rtp: Initial rtpbin2 element

Can receive and recevie one or more RTP sessions containing multiple
pt/ssrc combinations.

Demultiplexing happens internally instead of relying on separate
elements.

Co-Authored-By: François Laignel <francois@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>

Cargo.lock

@@ -2754,21 +2754,28 @@ dependencies = [
  "bitstream-io",
  "byte-slice-cast",
  "chrono",
+ "futures",
+ "gio",
  "gst-plugin-version-helper",
  "gstreamer",
  "gstreamer-app",
  "gstreamer-audio",
+ "gstreamer-base",
  "gstreamer-check",
+ "gstreamer-net",
  "gstreamer-rtp",
  "gstreamer-video",
  "hex",
+ "log",
  "once_cell",
  "rand",
+ "rtcp-types",
  "rtp-types",
  "slab",
  "smallvec",
  "thiserror",
  "time",
+ "tokio",
 ]
 
 [[package]]
@@ -5821,6 +5828,14 @@ dependencies = [
  "winapi",
 ]
 
+[[package]]
+name = "rtcp-types"
+version = "0.0.1"
+source = "git+https://github.com/ystreet/rtcp-types#f7fddfb87e9d7f4fed0b967fedc34995dd81ca86"
+dependencies = [
+ "thiserror",
+]
+
 [[package]]
 name = "rtp-types"
 version = "0.1.1"

docs/plugins/gst_plugins_cache.json

@@ -7191,6 +7191,92 @@
                 },
                 "rank": "marginal"
             },
+            "rtpbin2": {
+                "author": "Matthew Waters <matthew@centricular.com>",
+                "description": "RTP sessions management",
+                "hierarchy": [
+                    "GstRtpBin2",
+                    "GstElement",
+                    "GstObject",
+                    "GInitiallyUnowned",
+                    "GObject"
+                ],
+                "klass": "Network/RTP/Filter",
+                "pad-templates": {
+                    "rtcp_recv_sink_%%u": {
+                        "caps": "application/x-rtcp:\n",
+                        "direction": "sink",
+                        "presence": "request"
+                    },
+                    "rtcp_send_src_%%u": {
+                        "caps": "application/x-rtcp:\n",
+                        "direction": "src",
+                        "presence": "request"
+                    },
+                    "rtp_recv_sink_%%u": {
+                        "caps": "application/x-rtp:\n",
+                        "direction": "sink",
+                        "presence": "request"
+                    },
+                    "rtp_recv_src_%%u_%%u_%%u": {
+                        "caps": "application/x-rtp:\n",
+                        "direction": "src",
+                        "presence": "sometimes"
+                    },
+                    "rtp_send_sink_%%u": {
+                        "caps": "application/x-rtp:\n",
+                        "direction": "sink",
+                        "presence": "request"
+                    },
+                    "rtp_send_src_%%u": {
+                        "caps": "application/x-rtp:\n",
+                        "direction": "src",
+                        "presence": "sometimes"
+                    }
+                },
+                "properties": {
+                    "latency": {
+                        "blurb": "Amount of ms to buffer",
+                        "conditionally-available": false,
+                        "construct": false,
+                        "construct-only": false,
+                        "controllable": false,
+                        "default": "0",
+                        "max": "-1",
+                        "min": "0",
+                        "mutable": "ready",
+                        "readable": true,
+                        "type": "guint",
+                        "writable": true
+                    },
+                    "min-rtcp-interval": {
+                        "blurb": "Minimum time (in ms) between RTCP reports",
+                        "conditionally-available": false,
+                        "construct": false,
+                        "construct-only": false,
+                        "controllable": false,
+                        "default": "5000",
+                        "max": "-1",
+                        "min": "0",
+                        "mutable": "ready",
+                        "readable": true,
+                        "type": "guint",
+                        "writable": true
+                    },
+                    "stats": {
+                        "blurb": "Statistics about the session",
+                        "conditionally-available": false,
+                        "construct": false,
+                        "construct-only": false,
+                        "controllable": false,
+                        "mutable": "null",
+                        "readable": true,
+                        "type": "guint",
+                        "writable": false
+                    }
+                },
+                "rank": "none"
+            },
             "rtpgccbwe": {
                 "author": "Thibault Saunier <tsaunier@igalia.com>",
                 "description": "Estimates current network bandwidth using the Google Congestion Control algorithm notifying about it through the 'bitrate' property",

