Update version to 0.12.3

Update CHANGELOG.md for 0.12.3
Update Cargo.lock
2024-06-08 17:29:22 +00:00 · 2024-03-21 18:53:12 +02:00 · 2024-03-21 18:47:31 +02:00 · 2024-03-21 13:30:20 +02:00 · 2024-03-21 13:30:20 +02:00 · 2024-03-21 13:30:20 +02:00
44 changed files with 1978 additions and 1018 deletions
--- a/.gitignore
+++ b/.gitignore
@ -1,4 +1,3 @@
-Cargo.lock
 target
 *~
 *.bk
--- a/.gitlab-ci.yml
+++ b/.gitlab-ci.yml
@ -6,7 +6,7 @@ include:
    file: '/templates/debian.yml'

  - project: 'gstreamer/gstreamer-rs'
-    ref: main
+    ref: '0.22'
    file: '/ci/images_template.yml'

  - project: 'gstreamer/gstreamer'
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@ -5,6 +5,47 @@ The format is based on [Keep a Changelog](http://keepachangelog.com/en/1.0.0/)
 and this project adheres to [Semantic Versioning](http://semver.org/spec/v2.0.0.html),
 specifically the [variant used by Rust](http://doc.crates.io/manifest.html#the-version-field).

+## [0.12.3] - 2024-03-21
+### Fixed
+- gtk4paintablesink: Fix scaling of texture position.
+- janusvrwebrtcsink: Handle 64 bit numerical room ids.
+- janusvrwebrtcsink: Don't include deprecated audio/video fields in publish
+  messages.
+- janusvrwebrtcsink: Handle various other messages to avoid printing errors.
+- livekitwebrtc: Fix shutdown behaviour.
+- rtpgccbwe: Don't forward buffer lists with buffers from different SSRCs to
+  avoid breaking assumptions in rtpsession.
+- sccparse: Ignore invalid timecodes during seeking.
+- webrtcsink: Don't try parsing audio caps as video caps.
+
+### Changed
+- webrtc: Allow resolution and framerate changes.
+- webrtcsrc: Make produce-peer-id optional.
+
+### Added
+- livekitwebrtcsrc: Add new LiveKit source element.
+- regex: Add support for configuring regex behaviour.
+- spotifyaudiosrc: Document how to use with non-Facebook accounts.
+- webrtcsrc: Add `do-retransmission` property.
+
+## [0.12.2] - 2024-02-26
+### Fixed
+- rtpgccbwe: Don't reset PTS/DTS to `None` as otherwise `rtpsession` won't be
+  able to generate valid RTCP.
+- webrtcsink: Fix usage with 1.22.
+
+### Added
+- janusvrwebrtcsink: Add `secret-key` property.
+- janusvrwebrtcsink: Allow for string room ids and add `string-ids` property.
+- textwrap: Don't split on all whitespaces, especially not on non-breaking
+  whitespace.
+
+## [0.12.1] - 2024-02-13
+### Added
+- gtk4: Create a window for testing purposes when running in `gst-launch-1.0`
+  or if `GST_GTK4_WINDOW=1` is set.
+- webrtcsink: Add `msid` property.
+
 ## [0.12.0] - 2024-02-08
 ### Changed
 - ndi: `ndisrc` passes received data downstream without an additional copy, if
@ -36,7 +77,6 @@ specifically the [variant used by Rust](http://doc.crates.io/manifest.html#the-v
 - New `janusvrwebrtcsink` element for the Janus VideoRoom API.
 - New `rtspsrc2` element.
 - New `whipserversrc` element.
-
 - gtk4: New `background-color` property for setting the color of the
  background of the frame and the borders, if any.
 - gtk4: New `scale-filter` property for defining how to scale the frames.
@ -344,7 +384,10 @@ specifically the [variant used by Rust](http://doc.crates.io/manifest.html#the-v
 - webrtcsink: Make the `turn-server` property a `turn-servers` list
 - webrtcsink: Move from async-std to tokio

