de7260bfb9
This is a series of changes necessary to prepare the implementation of remote control feature: - XEP-0166: add a `priority` attribute to `ApplicationData`: this is needed when several applications are working in a same session, to know which one must be handled first. Will be used to make Remote Control have precedence over Call content. - XEP-0166: `_call_plugins` is now async and is not used with `DeferredList` anymore: the benefit to have methods called in parallels is very low, and it cause a lot of trouble as we can't predict order. Methods are now called sequentially so workflow can be predicted. - XEP-0167: fix `senders` XMPP attribute <=> SDP mapping - XEP-0234: preflight acceptance key is now `pre-accepted` instead of `file-accepted`, so the same key can be used with other jingle applications. - XEP-0167, XEP-0343: move some method to XEP-0167 - XEP-0353: use new `priority` feature to call preflight methods of applications according to it. - frontend (webrtc): refactor the sources/sink handling with a more flexible mechanism based on Pydantic models. It is now possible to have has many Data Channel as necessary, to have them in addition to A/V streams, to specify manually GStreamer sources and sinks, etc. - frontend (webrtc): rework of the pipeline to reduce latency. - frontend: new `portal_desktop` method. Screenshare portal handling has been moved there, and RemoteDesktop portal has been added. - frontend (webrtc): fix `extract_ufrag_pwd` method. rel 436
116 lines
3.3 KiB
Python
116 lines
3.3 KiB
Python
#!/usr/bin/env python3
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# Libervia WebRTC implementation
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# Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org)
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# This program is free software: you can redistribute it and/or modify
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# it under the terms of the GNU Affero General Public License as published by
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# the Free Software Foundation, either version 3 of the License, or
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# (at your option) any later version.
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# This program is distributed in the hope that it will be useful,
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# but WITHOUT ANY WARRANTY; without even the implied warranty of
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# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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# GNU Affero General Public License for more details.
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# You should have received a copy of the GNU Affero General Public License
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# along with this program. If not, see <http://www.gnu.org/licenses/>.
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from collections.abc import Awaitable
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from dataclasses import dataclass, field
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from typing import Any, Callable
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import uuid
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import gi
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gi.require_versions({"Gst": "1.0", "GstWebRTC": "1.0"})
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from gi.repository import GstWebRTC
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from pydantic import BaseModel, Field
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from libervia.frontends.tools import jid
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@dataclass
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class CallData:
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callee: jid.JID
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sid: str | None = None
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action_id: str | None = None
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kwargs: dict[str, Any] = field(default_factory=dict)
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class SourcesData(BaseModel):
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"""Data for Sources"""
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class SourcesNone(SourcesData):
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"""No source is used.
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This is used when the WebRTC connection will be used for data channels only."""
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class SourcesAuto(SourcesData):
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"""Automatic Sources (webcam/microphone)"""
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class SourcesTest(SourcesData):
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"""Test Sources (pattern)"""
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class SourcesDataChannel(SourcesData):
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"""Sources for transmitting data over Data Channel
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@param dc_open_cb: Called when Data Channel is open.
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This callback will be run in a GStreamer thread.
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"""
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name: str = Field(default_factory=lambda: str(uuid.uuid4()))
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dc_open_cb: Callable[[GstWebRTC.WebRTCDataChannel], None]
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class SourcesPipeline(SourcesData):
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"""Use custom pipeline description as a source.
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@param video_pipeline: pipeline description of video source.
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None to use automatic video source (same as SourcesAuto).
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Empty string to disable video.
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@param audio_pipeline: pipeline description of audio source.
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None to use automatic audio source (same as SourcesAuto).
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Empty string to disable audio.
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@param video_properties: Elements properties to set.
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@param audio_properties: Elements properties to set.
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"""
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video_pipeline: str|None = None
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audio_pipeline: str|None = None
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video_properties: dict = Field(default_factory=lambda: {})
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audio_properties: dict = Field(default_factory=lambda: {})
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class SinksData(BaseModel):
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"""Data for Sinks"""
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class SinksNone(SinksData):
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"""No sink is used.
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This is used when the WebRTC connection will be used for data channels only."""
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class SinksAuto(SinksData):
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"""Automatic Sinks (create windows/default audio)"""
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class SinksApp(SinksData):
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local_video_cb: Callable
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remote_video_cb: Callable | None
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class SinksDataChannel(SinksData):
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"""Sinks for transmitting data over Data Channel
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@param dc_on_data_channel: Called when Data Channel is created.
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This callback will be run in a GStreamer thread.
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"""
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dc_on_data_channel: (
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Callable[[GstWebRTC.WebRTCDataChannel], Awaitable[None]] | None
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) = None
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