@tomoxv/gstwebrtc-api
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Javascript API to integrate GStreamer WebRTC streams (webrtcsrc/webrtcsink) in a web browser
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JavaScript
/*
* gstwebrtc-api
*
* Copyright (C) 2022 Igalia S.L. <info@igalia.com>
* Author: Loïc Le Page <llepage@igalia.com>
*
* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/.
*/
/**
* Session states enumeration.<br>
* Session state always increases from idle to closed and never switches backwards.
* @typedef {enum} GstWebRTCAPI.SessionState
* @readonly
* @property {0} idle - Default state when creating a new session, goes to <i>connecting</i> when starting
* the session.
* @property {1} connecting - Session is trying to connect to remote peers, goes to <i>streaming</i> in case of
* success or <i>closed</i> in case of error.
* @property {2} streaming - Session is correctly connected to remote peers and currently streaming audio/video, goes
* to <i>closed</i> when any peer closes the session.
* @property {3} closed - Session is closed and can be garbage collected, state will not change anymore.
*/
const SessionState = Object.freeze({
idle: 0,
connecting: 1,
streaming: 2,
closed: 3
});
export { SessionState as default };