net/rtp/Cargo.toml

@@ -14,18 +14,26 @@ atomic_refcell = "0.1"
 bitstream-io = "2.1"
 byte-slice-cast = "1.2"
 chrono = { version = "0.4", default-features = false }
-gst     = { workspace = true, features = ["v1_20"] }
+gst       = { workspace = true, features = ["v1_20"] }
 gst-audio = { workspace = true, features = ["v1_20"] }
-gst-rtp = { workspace = true, features = ["v1_20"] }
+gst-base  = { workspace = true, features = ["v1_20"] }
+gst-net   = { workspace = true, features = ["v1_20"] }
+gst-rtp   = { workspace = true, features = ["v1_20"] }
 gst-video = { workspace = true, features = ["v1_20"] }
+futures = "0.3"
+gio.workspace = true
 hex = "0.4.3"
+log = "0.4"
 once_cell.workspace = true
 rand = { version = "0.8", default-features = false, features = ["std", "std_rng" ] }
 rtp-types = { version = "0.1" }
+rtcp-types = { git = "https://github.com/ystreet/rtcp-types", version = "0.0" }
 slab = "0.4.9"
 smallvec = { version = "1.11", features = ["union", "write", "const_generics", "const_new"] }
 thiserror = "1"
 time = { version = "0.3", default-features = false, features = ["std"] }
+# TODO: experiment with other async executors (mio, async-std, etc)
+tokio = { version = "1", default-features = false, features = ["rt-multi-thread", "time"] }
 
 [dev-dependencies]
 gst-check = { workspace = true, features = ["v1_20"] }
@@ -56,4 +64,4 @@ versioning = false
 import_library = false
 
 [package.metadata.capi.pkg_config]
-requires_private = "gstreamer-1.0, gstreamer-base-1.0, gstreamer-rtp-1.0, gobject-2.0, glib-2.0, gmodule-2.0"
+requires_private = "gstreamer-1.0, gstreamer-base-1.0, gstreamer-rtp-1.0, gstreamer-net-1.0, gobject-2.0, glib-2.0, gmodule-2.0, gio-2.0"

net/rtp/src/lib.rs

@@ -14,12 +14,17 @@
  *
  * Since: plugins-rs-0.9.0
  */
+
+#[macro_use]
+extern crate log;
+
 use gst::glib;
 
 #[macro_use]
 mod utils;
 
 mod gcc;
+mod rtpbin2;
 
 mod audio_discont;
 mod baseaudiopay;
@@ -42,6 +47,7 @@ mod tests;
 
 fn plugin_init(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
     gcc::register(plugin)?;
+    rtpbin2::register(plugin)?;
 
     #[cfg(feature = "doc")]
     {

net/rtp/src/rtpbin2/mod.rs

@@ -0,0 +1,30 @@
+// SPDX-License-Identifier: MPL-2.0
+
+use gst::glib;
+use gst::prelude::*;
+use once_cell::sync::Lazy;
+mod imp;
+mod session;
+mod source;
+mod time;
+
+glib::wrapper! {
+    pub struct RtpBin2(ObjectSubclass<imp::RtpBin2>) @extends gst::Element, gst::Object;
+}
+
+pub fn register(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
+    gst::Element::register(
+        Some(plugin),
+        "rtpbin2",
+        gst::Rank::NONE,
+        RtpBin2::static_type(),
+    )
+}
+
+pub static RUNTIME: Lazy<tokio::runtime::Runtime> = Lazy::new(|| {
+    tokio::runtime::Builder::new_multi_thread()
+        .enable_time()
+        .worker_threads(1)
+        .build()
+        .unwrap()
+});