-[Unreleased]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.0...HEAD
+[Unreleased]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.3...HEAD
+[0.12.3]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.2...0.12.3
+[0.12.2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.1...0.12.2
+[0.12.1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.0...0.12.1
 [0.12.0]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.11.3...0.12.0
 [0.11.3]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.11.2...0.11.3
 [0.11.2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.11.1...0.11.2
--- a/Cargo.lock
+++ b/Cargo.lock
--- a/Cargo.toml
+++ b/Cargo.toml
@ -113,36 +113,36 @@ panic = 'unwind'
 opt-level = 1

 [workspace.package]
-version = "0.12.0-alpha.1"
+version = "0.12.3"
 repository = "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
 edition = "2021"
 rust-version = "1.70"

 [workspace.dependencies]
 once_cell = "1"
-glib = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "master" }
-gio = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "master" }
-cairo-rs = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "master", features=["use_glib"] }
-pango = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "master" }
-pangocairo = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "master" }
-gtk = { package = "gtk4", git = "https://github.com/gtk-rs/gtk4-rs", branch = "master"}
-gdk-wayland = { package = "gdk4-wayland", git = "https://github.com/gtk-rs/gtk4-rs", branch = "master"}
-gdk-x11 = { package = "gdk4-x11", git = "https://github.com/gtk-rs/gtk4-rs", branch = "master"}
-gdk-win32 = { package = "gdk4-win32", git = "https://github.com/gtk-rs/gtk4-rs", branch = "master"}
-gst = { package = "gstreamer", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-app = { package = "gstreamer-app", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-audio = { package = "gstreamer-audio", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-base = { package = "gstreamer-base", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-check = { package = "gstreamer-check", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-gl = { package = "gstreamer-gl", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-gl-egl = { package = "gstreamer-gl-egl", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-gl-wayland = { package = "gstreamer-gl-wayland", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-gl-x11 = { package = "gstreamer-gl-x11", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-net = { package = "gstreamer-net", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-pbutils = { package = "gstreamer-pbutils", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-plugin-version-helper = { path="./version-helper" }
-gst-rtp = { package = "gstreamer-rtp", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-sdp = { package = "gstreamer-sdp", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-utils = { package = "gstreamer-utils", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-video = { package = "gstreamer-video", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
-gst-webrtc = { package = "gstreamer-webrtc", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "main" }
+glib = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "0.19", version = "0.19" }
+gio = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "0.19", version = "0.19" }
+cairo-rs = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "0.19", version = "0.19", features=["use_glib"] }
+pango = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "0.19", version = "0.19" }
+pangocairo = { git = "https://github.com/gtk-rs/gtk-rs-core", branch = "0.19", version = "0.19" }
+gtk = { package = "gtk4", git = "https://github.com/gtk-rs/gtk4-rs", branch = "0.8", version = "0.8"}
+gdk-wayland = { package = "gdk4-wayland", git = "https://github.com/gtk-rs/gtk4-rs", branch = "0.8", version = "0.8"}
+gdk-x11 = { package = "gdk4-x11", git = "https://github.com/gtk-rs/gtk4-rs", branch = "0.8", version = "0.8"}
+gdk-win32 = { package = "gdk4-win32", git = "https://github.com/gtk-rs/gtk4-rs", branch = "0.8", version = "0.8"}
+gst = { package = "gstreamer", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-app = { package = "gstreamer-app", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-audio = { package = "gstreamer-audio", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-base = { package = "gstreamer-base", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-check = { package = "gstreamer-check", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-gl = { package = "gstreamer-gl", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-gl-egl = { package = "gstreamer-gl-egl", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-gl-wayland = { package = "gstreamer-gl-wayland", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-gl-x11 = { package = "gstreamer-gl-x11", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-net = { package = "gstreamer-net", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-pbutils = { package = "gstreamer-pbutils", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-plugin-version-helper = { path="./version-helper", version = "0.8" }
+gst-rtp = { package = "gstreamer-rtp", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-sdp = { package = "gstreamer-sdp", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-utils = { package = "gstreamer-utils", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-video = { package = "gstreamer-video", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
+gst-webrtc = { package = "gstreamer-webrtc", git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.22", version = "0.22" }
--- a/audio/spotify/README.md
+++ b/audio/spotify/README.md
@ -8,10 +8,11 @@ to respect their legal/licensing restrictions.

