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rtp: Add MPEG-TS RTP depayloader

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Can handle different packet sizes, also see:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1310

Has clock-rate=90000 as spec prescribes, see:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/691

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
This commit is contained in:
Tim-Philipp Müller 2023-10-21 16:55:03 +01:00 committed by GStreamer Marge Bot
parent f4366f8b2e
commit 9f07ec35e6
5 changed files with 392 additions and 0 deletions
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42
docs/plugins/gst_plugins_cache.json
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@ -6374,6 +6374,48 @@
}, },
"rank": "none" "rank": "none"
}, },
"rtpmp2tdepay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Depayload an MPEG Transport Stream from RTP packets (RFC 2250)",
"hierarchy": [
"GstRtpMP2TDepay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: MP2T\napplication/x-rtp:\n media: video\n payload: 33\n clock-rate: 90000\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/mpegts:\n packetsize: { (int)188, (int)192, (int)204, (int)208 }\n systemstream: true\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"skip-first-bytes": {
"blurb": "Number of bytes to skip at the beginning of the payload",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
}
},
"rank": "marginal"
},
"rtppcmadepay2": { "rtppcmadepay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>", "author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Depayload A-law from RTP packets (RFC 3551)", "description": "Depayload A-law from RTP packets (RFC 3551)",

3
net/rtp/src/lib.rs
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@ -24,6 +24,7 @@ mod basedepay;
mod basepay; mod basepay;
mod av1; mod av1;
mod mp2t;
mod pcmau; mod pcmau;
#[cfg(test)] #[cfg(test)]
@ -47,6 +48,8 @@ fn plugin_init(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
av1::depay::register(plugin)?; av1::depay::register(plugin)?;
av1::pay::register(plugin)?; av1::pay::register(plugin)?;
mp2t::depay::register(plugin)?;
pcmau::depay::register(plugin)?; pcmau::depay::register(plugin)?;
pcmau::pay::register(plugin)?; pcmau::pay::register(plugin)?;