net/rtp/src/rtpbin2/time.rs

@@ -0,0 +1,58 @@
+// SPDX-License-Identifier: MPL-2.0
+
+use std::{
+    ops::{Add, Sub},
+    time::{Duration, SystemTime},
+};
+
+// time between the NTP time at 1900-01-01 and the unix EPOCH (1970-01-01)
+const NTP_OFFSET: Duration = Duration::from_secs((365 * 70 + 17) * 24 * 60 * 60);
+
+// 2^32
+const F32: f64 = 4_294_967_296.0;
+
+#[derive(Clone, Copy, Debug, PartialEq, Eq, PartialOrd, Ord)]
+pub struct NtpTime(u64);
+
+impl NtpTime {
+    pub fn from_duration(dur: Duration) -> Self {
+        Self((dur.as_secs_f64() * F32) as u64)
+    }
+
+    pub fn as_u32(self) -> u32 {
+        ((self.0 >> 16) & 0xffffffff) as u32
+    }
+
+    pub fn as_u64(self) -> u64 {
+        self.0
+    }
+}
+
+impl Sub for NtpTime {
+    type Output = NtpTime;
+    fn sub(self, rhs: Self) -> Self::Output {
+        NtpTime(self.0 - rhs.0)
+    }
+}
+
+impl Add for NtpTime {
+    type Output = NtpTime;
+    fn add(self, rhs: Self) -> Self::Output {
+        NtpTime(self.0 + rhs.0)
+    }
+}
+
+pub fn system_time_to_ntp_time_u64(time: SystemTime) -> NtpTime {
+    let dur = time
+        .duration_since(SystemTime::UNIX_EPOCH)
+        .expect("time is before unix epoch?!")
+        + NTP_OFFSET;
+
+    NtpTime::from_duration(dur)
+}
+
+impl From<u64> for NtpTime {
+    fn from(value: u64) -> Self {
+        NtpTime(value)
+    }
+}

