None |
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314 self.callee: str | None = None |
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315 self._media_types = None |
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316 self._media_types_inv = None |
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317 self._sdp_set: bool = False |
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318 self.remote_candidates_buffer: dict[str, dict[str, list]] = { |
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319 "audio": {"candidates": []}, |
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320 "video": {"candidates": []}, |
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321 } |
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322 self._media_types = None |
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323 self._media_types_inv = None |
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324 self.audio_valve = None |
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325 self.video_valve = None |
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326 if self.reset_cb is not None: |
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327 self.reset_cb() |
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328 |
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329 |
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330 async def setup_call( |
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331 self, |
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332 role: str, |
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333 audio_pt: int | None = 96, |
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334 video_pt: int | None = 97, |
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335 ) -> None: |
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336 """Sets up the call. |
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337 |
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338 This method establishes the Gstreamer pipeline for audio and video communication. |
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339 The method also manages STUN and TURN server configurations, signal watchers, and |
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340 various connection handlers for the webrtcbin. |
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341 |
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342 @param role: The role for the call, either 'initiator' or 'responder'. |
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343 @param audio_pt: The payload type for the audio stream. |
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344 @param video_pt: The payload type for the video stream |
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345 |
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346 @raises NotImplementedError: If audio_pt or video_pt is set to None. |
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347 @raises AssertionError: If the role is not 'initiator' or 'responder'. |
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348 """ |
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349 assert role in ("initiator", "responder") |
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350 self.role = role |
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351 if audio_pt is None or video_pt is None: |
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352 raise NotImplementedError("None value is not handled yet") |
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353 |
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354 if self.sources == SOURCES_AUTO: |
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355 video_source_elt = "v4l2src" |
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356 audio_source_elt = "pulsesrc" |
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357 elif self.sources == SOURCES_TEST: |
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358 video_source_elt = "videotestsrc is-live=true pattern=ball" |
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359 audio_source_elt = "audiotestsrc" |
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360 else: |
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361 raise exceptions.InternalError(f'Unknown "sources" value: {self.sources!r}') |
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362 |
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363 extra_elt = "" |
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364 |
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365 if self.sinks == SINKS_APP: |
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366 local_video_sink_elt = ( |
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367 "appsink name=local_video_sink emit-signals=true drop=true max-buffers=1 " |
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368 "sync=True" |
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369 ) |
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370 elif self.sinks == SINKS_AUTO: |
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371 extra_elt = "compositor name=compositor ! autovideosink" |
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372 local_video_sink_elt = """compositor.sink_1""" |
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373 else: |
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374 raise exceptions.InternalError(f"Unknown sinks value {self.sinks!r}") |
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375 |
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376 self.gst_pipe_desc = f""" |
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377 webrtcbin latency=100 name=sendrecv bundle-policy=max-bundle |
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378 |
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379 input-selector name=video_selector |
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380 ! videorate |
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381 ! video/x-raw,framerate=30/1 |
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382 ! tee name=t |
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383 |
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384 {extra_elt} |
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385 |
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386 {video_source_elt} name=video_src ! queue leaky=downstream ! video_selector. |
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387 videotestsrc is-live=true pattern=black ! queue leaky=downstream ! video_selector. |
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388 |
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389 t. |
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390 ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream |
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391 ! videoconvert |
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392 ! vp8enc deadline=1 keyframe-max-dist=60 |
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393 ! rtpvp8pay picture-id-mode=15-bit |
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394 ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt} |
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395 ! sendrecv. |
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396 |
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397 t. |
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398 ! queue max-size-buffers=5 max-size-time=0 max-size-bytes=0 leaky=downstream |
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399 ! videoconvert |
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400 ! {local_video_sink_elt} |
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401 |