316
net/rtp/src/mp2t/depay/imp.rs Normal file
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@ -0,0 +1,316 @@
// GStreamer RTP MPEG-TS Depayloader
//
// Copyright (C) 2023-2024 Tim-Philipp Müller <tim centricular com>
//
// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
// If a copy of the MPL was not distributed with this file, You can obtain one at
// <https://mozilla.org/MPL/2.0/>.
//
// SPDX-License-Identifier: MPL-2.0
/**
* SECTION:element-rtpmp2tdepay2
* @see_also: rtpmp2tpay2, rtpmp2tdepay, rtpmp2tpay, tsdemux, mpegtsmux
*
* Depayload an MPEG Transport Stream from RTP packets as per [RFC 2250][rfc-2250].
*
* [rfc-2250]: https://www.rfc-editor.org/rfc/rfc2250.html
*
* ## Example pipeline
*
* |[
* gst-launch-1.0 udpsrc address=127.0.0.1 port=5555 caps='application/x-rtp,media=video,clock-rate=90000,encoding-name=MP2T' ! rtpjitterbuffer latency=100 ! rtpmp2tdepay2 ! decodebin3 ! videoconvertscale ! autovideosink
* ]| This will depayload an incoming RTP MPEG-TS stream. You can use the #rtpmp2tpay2 or #rtpmp2tpay
* element to create such an RTP stream.
*
* Since: plugins-rs-0.13.0
*/
use atomic_refcell::AtomicRefCell;
use gst::{glib, prelude::*, subclass::prelude::*};
use once_cell::sync::Lazy;
use std::num::NonZeroUsize;
use std::sync::Mutex;
use crate::basedepay::RtpBaseDepay2Ext;
const TS_PACKET_SYNC: u8 = 0x47;
#[derive(Default)]
pub struct RtpMP2TDepay {
state: AtomicRefCell<State>,
settings: Mutex<Settings>,
}
#[derive(Default)]
struct State {
packet_size: Option<NonZeroUsize>,
bytes_to_skip: usize,
}
#[derive(Debug, Clone)]
struct Settings {
skip_first_bytes: u32,
}
const DEFAULT_SKIP_FIRST_BYTES: u32 = 0;
impl Default for Settings {
fn default() -> Self {
Settings {
skip_first_bytes: DEFAULT_SKIP_FIRST_BYTES,
}
}
}
static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
gst::DebugCategory::new(
"rtpmp2tdepay2",
gst::DebugColorFlags::empty(),
Some("RTP MPEG-TS Depayloader"),
)
});
#[glib::object_subclass]
impl ObjectSubclass for RtpMP2TDepay {
const NAME: &'static str = "GstRtpMP2TDepay2";
type Type = super::RtpMP2TDepay;
type ParentType = crate::basedepay::RtpBaseDepay2;
}
impl ObjectImpl for RtpMP2TDepay {
fn properties() -> &'static [glib::ParamSpec] {
static PROPERTIES: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
vec![glib::ParamSpecUInt::builder("skip-first-bytes")
.nick("Skip first bytes")
.blurb("Number of bytes to skip at the beginning of the payload")
.default_value(DEFAULT_SKIP_FIRST_BYTES)
.mutable_ready()
.build()]
});
PROPERTIES.as_ref()
}
fn set_property(&self, _id: usize, value: &glib::Value, pspec: &glib::ParamSpec) {
match pspec.name() {
"skip-first-bytes" => {
let mut settings = self.settings.lock().unwrap();
settings.skip_first_bytes = value.get().expect("type checked upstream");
}
name => unimplemented!("Property '{name}'"),
};
}
fn property(&self, _id: usize, pspec: &glib::ParamSpec) -> glib::Value {
match pspec.name() {
"skip-first-bytes" => {
let settings = self.settings.lock().unwrap();
settings.skip_first_bytes.to_value()
}
name => unimplemented!("Property '{name}'"),
}
}
}
impl GstObjectImpl for RtpMP2TDepay {}
impl ElementImpl for RtpMP2TDepay {
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
gst::subclass::ElementMetadata::new(
"RTP MPEG-TS Depayloader",
"Codec/Depayloader/Network/RTP",
"Depayload an MPEG Transport Stream from RTP packets (RFC 2250)",
"Tim-Philipp Müller <tim centricular com>",
)
});
Some(&*ELEMENT_METADATA)
}
fn pad_templates() -> &'static [gst::PadTemplate] {
static PAD_TEMPLATES: Lazy<Vec<gst::PadTemplate>> = Lazy::new(|| {
let sink_pad_template = gst::PadTemplate::new(
"sink",
gst::PadDirection::Sink,
gst::PadPresence::Always,
&gst::Caps::builder_full()
.structure(
// Note: C depayloader accepts MP2T-ES as well but that was just for
// backward compatibility because the GStreamer 0.10 payloader used
// to (wrongly) produce that at some point a long time ago.
// Also spec (and common sense) say clock-rate should always be 90000
// (C depayloader accepts any clock rate in caps), see
// https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/691
gst::Structure::builder("application/x-rtp")
.field("media", "video")
.field("clock-rate", 90000i32)
.field("encoding-name", "MP2T")
.build(),
)
.structure(
gst::Structure::builder("application/x-rtp")
.field("media", "video")
.field("payload", 33i32)
.field("clock-rate", 90000i32)
.build(),
)
.build(),
)
.unwrap();
let src_pad_template = gst::PadTemplate::new(
"src",
gst::PadDirection::Src,
gst::PadPresence::Always,
&gst::Caps::builder("video/mpegts")
.field("packetsize", gst::List::new([188i32, 192, 204, 208]))
.field("systemstream", true)
.build(),
)
.unwrap();
vec![src_pad_template, sink_pad_template]
});
PAD_TEMPLATES.as_ref()
}
}