net/rtp/tests/rtpbin2.rs

@@ -0,0 +1,159 @@
+//
+// Copyright (C) 2023 Matthew Waters <matthew@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 std::sync::{Arc, Mutex};
+
+use gst::{prelude::*, Caps};
+use gst_check::Harness;
+use rtp_types::*;
+
+fn init() {
+    use std::sync::Once;
+    static INIT: Once = Once::new();
+
+    INIT.call_once(|| {
+        gst::init().unwrap();
+        gstrsrtp::plugin_register_static().expect("rtpbin2 test");
+    });
+}
+
+const TEST_SSRC: u32 = 0x12345678;
+const TEST_PT: u8 = 96;
+const TEST_CLOCK_RATE: u32 = 48000;
+
+fn generate_rtp_buffer(seqno: u16, rtpts: u32, payload_len: usize) -> gst::Buffer {
+    let payload = vec![4; payload_len];
+    let packet = RtpPacketBuilder::new()
+        .ssrc(TEST_SSRC)
+        .payload_type(TEST_PT)
+        .sequence_number(seqno)
+        .timestamp(rtpts)
+        .payload(&payload);
+    let size = packet.calculate_size().unwrap();
+    let mut data = vec![0; size];
+    packet.write_into(&mut data).unwrap();
+    gst::Buffer::from_mut_slice(data)
+}
+
+#[test]
+fn test_send() {
+    init();
+
+    let mut h = Harness::with_padnames("rtpbin2", Some("rtp_send_sink_0"), Some("rtp_send_src_0"));
+    h.play();
+
+    let caps = Caps::builder("application/x-rtp")
+        .field("media", "audio")
+        .field("payload", TEST_PT as i32)
+        .field("clock-rate", TEST_CLOCK_RATE as i32)
+        .field("encoding-name", "custom-test")
+        .build();
+    h.set_src_caps(caps);
+
+    h.push(generate_rtp_buffer(500, 20, 9)).unwrap();
+    h.push(generate_rtp_buffer(501, 30, 11)).unwrap();
+
+    let buffer = h.pull().unwrap();
+    let mapped = buffer.map_readable().unwrap();
+    let rtp = rtp_types::RtpPacket::parse(&mapped).unwrap();
+    assert_eq!(rtp.sequence_number(), 500);
+
+    let buffer = h.pull().unwrap();
+    let mapped = buffer.map_readable().unwrap();
+    let rtp = rtp_types::RtpPacket::parse(&mapped).unwrap();
+    assert_eq!(rtp.sequence_number(), 501);
+
+    let stats = h.element().unwrap().property::<gst::Structure>("stats");
+    let session_stats = stats.get::<gst::Structure>("0").unwrap();
+    let source_stats = session_stats
+        .get::<gst::Structure>(TEST_SSRC.to_string())
+        .unwrap();
+    assert_eq!(source_stats.get::<u32>("ssrc").unwrap(), TEST_SSRC);
+    assert_eq!(
+        source_stats.get::<u32>("clock-rate").unwrap(),
+        TEST_CLOCK_RATE
+    );
+    assert!(source_stats.get::<bool>("sender").unwrap());
+    assert!(source_stats.get::<bool>("local").unwrap());
+    assert_eq!(source_stats.get::<u64>("packets-sent").unwrap(), 2);
+    assert_eq!(source_stats.get::<u64>("octets-sent").unwrap(), 20);
+}
+
+#[test]
+fn test_receive() {
+    init();
+
+    let h = Arc::new(Mutex::new(Harness::with_padnames(
+        "rtpbin2",
+        Some("rtp_recv_sink_0"),
+        None,
+    )));
+    let weak_h = Arc::downgrade(&h);
+    let mut inner = h.lock().unwrap();
+    inner
+        .element()
+        .unwrap()
+        .connect_pad_added(move |_elem, pad| {
+            weak_h
+                .upgrade()
+                .unwrap()
+                .lock()
+                .unwrap()
+                .add_element_src_pad(pad)
+        });
+    inner.play();
+
+    let caps = Caps::builder("application/x-rtp")
+        .field("media", "audio")
+        .field("payload", TEST_PT as i32)
+        .field("clock-rate", TEST_CLOCK_RATE as i32)
+        .field("encoding-name", "custom-test")
+        .build();
+    inner.set_src_caps(caps);
+
+    // Cannot push with harness lock as the 'pad-added' handler needs to add the newly created pad to
+    // the harness and needs to also take the harness lock.  Workaround by pushing from the
+    // internal harness pad directly.
+    let push_pad = inner
+        .element()
+        .unwrap()
+        .static_pad("rtp_recv_sink_0")
+        .unwrap()
+        .peer()
+        .unwrap();
+    drop(inner);
+    push_pad.push(generate_rtp_buffer(500, 20, 9)).unwrap();
+    push_pad.push(generate_rtp_buffer(501, 30, 11)).unwrap();
+    let mut inner = h.lock().unwrap();
+
+    let buffer = inner.pull().unwrap();
+    let mapped = buffer.map_readable().unwrap();
+    let rtp = rtp_types::RtpPacket::parse(&mapped).unwrap();
+    assert_eq!(rtp.sequence_number(), 500);
+
+    let buffer = inner.pull().unwrap();
+    let mapped = buffer.map_readable().unwrap();
+    let rtp = rtp_types::RtpPacket::parse(&mapped).unwrap();
+    assert_eq!(rtp.sequence_number(), 501);
+
+    let stats = inner.element().unwrap().property::<gst::Structure>("stats");
+    let session_stats = stats.get::<gst::Structure>("0").unwrap();
+    let source_stats = session_stats
+        .get::<gst::Structure>(TEST_SSRC.to_string())
+        .unwrap();
+    assert_eq!(source_stats.get::<u32>("ssrc").unwrap(), TEST_SSRC);
+    assert_eq!(
+        source_stats.get::<u32>("clock-rate").unwrap(),
+        TEST_CLOCK_RATE
+    );
+    assert!(source_stats.get::<bool>("sender").unwrap());
+    assert!(!source_stats.get::<bool>("local").unwrap());
+    assert_eq!(source_stats.get::<u64>("packets-received").unwrap(), 2);
+    assert_eq!(source_stats.get::<u64>("octets-received").unwrap(), 20);
+}