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402 {audio_source_elt} name=audio_src |
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403 ! valve |
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404 ! queue max-size-buffers=10 max-size-time=0 max-size-bytes=0 leaky=downstream |
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405 ! audioconvert |
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406 ! audioresample |
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407 ! opusenc audio-type=voice |
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408 ! rtpopuspay |
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409 ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} |
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410 ! sendrecv. |
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411 """ |
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412 |
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413 log.debug(f"Gstreamer pipeline: {self.gst_pipe_desc}") |
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414 |
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415 # Create the pipeline |
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416 self.pipeline = Gst.parse_launch(self.gst_pipe_desc) |
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417 if not self.pipeline: |
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418 raise exceptions.InternalError("Failed to create Gstreamer pipeline.") |
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419 |
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420 self.webrtcbin = self.pipeline.get_by_name("sendrecv") |
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421 self.video_src = self.pipeline.get_by_name("video_src") |
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422 self.video_selector = self.pipeline.get_by_name("video_selector") |
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423 self.audio_valve = self.pipeline.get_by_name("audio_valve") |
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424 |
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425 if self.video_muted: |
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426 self.on_video_mute(True) |
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427 if self.audio_muted: |
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428 self.on_audio_mute(True) |
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429 |
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430 # set STUN and TURN servers |
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431 external_disco = data_format.deserialise( |
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432 await self.bridge.external_disco_get("", self.profile), type_check=list |
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433 ) |
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434 |
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435 for server in external_disco: |
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436 if server["type"] == "stun": |
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437 if server["transport"] == "tcp": |
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438 log.info( |
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439 "ignoring TCP STUN server, GStreamer only support one STUN server" |
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440 ) |
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441 url = f"stun://{server['host']}:{server['port']}" |
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442 log.debug(f"adding stun server: {url}") |
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443 self.webrtcbin.set_property("stun-server", url) |
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444 elif server["type"] == "turn": |
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445 url = "{scheme}://{username}:{password}@{host}:{port}".format( |
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446 scheme="turns" if server["transport"] == "tcp" else "turn", |
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447 username=quote_plus(server["username"]), |
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448 password=quote_plus(server["password"]), |
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449 host=server["host"], |
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450 port=server["port"], |
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451 ) |
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452 log.debug(f"adding turn server: {url}") |
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453 |
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454 if not self.webrtcbin.emit("add-turn-server", url): |
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455 log.warning(f"Erreur while adding TURN server {url}") |
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456 |
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457 # local video feedback |
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458 if self.sinks == SINKS_APP: |
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459 assert self.appsink_data is not None |
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460 local_video_sink = self.pipeline.get_by_name("local_video_sink") |
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461 local_video_sink.set_property("emit-signals", True) |
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462 local_video_sink.connect("new-sample", self.appsink_data.local_video_cb) |
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463 local_video_sink_caps = Gst.Caps.from_string(f"video/x-raw,format=RGB") |
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464 local_video_sink.set_property("caps", local_video_sink_caps) |
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465 |
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466 # Create bus and associate signal watchers |
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467 self.bus = self.pipeline.get_bus() |
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468 if not self.bus: |
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469 log.error("Failed to get bus from pipeline.") |
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470 return |
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471 |
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472 self.bus.add_signal_watch() |
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473 self.webrtcbin.connect("pad-added", self.on_pad_added) |
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474 self.bus.connect("message::error", self.on_bus_error) |
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475 self.bus.connect("message::eos", self.on_bus_eos) |
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476 self.webrtcbin.connect("on-negotiation-needed", self.on_negotiation_needed) |
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477 self.webrtcbin.connect("on-ice-candidate", self.on_ice_candidate) |
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478 self.webrtcbin.connect( |
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479 "notify::ice-gathering-state", self.on_ice_gathering_state_change |
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480 ) |
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481 self.webrtcbin.connect( |
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482 "notify::ice-connection-state", self.on_ice_connection_state |
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483 ) |
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484 |
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485 def start_pipeline(self) -> None: |