 ## Spotify Credentials

-This plugin requires a [Spotify Premium](https://www.spotify.com/premium/) account configured
-with a [device password](https://www.spotify.com/us/account/set-device-password/).
+This plugin requires a [Spotify Premium](https://www.spotify.com/premium/) account.
+If your account is linked with Facebook, you'll need to setup
+a [device username and password](https://www.spotify.com/us/account/set-device-password/).

-You can then set the device username and password using the `username` and `password` properties.
+Those username and password are then set using the `username` and `password` properties.

 You may also want to cache credentials and downloaded files, see the `cache-` properties on the element.

--- a/audio/spotify/src/common.rs
+++ b/audio/spotify/src/common.rs
@ -30,13 +30,13 @@ impl Settings {
    pub fn properties() -> Vec<glib::ParamSpec> {
        vec![glib::ParamSpecString::builder("username")
                    .nick("Username")
-                    .blurb("Spotify device username from https://www.spotify.com/us/account/set-device-password/")
+                    .blurb("Spotify username, Facebook accounts need a device username from https://www.spotify.com/us/account/set-device-password/")
                    .default_value(Some(""))
                    .mutable_ready()
                    .build(),
                glib::ParamSpecString::builder("password")
                    .nick("Password")
-                    .blurb("Spotify device password from https://www.spotify.com/us/account/set-device-password/")
+                    .blurb("Spotify password, Facebook accounts need a device password from https://www.spotify.com/us/account/set-device-password/")
                    .default_value(Some(""))
                    .mutable_ready()
                    .build(),
--- a/deny.toml
+++ b/deny.toml
@ -70,6 +70,9 @@ version = "0.9"
 [[bans.skip]]
 name = "hmac"
 version = "0.11"
+[[bans.skip]]
+name = "zerocopy"
+version = "0.6"

 # field-offset and nix depend on an older memoffset
 # https://github.com/Diggsey/rust-field-offset/pull/23
@ -184,6 +187,14 @@ version = "0.2"
 [[bans.skip]]
 name = "toml_edit"
 version = "0.21"
+[[bans.skip]]
+name = "winnow"
+version = "0.5"
+
+# Various crates depend on an older version of heck
+[[bans.skip]]
+name = "heck"
+version = "0.4"

 [sources]
 unknown-registry = "deny"
--- a/docs/meson.build
+++ b/docs/meson.build
@ -1,5 +1,9 @@
 build_hotdoc = false