impl crate::basedepay::RtpBaseDepay2Impl for RtpMP2TDepay {
fn start(&self) -> Result<(), gst::ErrorMessage> {
let settings = self.settings.lock().unwrap();
// Copy skip bytes into state so we don't have to take the settings lock all the time
*self.state.borrow_mut() = State {
packet_size: None,
bytes_to_skip: settings.skip_first_bytes as usize,
};
Ok(())
}
fn stop(&self) -> Result<(), gst::ErrorMessage> {
*self.state.borrow_mut() = State::default();
Ok(())
}
// Encapsulation of MPEG System and Transport Streams:
// https://www.rfc-editor.org/rfc/rfc2250.html#section-2
//
fn handle_packet(
&self,
packet: &crate::basedepay::Packet,
) -> Result<gst::FlowSuccess, gst::FlowError> {
let mut state = self.state.borrow_mut();
let bytes_to_skip = state.bytes_to_skip;
let payload = packet.payload();
if payload.len() < 188 + bytes_to_skip {
gst::warning!(CAT, imp: self,
"Payload too small: {} bytes, but need at least {} bytes",
payload.len(), 188 + bytes_to_skip
);
self.obj().drop_packet(packet);
return Ok(gst::FlowSuccess::Ok);
}
let (_, payload) = payload.split_at(bytes_to_skip);
if state.packet_size.is_none() {
state.packet_size = self.detect_packet_size(payload);
if let Some(packet_size) = state.packet_size {
let src_caps = gst::Caps::builder("video/mpegts")
.field("packetsize", packet_size.get() as i32)
.field("systemstream", true)
.build();
self.obj().set_src_caps(&src_caps);
}
}
let Some(packet_size) = state.packet_size else {
gst::debug!(CAT, imp: self, "Could not determine packet size, dropping packet {packet:?}");
self.obj().drop_packet(packet);
return Ok(gst::FlowSuccess::Ok);
};
let packet_size = packet_size.get();
// For MPEG2 Transport Streams the RTP payload will contain an integral
// number of MPEG transport packets.
let n_packets = payload.len() / packet_size;
if payload.len() % packet_size != 0 {
gst::warning!(CAT, imp: self,
"Payload does not contain an integral number of MPEG-TS packets! ({} left over)",
payload.len() % packet_size);
}
let output_size = n_packets * packet_size;
gst::trace!(CAT, imp: self, "Packet with {n_packets} MPEG-TS packets of size {packet_size}");
let mut buffer =
packet.payload_subbuffer_from_offset_with_length(bytes_to_skip, output_size);
// Marker flag indicates MPEG-TS timestamping discontinuity
if packet.marker_bit() {
let buffer_ref = buffer.get_mut().unwrap();
buffer_ref.set_flags(gst::BufferFlags::RESYNC);
}
gst::trace!(CAT, imp: self, "Finishing buffer {buffer:?}");
self.obj().queue_buffer(packet.into(), buffer)
}
}
impl RtpMP2TDepay {
fn detect_packet_size(&self, payload: &[u8]) -> Option<NonZeroUsize> {
const PACKET_SIZES: [(usize, usize); 4] = [(188, 0), (192, 4), (204, 0), (208, 0)];
for (size, offset) in PACKET_SIZES {
gst::debug!(CAT, imp: self, "Trying MPEG-TS packet size of {size} bytes..");
// Try exact size match for the payload first
if payload.len() >= size
&& payload.len() % size == 0
&& payload
.chunks_exact(size)
.all(|packet| packet[offset] == TS_PACKET_SYNC)
{
gst::info!(CAT, imp: self, "Detected MPEG-TS packet size of {size} bytes, {} packets", payload.len() / size);
return NonZeroUsize::new(size);
}
}
gst::warning!(CAT, imp: self, "Could not detect MPEG-TS packet size using full payload");
// No match? Try if we find a size if we ignore any leftover bytes
for (size, offset) in PACKET_SIZES {
gst::debug!(CAT, imp: self, "Trying MPEG-TS packet size of {size} bytes with remainder..");
if payload.len() >= size
&& payload.len() % size != 0
&& payload
.chunks_exact(size)
.all(|packet| packet[offset] == TS_PACKET_SYNC)
{
gst::info!(CAT, imp: self, "Detected MPEG-TS packet size of {size} bytes, {} packets, {} bytes leftover",
payload.len() / size, payload.len() % size);
return NonZeroUsize::new(size);
}
}
gst::warning!(CAT, imp: self, "Could not detect MPEG-TS packet size");
None
}
}

28
net/rtp/src/mp2t/depay/mod.rs Normal file
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@ -0,0 +1,28 @@
// GStreamer RTP MPEG-TS Depayloader
//
// Copyright (C) 2023-2024 Tim-Philipp Müller <tim centricular com>
//
// This Source Code Form is subject to the terms of the Mozilla Public License, v2.0.
// If a copy of the MPL was not distributed with this file, You can obtain one at
// <https://mozilla.org/MPL/2.0/>.
//
// SPDX-License-Identifier: MPL-2.0
use gst::glib;
use gst::prelude::*;
pub mod imp;
glib::wrapper! {
pub struct RtpMP2TDepay(ObjectSubclass<imp::RtpMP2TDepay>)
@extends crate::basedepay::RtpBaseDepay2, gst::Element, gst::Object;
}
pub fn register(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
gst::Element::register(
Some(plugin),
"rtpmp2tdepay2",
gst::Rank::MARGINAL,
RtpMP2TDepay::static_type(),
)
}

3
net/rtp/src/mp2t/mod.rs Normal file
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@ -0,0 +1,3 @@
// SPDX-License-Identifier: MPL-2.0
pub mod depay;