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486 """Starts the GStreamer pipeline.""" |
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487 log.debug("starting the pipeline") |
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488 self.pipeline.set_state(Gst.State.PLAYING) |
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489 |
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490 def on_negotiation_needed(self, webrtc): |
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491 """Initiate SDP offer when negotiation is needed.""" |
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492 log.debug("Negotiation needed.") |
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493 if self.role == "initiator": |
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494 log.debug("Creating offer…") |
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495 promise = Gst.Promise.new_with_change_func(self.on_offer_created) |
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496 self.webrtcbin.emit("create-offer", None, promise) |
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497 |
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498 def on_offer_created(self, promise): |
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499 """Callback for when SDP offer is created.""" |
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500 log.info("on_offer_created called") |
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501 assert promise.wait() == Gst.PromiseResult.REPLIED |
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502 reply = promise.get_reply() |
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503 if reply is None: |
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504 log.error("Promise reply is None. Offer creation might have failed.") |
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505 return |
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506 offer = reply["offer"] |
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507 self.offer = offer.sdp.as_text() |
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508 log.info(f"SDP offer created: \n{self.offer}") |
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509 self._set_media_types(self.offer) |
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510 promise = Gst.Promise.new() |
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511 self.webrtcbin.emit("set-local-description", offer, promise) |
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512 promise.interrupt() |
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513 self._a_call(self._start_call) |
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514 |
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515 def on_answer_set(self, promise): |
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516 assert promise.wait() == Gst.PromiseResult.REPLIED |
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517 |
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518 def on_answer_created(self, promise, _, __): |
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519 """Callback for when SDP answer is created.""" |
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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 answer = reply["answer"] |
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523 promise = Gst.Promise.new() |
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524 self.webrtcbin.emit("set-local-description", answer, promise) |
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525 promise.interrupt() |
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526 answer_sdp = answer.sdp.as_text() |
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527 log.info(f"SDP answer set: \n{answer_sdp}") |
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528 self.sdp_set = True |
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529 self._a_call(self.bridge.call_answer_sdp, self.sid, answer_sdp, self.profile) |
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530 |
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531 def on_offer_set(self, promise): |
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532 assert promise.wait() == Gst.PromiseResult.REPLIED |
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533 promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None) |
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534 self.webrtcbin.emit("create-answer", None, promise) |
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535 |
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536 def link_element_or_pad( |
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537 self, source: Gst.Element, dest: Gst.Element | Gst.Pad |
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538 ) -> bool: |
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539 """Check if dest is a pad or an element, and link appropriately""" |
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540 src_pad = source.get_static_pad("src") |
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541 |
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542 if isinstance(dest, Gst.Pad): |
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543 # If the dest is a pad, link directly |
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544 if not src_pad.link(dest) == Gst.PadLinkReturn.OK: |
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545 log.error( |
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546 "Failed to link 'conv' to the compositor's newly requested pad!" |
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547 ) |
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548 return False |
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549 elif isinstance(dest, Gst.Element): |
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550 return source.link(dest) |
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551 else: |
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552 log.error(f"Unexpected type for dest: {type(sink)}") |
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553 return False |
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554 |
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555 return True |
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556 |
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557 def scaled_dimensions( |
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558 self, original_width: int, original_height: int, max_width: int, max_height: int |
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559 ) -> tuple[int, int]: |
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560 """Calculates the scaled dimensions preserving aspect ratio. |
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561 |
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562 @param original_width: Original width of the video stream. |
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563 @param original_height: Original height of the video stream. |
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564 @param max_width: Maximum desired width for the scaled video. |
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565 @param max_height: Maximum desired height for the scaled video. |
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566 @return: The width and height of the scaled video. |
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567 """ |
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568 aspect_ratio = original_width / original_height |
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569 new_width = int(max_height * aspect_ratio) |
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570 |