+if get_option('doc').disabled()
+  subdir_done()
+endif
+
 if meson.is_cross_build()
    if get_option('doc').enabled()
        error('Documentation enabled but building the doc while cross building is not supported yet.')
--- a/docs/plugins/gst_plugins_cache.json
+++ b/docs/plugins/gst_plugins_cache.json
@ -6704,12 +6704,14 @@
                    "audio_%%u": {
                        "caps": "audio/x-raw:\naudio/x-opus:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    },
                    "video_%%u": {
                        "caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    }
                },
                "rank": "none"
@ -6735,12 +6737,14 @@
                    "audio_%%u": {
                        "caps": "audio/x-raw:\naudio/x-opus:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    },
                    "video_%%u": {
                        "caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    }
                },
                "rank": "none"
@ -6766,12 +6770,46 @@
                    "audio_%%u": {
                        "caps": "audio/x-raw:\naudio/x-opus:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    },
                    "video_%%u": {
                        "caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
+                    }
+                },
+                "rank": "none"
+            },
+            "livekitwebrtcsrc": {
+                "author": "Jordan Yelloz <jordan.yelloz@collabora.com>",
+                "description": "WebRTC source with LiveKit signaller",
+                "hierarchy": [
+                    "GstLiveKitWebRTCSrc",
+                    "GstBaseWebRTCSrc",
+                    "GstBin",
+                    "GstElement",
+                    "GstObject",
+                    "GInitiallyUnowned",
+                    "GObject"
+                ],
+                "interfaces": [
+                    "GstChildProxy"
+                ],
+                "klass": "Source/Network/WebRTC",
+                "pad-templates": {
+                    "audio_%%u": {
+                        "caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n",
+                        "direction": "src",
+                        "presence": "sometimes",
+                        "type": "GstWebRTCSrcPad"
+                    },
+                    "video_%%u": {
+                        "caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n",
+                        "direction": "src",
+                        "presence": "sometimes",
+                        "type": "GstWebRTCSrcPad"
                    }
                },
                "rank": "none"
@ -6797,12 +6835,14 @@
                    "audio_%%u": {
                        "caps": "audio/x-raw:\naudio/x-opus:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    },
                    "video_%%u": {
                        "caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    }
                },
                "rank": "none"
@ -6862,12 +6902,14 @@
                    "audio_%%u": {
                        "caps": "audio/x-raw:\naudio/x-opus:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    },
                    "video_%%u": {
                        "caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n",
                        "direction": "sink",
-                        "presence": "request"
+                        "presence": "request",
+                        "type": "GstWebRTCSinkPad"
                    }
                },
                "rank": "none"
@ -7237,6 +7279,18 @@
                        "type": "GstValueArray",
                        "writable": true
                    },
+                    "do-retransmission": {
+                        "blurb": "Send retransmission events upstream when a packet is late",
+                        "conditionally-available": false,
+                        "construct": false,
+                        "construct-only": false,
+                        "controllable": false,
+                        "default": "true",
+                        "mutable": "ready",
+                        "readable": true,
+                        "type": "gboolean",
+                        "writable": true
+                    },
                    "enable-data-channel-navigation": {
                        "blurb": "Enable navigation events through a dedicated WebRTCDataChannel",
                        "conditionally-available": false,
@ -7590,6 +7644,32 @@
                    }
                ]
            },
+            "GstWebRTCSinkPad": {
+                "hierarchy": [
+                    "GstWebRTCSinkPad",
+                    "GstGhostPad",
+                    "GstProxyPad",
+                    "GstPad",
+                    "GstObject",
+                    "GInitiallyUnowned",
+                    "GObject"
+                ],
+                "kind": "object",
+                "properties": {
+                    "msid": {
+                        "blurb": "Remote MediaStream ID in use for this pad",
+                        "conditionally-available": false,
+                        "construct": false,
+                        "construct-only": false,
+                        "controllable": false,
+                        "default": "NULL",
+                        "mutable": "ready",
+                        "readable": true,
+                        "type": "gchararray",
+                        "writable": true
+                    }
+                }
+            },
            "GstWebRTCSrcPad": {
                "hierarchy": [
                    "GstWebRTCSrcPad",
@ -7813,7 +7893,7 @@
                        "writable": true
                    },
                    "password": {
-                        "blurb": "Spotify device password from https://www.spotify.com/us/account/set-device-password/",
+                        "blurb": "Spotify password, Facebook accounts need a device password from https://www.spotify.com/us/account/set-device-password/",
                        "conditionally-available": false,
                        "construct": false,
                        "construct-only": false,
@ -7837,7 +7917,7 @@
                        "writable": true
                    },
                    "username": {
-                        "blurb": "Spotify device username from https://www.spotify.com/us/account/set-device-password/",
+                        "blurb": "Spotify username, Facebook accounts need a device username from https://www.spotify.com/us/account/set-device-password/",
                        "conditionally-available": false,
                        "construct": false,
                        "construct-only": false,
--- a/generic/threadshare/tests/pipeline.rs
+++ b/generic/threadshare/tests/pipeline.rs
@ -609,6 +609,8 @@ fn premature_shutdown() {
 }

 #[test]
+// FIXME: racy: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/250
+#[ignore]
 fn socket_play_null_play() {
    use gio::{
        prelude::SocketExt, InetAddress, InetSocketAddress, SocketFamily, SocketProtocol,
--- a/generic/threadshare/tests/udpsink.rs
+++ b/generic/threadshare/tests/udpsink.rs
@ -76,6 +76,8 @@ fn test_client_management() {
 }

 #[test]
+// FIXME: racy: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/250
+#[ignore]
 fn test_chain() {
    init();

--- a/meson.build
+++ b/meson.build
@ -1,7 +1,7 @@
 project('gst-plugins-rs',
        'rust',
        'c',
-        version: '0.12.0-alpha.1',
+        version: '0.12.2',
        meson_version : '>= 1.1')

