Mercurial > libervia-backend
annotate libervia/frontends/tools/webrtc_models.py @ 4242:8acf46ed7f36
frontends: remote control implementation:
This is the frontends common part of remote control implementation. It handle the creation
of WebRTC session, and management of inputs. For now the reception use freedesktop.org
Desktop portal, and works mostly with Wayland based Desktop Environments.
rel 436
author | Goffi <goffi@goffi.org> |
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date | Sat, 11 May 2024 13:52:43 +0200 |
parents | 79c8a70e1813 |
children | 0d7bb4df2343 |
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1 #!/usr/bin/env python3 |
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2 |
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3 # Libervia WebRTC implementation |
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4 # Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org) |
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5 |
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6 # This program is free software: you can redistribute it and/or modify |
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7 # it under the terms of the GNU Affero General Public License as published by |
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8 # the Free Software Foundation, either version 3 of the License, or |
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9 # (at your option) any later version. |
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10 |
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11 # This program is distributed in the hope that it will be useful, |
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12 # but WITHOUT ANY WARRANTY; without even the implied warranty of |
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13 # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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14 # GNU Affero General Public License for more details. |
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15 |
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16 # You should have received a copy of the GNU Affero General Public License |
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17 # along with this program. If not, see <http://www.gnu.org/licenses/>. |
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18 |
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19 from collections.abc import Awaitable |
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20 from dataclasses import dataclass, field |
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21 from typing import Any, Callable |
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22 import uuid |
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23 import gi |
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24 |
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25 gi.require_versions({"Gst": "1.0", "GstWebRTC": "1.0"}) |
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26 from gi.repository import GstWebRTC |
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27 from pydantic import BaseModel, Field |
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28 |
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29 from libervia.frontends.tools import jid |
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30 |
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31 |
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32 @dataclass |
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33 class CallData: |
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34 callee: jid.JID |
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35 sid: str | None = None |
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36 action_id: str | None = None |
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37 kwargs: dict[str, Any] = field(default_factory=dict) |
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38 |
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39 |
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40 class SourcesData(BaseModel): |
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41 """Data for Sources""" |
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42 |
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43 |
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44 class SourcesNone(SourcesData): |
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45 """No source is used. |
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46 |
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47 This is used when the WebRTC connection will be used for data channels only.""" |
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48 |
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49 |
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50 class SourcesAuto(SourcesData): |
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51 """Automatic Sources (webcam/microphone)""" |
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52 |
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53 |
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54 class SourcesTest(SourcesData): |
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55 """Test Sources (pattern)""" |
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56 |
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57 |
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58 class SourcesDataChannel(SourcesData): |
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59 """Sources for transmitting data over Data Channel |
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60 |
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61 |
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62 @param dc_open_cb: Called when Data Channel is open. |
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63 This callback will be run in a GStreamer thread. |
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64 """ |
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65 name: str = Field(default_factory=lambda: str(uuid.uuid4())) |
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66 dc_open_cb: Callable[[GstWebRTC.WebRTCDataChannel], None] |
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67 |
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68 |
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69 class SourcesPipeline(SourcesData): |
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70 """Use custom pipeline description as a source. |
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71 |
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72 @param video_pipeline: pipeline description of video source. |
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73 None to use automatic video source (same as SourcesAuto). |
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74 Empty string to disable video. |
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75 @param audio_pipeline: pipeline description of audio source. |
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76 None to use automatic audio source (same as SourcesAuto). |
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77 Empty string to disable audio. |
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78 @param video_properties: Elements properties to set. |
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79 @param audio_properties: Elements properties to set. |
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80 |
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81 """ |
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82 video_pipeline: str|None = None |
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83 audio_pipeline: str|None = None |
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84 video_properties: dict = Field(default_factory=lambda: {}) |
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85 audio_properties: dict = Field(default_factory=lambda: {}) |
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86 |
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87 |
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88 class SinksData(BaseModel): |
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89 """Data for Sinks""" |
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90 |
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91 |
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92 class SinksNone(SinksData): |
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93 """No sink is used. |
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94 |
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95 This is used when the WebRTC connection will be used for data channels only.""" |
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96 |
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97 |
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98 class SinksAuto(SinksData): |
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99 """Automatic Sinks (create windows/default audio)""" |
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100 |
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101 |
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102 class SinksApp(SinksData): |
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103 local_video_cb: Callable |
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104 remote_video_cb: Callable | None |
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105 |
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106 |
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107 class SinksDataChannel(SinksData): |
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108 """Sinks for transmitting data over Data Channel |
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109 |
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110 @param dc_on_data_channel: Called when Data Channel is created. |
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111 This callback will be run in a GStreamer thread. |
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112 """ |
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113 |
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114 dc_on_data_channel: ( |
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115 Callable[[GstWebRTC.WebRTCDataChannel], Awaitable[None]] | None |
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116 ) = None |