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571 if new_width <= max_width: |
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572 return new_width, max_height |
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573 |
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574 new_height = int(max_width / aspect_ratio) |
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575 return max_width, new_height |
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576 |
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577 def on_remote_decodebin_stream(self, _, pad: Gst.Pad) -> None: |
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578 """Handle the stream from the remote decodebin. |
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579 |
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580 This method processes the incoming stream from the remote decodebin, determining |
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581 whether it's video or audio. It then sets up the appropriate GStreamer elements |
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582 for video/audio processing and adds them to the pipeline. |
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583 |
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584 @param pad: The Gst.Pad from the remote decodebin producing the stream. |
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585 """ |
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586 assert self.pipeline is not None |
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587 if not pad.has_current_caps(): |
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588 log.error(f"{pad} has no caps, ignoring") |
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589 return |
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590 |
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591 caps = pad.get_current_caps() |
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592 assert len(caps) |
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593 s = caps[0] |
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594 name = s.get_name() |
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595 log.debug(f"====> NAME START: {name}") |
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596 |
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597 q = Gst.ElementFactory.make("queue") |
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598 |
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|
599 if name.startswith("video"): |
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600 log.debug("===> VIDEO OK") |
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601 |
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602 self._remote_video_pad = pad |
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603 |
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|
604 # Check and log the original size of the video |
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605 width = s.get_int("width").value |
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606 height = s.get_int("height").value |
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607 log.info(f"Original video size: {width}x{height}") |
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608 |
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609 # This is a fix for an issue found with Movim on desktop: a non standard |
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610 # resolution is used (990x557) resulting in bad alignement and no color in |
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611 # rendered image |
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612 adjust_resolution = width % 4 != 0 or height % 4 != 0 |
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613 if adjust_resolution: |
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614 log.info("non standard resolution, we need to adjust size") |
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615 width = (width + 3) // 4 * 4 |
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616 height = (height + 3) // 4 * 4 |
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617 log.info(f"Adjusted video size: {width}x{height}") |
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618 |
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619 conv = Gst.ElementFactory.make("videoconvert") |
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620 if self.sinks == SINKS_APP: |
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621 assert self.appsink_data is not None |
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622 remote_video_sink = Gst.ElementFactory.make("appsink") |
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623 |
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624 appsink_caps = Gst.Caps.from_string("video/x-raw,format=RGB") |
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625 remote_video_sink.set_property("caps", appsink_caps) |
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626 |
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627 remote_video_sink.set_property("emit-signals", True) |
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628 remote_video_sink.set_property("drop", True) |
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629 remote_video_sink.set_property("max-buffers", 1) |
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630 remote_video_sink.set_property("sync", True) |
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631 remote_video_sink.connect("new-sample", self.appsink_data.remote_video_cb) |
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632 self.pipeline.add(remote_video_sink) |
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633 elif self.sinks == SINKS_AUTO: |
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634 compositor = self.pipeline.get_by_name("compositor") |
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635 |
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636 sink1_pad = compositor.get_static_pad("sink_1") |
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637 |
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638 local_width, local_height = self.scaled_dimensions( |
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639 sink1_pad.get_property("width"), |
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640 sink1_pad.get_property("height"), |
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641 width // 3, |
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642 height // 3, |
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643 ) |
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644 |
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645 sink1_pad.set_property("xpos", width - local_width) |
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646 sink1_pad.set_property("ypos", height - local_height) |
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647 sink1_pad.set_property("width", local_width) |
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648 sink1_pad.set_property("height", local_height) |
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649 sink1_pad.set_property("zorder", 1) |
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650 |
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651 # Request a new pad for the remote stream |
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652 sink_pad_template = compositor.get_pad_template("sink_%u") |
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653 remote_video_sink = compositor.request_pad(sink_pad_template, None, None) |
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654 remote_video_sink.set_property("zorder", 0) |
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655 |
