view libervia/frontends/tools/webrtc_models.py @ 4259:49019947cc76

component AP Gateway: implement HTTP GET signature.
author Goffi <goffi@goffi.org>
date Wed, 05 Jun 2024 22:34:09 +0200
parents 79c8a70e1813
children 0d7bb4df2343
line wrap: on
line source

#!/usr/bin/env python3

# Libervia WebRTC implementation
# Copyright (C) 2009-2023 Jérôme Poisson (goffi@goffi.org)

# This program is free software: you can redistribute it and/or modify
# it under the terms of the GNU Affero General Public License as published by
# the Free Software Foundation, either version 3 of the License, or
# (at your option) any later version.

# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
# GNU Affero General Public License for more details.

# You should have received a copy of the GNU Affero General Public License
# along with this program.  If not, see <http://www.gnu.org/licenses/>.

from collections.abc import Awaitable
from dataclasses import dataclass, field
from typing import Any, Callable
import uuid
import gi

gi.require_versions({"Gst": "1.0", "GstWebRTC": "1.0"})
from gi.repository import GstWebRTC
from pydantic import BaseModel, Field

from libervia.frontends.tools import jid


@dataclass
class CallData:
    callee: jid.JID
    sid: str | None = None
    action_id: str | None = None
    kwargs: dict[str, Any] = field(default_factory=dict)


class SourcesData(BaseModel):
    """Data for Sources"""


class SourcesNone(SourcesData):
    """No source is used.

    This is used when the WebRTC connection will be used for data channels only."""


class SourcesAuto(SourcesData):
    """Automatic Sources (webcam/microphone)"""


class SourcesTest(SourcesData):
    """Test Sources (pattern)"""


class SourcesDataChannel(SourcesData):
    """Sources for transmitting data over Data Channel


    @param dc_open_cb: Called when Data Channel is open.
        This callback will be run in a GStreamer thread.
    """
    name: str = Field(default_factory=lambda: str(uuid.uuid4()))
    dc_open_cb: Callable[[GstWebRTC.WebRTCDataChannel], None]


class SourcesPipeline(SourcesData):
    """Use custom pipeline description as a source.

    @param video_pipeline: pipeline description of video source.
        None to use automatic video source (same as SourcesAuto).
        Empty string to disable video.
    @param audio_pipeline: pipeline description of audio source.
        None to use automatic audio source (same as SourcesAuto).
        Empty string to disable audio.
    @param video_properties: Elements properties to set.
    @param audio_properties: Elements properties to set.

    """
    video_pipeline: str|None = None
    audio_pipeline: str|None = None
    video_properties: dict = Field(default_factory=lambda: {})
    audio_properties: dict = Field(default_factory=lambda: {})


class SinksData(BaseModel):
    """Data for Sinks"""


class SinksNone(SinksData):
    """No sink is used.

    This is used when the WebRTC connection will be used for data channels only."""


class SinksAuto(SinksData):
    """Automatic Sinks (create windows/default audio)"""


class SinksApp(SinksData):
    local_video_cb: Callable
    remote_video_cb: Callable | None


class SinksDataChannel(SinksData):
    """Sinks for transmitting data over Data Channel

    @param dc_on_data_channel: Called when Data Channel is created.
        This callback will be run in a GStreamer thread.
    """

    dc_on_data_channel: (
        Callable[[GstWebRTC.WebRTCDataChannel], Awaitable[None]] | None
    ) = None