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https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
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67e9ba8286
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
27 lines
731 B
Rust
27 lines
731 B
Rust
// Copyright (C) 2022, Asymptotic Inc.
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// Author: Taruntej Kanakamalla <taruntej@asymptotic.io>
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//
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// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
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// If a copy of the MPL was not distributed with this file, You can obtain one at
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// <https://mozilla.org/MPL/2.0/>.
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//
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// SPDX-License-Identifier: MPL-2.0
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use gst::glib;
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use gst::prelude::*;
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pub mod imp;
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glib::wrapper! {
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pub struct WhipSink(ObjectSubclass<imp::WhipSink>) @extends gst::Bin, gst::Element, gst::Object;
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}
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pub fn register(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
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gst::Element::register(
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Some(plugin),
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"whipsink",
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gst::Rank::Marginal,
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WhipSink::static_type(),
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)
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}
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