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656 else: |
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657 raise exceptions.InternalError(f'Unhandled "sinks" value: {self.sinks!r}') |
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658 |
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659 if adjust_resolution: |
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660 videoscale = Gst.ElementFactory.make("videoscale") |
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661 adjusted_caps = Gst.Caps.from_string( |
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662 f"video/x-raw,width={width},height={height}" |
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663 ) |
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664 capsfilter = Gst.ElementFactory.make("capsfilter") |
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665 capsfilter.set_property("caps", adjusted_caps) |
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666 |
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667 self.pipeline.add(q, conv, videoscale, capsfilter) |
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668 |
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669 |
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670 self.pipeline.sync_children_states() |
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671 ret = pad.link(q.get_static_pad("sink")) |
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672 if ret != Gst.PadLinkReturn.OK: |
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673 log.error(f"Error linking pad: {ret}") |
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674 q.link(conv) |
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675 conv.link(videoscale) |
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676 videoscale.link(capsfilter) |
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677 self.link_element_or_pad(capsfilter.link, remote_video_sink) |
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678 |
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679 else: |
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680 self.pipeline.add(q, conv) |
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681 |
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682 self.pipeline.sync_children_states() |
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683 ret = pad.link(q.get_static_pad("sink")) |
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684 if ret != Gst.PadLinkReturn.OK: |
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685 log.error(f"Error linking pad: {ret}") |
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686 q.link(conv) |
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687 self.link_element_or_pad(conv, remote_video_sink) |
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688 |
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689 elif name.startswith("audio"): |
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690 log.debug("===> Audio OK") |
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691 conv = Gst.ElementFactory.make("audioconvert") |
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692 resample = Gst.ElementFactory.make("audioresample") |
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693 remote_audio_sink = Gst.ElementFactory.make("autoaudiosink") |
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694 self.pipeline.add(q, conv, resample, remote_audio_sink) |
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695 self.pipeline.sync_children_states() |
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696 ret = pad.link(q.get_static_pad("sink")) |
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697 if ret != Gst.PadLinkReturn.OK: |
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698 log.error(f"Error linking pad: {ret}") |
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699 q.link(conv) |
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700 conv.link(resample) |
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701 resample.link(remote_audio_sink) |
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702 |
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703 else: |
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704 log.warning(f"unmanaged name: {name!r}") |
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705 |
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706 def on_pad_added(self, __, pad: Gst.Pad) -> None: |
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707 """Handle the addition of a new pad to the element. |
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708 |
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709 When a new source pad is added to the element, this method creates a decodebin, |
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710 connects it to handle the stream, and links the pad to the decodebin. |
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711 |
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712 @param __: Placeholder for the signal source. Not used in this method. |
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713 @param pad: The newly added pad. |
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714 """ |
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715 log.debug("on_pad_added") |
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716 if pad.direction != Gst.PadDirection.SRC: |
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717 return |
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718 |
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719 decodebin = Gst.ElementFactory.make("decodebin") |
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720 decodebin.connect("pad-added", self.on_remote_decodebin_stream) |
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721 self.pipeline.add(decodebin) |
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722 decodebin.sync_state_with_parent() |
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723 pad.link(decodebin.get_static_pad("sink")) |
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724 |
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725 async def _start_call(self) -> None: |
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726 """Initiate the call. |
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727 |
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728 Initiates a call with the callee using the stored offer. If there are any buffered |
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729 local ICE candidates, they are sent as part of the initiation. |
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730 """ |
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731 assert self.callee |
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732 self.sid = await self.bridge.call_start( |
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733 str(self.callee), data_format.serialise({"sdp": self.offer}), self.profile |
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734 ) |
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735 if self.local_candidates_buffer: |
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736 log.debug( |
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737 f"sending buffered local ICE candidates: {self.local_candidates_buffer}" |
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738 ) |
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739 if self.pwd is None: |
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740 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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741 self.extract_ufrag_pwd(sdp) |
