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GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI] Features: * Live streaming N audio and N video - With RTCP-based A/V sync * Lower transports: TCP, UDP, UDP-Multicast * RTP, RTCP SR, RTCP RR * OPTIONS DESCRIBE SETUP PLAY TEARDOWN * Custom UDP socket management, does not use udpsrc/udpsink * Supports both rtpbin and the rtpbin2 rust rewrite - Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs) * Properties: - protocols selection and priority (NEW!) - location supports rtsp[ut]:// - port-start instead of port-range Co-Authored-by: Tim-Philipp Müller <tim@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
31 lines
755 B
Rust
31 lines
755 B
Rust
// GStreamer RTSP Source v2
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//
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// Copyright (C) 2023 Tim-Philipp Müller <tim centricular com>
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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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mod body;
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mod imp;
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mod tcp_message;
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mod transport;
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glib::wrapper! {
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pub struct RtspSrc(ObjectSubclass<imp::RtspSrc>) @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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"rtspsrc2",
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gst::Rank::NONE,
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RtspSrc::static_type(),
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)
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}
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