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742 ice_data = {} |
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743 for media_type, candidates in self.local_candidates_buffer.items(): |
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744 ice_data[media_type] = { |
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745 "ufrag": self.ufrag, |
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746 "pwd": self.pwd, |
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747 "candidates": candidates, |
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748 } |
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749 await self.bridge.ice_candidates_add( |
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750 self.sid, data_format.serialise(ice_data), self.profile |
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751 ) |
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752 self.local_candidates_buffer.clear() |
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753 |
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754 def _remote_sdp_set(self, promise) -> None: |
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755 assert promise.wait() == Gst.PromiseResult.REPLIED |
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756 self.sdp_set = True |
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757 |
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758 def on_accepted_call(self, sdp: str, profile: str) -> None: |
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759 """Outgoing call has been accepted. |
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760 |
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761 @param sdp: The SDP answer string received from the other party. |
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762 @param profile: Profile used for the call. |
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763 """ |
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764 log.debug(f"SDP answer received: \n{sdp}") |
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765 |
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766 __, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp) |
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767 answer = GstWebRTC.WebRTCSessionDescription.new( |
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768 GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg |
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769 ) |
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770 promise = Gst.Promise.new_with_change_func(self._remote_sdp_set) |
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771 self.webrtcbin.emit("set-remote-description", answer, promise) |
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772 |
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773 async def answer_call(self, sdp: str, profile: str) -> None: |
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774 """Answer an incoming call |
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775 |
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776 @param sdp: The SDP offer string received from the initiator. |
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777 @param profile: Profile used for the call. |
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778 |
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779 @raise AssertionError: Raised when either "VP8" or "OPUS" is not present in |
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780 payload types. |
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781 """ |
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782 log.debug(f"SDP offer received: \n{sdp}") |
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783 self._set_media_types(sdp) |
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784 __, offer_sdp_msg = GstSdp.SDPMessage.new_from_text(sdp) |
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785 payload_types = self.get_payload_types( |
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786 offer_sdp_msg, video_encoding="VP8", audio_encoding="OPUS" |
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787 ) |
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788 assert "VP8" in payload_types |
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789 assert "OPUS" in payload_types |
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790 await self.setup_call( |
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791 "responder", audio_pt=payload_types["OPUS"], video_pt=payload_types["VP8"] |
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792 ) |
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793 self.start_pipeline() |
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794 offer = GstWebRTC.WebRTCSessionDescription.new( |
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795 GstWebRTC.WebRTCSDPType.OFFER, offer_sdp_msg |
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796 ) |
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797 promise = Gst.Promise.new_with_change_func(self.on_offer_set) |
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798 self.webrtcbin.emit("set-remote-description", offer, promise) |
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|
799 |
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800 def on_ice_candidate(self, webrtc, mline_index, candidate_sdp): |
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801 """Handles the on-ice-candidate signal of webrtcbin. |
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802 |
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803 @param webrtc: The webrtcbin element. |
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804 @param mlineindex: The mline index. |
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805 @param candidate: The ICE candidate. |
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|
806 """ |
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|
807 log.debug( |
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|
808 f"Local ICE candidate. MLine Index: {mline_index}, Candidate: {candidate_sdp}" |
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809 ) |
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810 parsed_candidate = self.parse_ice_candidate(candidate_sdp) |
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811 try: |
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812 media_type = self.media_types[mline_index] |
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|
813 except KeyError: |
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814 raise exceptions.InternalError("can't find media type") |
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|
815 |
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816 if self.sid is None: |
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817 log.debug("buffering local ICE candidate") |
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818 self.local_candidates_buffer.setdefault(media_type, []).append( |
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819 parsed_candidate |
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|
820 ) |
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821 else: |
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822 sdp = self.webrtcbin.props.local_description.sdp.as_text() |
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823 assert sdp is not None |
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824 ufrag, pwd = self.extract_ufrag_pwd(sdp) |
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825 ice_data = {"ufrag": ufrag, "pwd": pwd, "candidates": [parsed_candidate]} |
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826 self._a_call( |
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827 self.bridge.ice_candidates_add, |
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828 self.sid, |
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829 data_format.serialise({media_type: ice_data}), |
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830 self.profile, |
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831 ) |
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832 |
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833 def on_ice_candidates_new(self, candidates: dict) -> None: |
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834 """Handle new ICE candidates. |
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|
835 |
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836 @param candidates: A dictionary containing media types ("audio" or "video") as |
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837 keys and corresponding ICE data as values. |
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838 |
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839 @raise exceptions.InternalError: Raised when sdp mline index is not found. |
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840 """ |
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841 if not self.sdp_set: |
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842 log.debug("buffering remote ICE candidate") |
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843 for media_type in ("audio", "video"): |
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844 media_candidates = candidates.get(media_type) |
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845 if media_candidates: |
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846 buffer = self.remote_candidates_buffer[media_type] |
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847 buffer["candidates"].extend(media_candidates["candidates"]) |
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848 return |
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849 for media_type, ice_data in candidates.items(): |
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850 for candidate in ice_data["candidates"]: |
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851 candidate_sdp = self.build_ice_candidate(candidate) |
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852 try: |
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853 mline_index = self.get_sdp_mline_index(media_type) |
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854 except Exception as e: |
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855 raise exceptions.InternalError(f"Can't find sdp mline index: {e}") |
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856 self.webrtcbin.emit("add-ice-candidate", mline_index, candidate_sdp) |
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857 log.debug( |
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858 f"Remote ICE candidate added. MLine Index: {mline_index}, " |
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859 f"{candidate_sdp}" |
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860 ) |
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861 |
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862 def on_ice_gathering_state_change(self, pspec, __): |
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863 state = self.webrtcbin.get_property("ice-gathering-state") |
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864 log.debug(f"ICE gathering state changed to {state}") |
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865 |
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866 def on_ice_connection_state(self, pspec, __): |
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867 state = self.webrtcbin.props.ice_connection_state |
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868 if state == GstWebRTC.WebRTCICEConnectionState.FAILED: |
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869 log.error("ICE connection failed") |
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870 log.info(f"ICE connection state changed to {state}") |
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871 |
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872 def on_bus_error(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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873 """Handles the GStreamer bus error messages. |
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874 |
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875 @param bus: The GStreamer bus. |
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876 @param message: The error message. |
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877 """ |
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878 err, debug = message.parse_error() |
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879 log.error(f"Error from {message.src.get_name()}: {err.message}") |
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880 log.error(f"Debugging info: {debug}") |
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881 |
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882 def on_bus_eos(self, bus: Gst.Bus, message: Gst.Message) -> None: |
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883 """Handles the GStreamer bus eos messages. |
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884 |
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885 @param bus: The GStreamer bus. |
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886 @param message: The eos message. |
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887 """ |
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888 log.info("End of stream") |
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889 |
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890 def on_audio_mute(self, muted: bool) -> None: |
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891 if self.audio_valve is not None: |
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892 self.audio_valve.set_property("drop", muted) |
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893 state = "muted" if muted else "unmuted" |
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894 log.info(f"audio is now {state}") |
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895 |
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896 def on_video_mute(self, muted: bool) -> None: |
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897 if self.video_selector is not None: |
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898 # when muted, we switch to a black image and deactivate the camera |
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899 if not muted: |
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900 self.video_src.set_state(Gst.State.PLAYING) |
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901 pad = self.video_selector.get_static_pad("sink_1" if muted else "sink_0") |
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902 self.video_selector.props.active_pad = pad |
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903 if muted: |
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904 self.video_src.set_state(Gst.State.NULL) |
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905 state = "muted" if muted else "unmuted" |
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906 log.info(f"video is now {state}") |
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907 |
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908 async def end_call(self) -> None: |
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909 """Stop streaming and clean instance""" |
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910 self.reset_instance() |