 # dependencies.py needs a toml parsing module
@ -489,6 +489,16 @@ foreach plugin : plugins
    plugin_name = plugin_name.substring(3)
  endif

+  plugin_display_name = plugin_name
+  if plugin_name.startswith('gst')
+    plugin_display_name = plugin_name.substring(3)
+  endif
+  if plugin_display_name in plugin_names
+    # When default_library=both plugins are duplicated.
+    continue
+  endif
+  plugin_names += plugin_display_name
+
  option_name = plugin_name.substring(3)
  if option_name.startswith('rs')
    option_name = option_name.substring(2)
@ -533,13 +543,7 @@ foreach plugin : plugins
    warning('Static plugin @0@ is known to fail. It will not be included in libgstreamer-full.'.format(plugin_name))
  else
    gst_plugins += dep
-
    pc_files += [plugin_name + '.pc']
-    if plugin_name.startswith('gst')
-      plugin_names += [plugin_name.substring(3)]
-    else
-      plugin_names += [plugin_name]
-    endif
  endif
 endforeach

--- a/mux/fmp4/Cargo.toml
+++ b/mux/fmp4/Cargo.toml
@ -26,8 +26,8 @@ path = "src/lib.rs"
 gst-app = { workspace = true, features = ["v1_18"] }
 gst-check = { workspace = true, features = ["v1_18"] }
 m3u8-rs = "5.0"
-chrono = "0.4"
-dash-mpd = { version = "0.14", default-features = false }
+chrono = "0.4.35"
+dash-mpd = { version = "0.15", default-features = false }
 quick-xml = { version = "0.31", features = ["serialize"] }
 serde = "1"

--- a/mux/fmp4/examples/dash_vod.rs
+++ b/mux/fmp4/examples/dash_vod.rs
@ -182,8 +182,8 @@ fn main() -> Result<(), Error> {
                let mut write_segment =
                    |start: gst::ClockTime, duration: gst::ClockTime, repeat: usize| {
                        let mut s = dash_mpd::S {
-                            t: Some(start.mseconds() as i64),
-                            d: duration.mseconds() as i64,
+                            t: Some(start.mseconds()),
+                            d: duration.mseconds(),
                            ..Default::default()
                        };
                        if repeat > 0 {
--- a/mux/fmp4/examples/hls_live.rs
+++ b/mux/fmp4/examples/hls_live.rs
@ -153,7 +153,7 @@ fn trim_segments(state: &mut StreamState) {
            // safe side
            removal_time: segment
                .date_time
-                .checked_add_signed(Duration::seconds(20))
+                .checked_add_signed(Duration::try_seconds(20).unwrap())
                .unwrap(),
            path: segment.path.clone(),
        });
--- a/net/rtp/Cargo.toml
+++ b/net/rtp/Cargo.toml
@ -12,8 +12,9 @@ rust-version.workspace = true
 bitstream-io = "2.0"
 gst      = { workspace = true, features = ["v1_20"] }
 gst-rtp  = { workspace = true, features = ["v1_20"]}
-chrono = { version = "0.4", default-features = false }
+time = { version = "0.3", default-features = false, features = ["std"] }
 once_cell.workspace = true
+smallvec = { version = "1.11", features = ["union", "write", "const_generics", "const_new"] }

 [dev-dependencies]
 gst-check = { workspace = true, features = ["v1_20"] }
--- a/net/rtp/src/gcc/imp.rs
+++ b/net/rtp/src/gcc/imp.rs
@ -17,19 +17,20 @@
 * responsible for determining what bitrate to give to each encode)
 *
 */
-use chrono::Duration;
 use gst::{glib, prelude::*, subclass::prelude::*};
 use once_cell::sync::Lazy;
+use smallvec::SmallVec;
 use std::{
    collections::{BTreeMap, VecDeque},
    fmt,
    fmt::Debug,
    mem,
    sync::Mutex,
-    time,
 };
+use time::Duration;

 type Bitrate = u32;
+type BufferList = SmallVec<[gst::Buffer; 10]>;

 const DEFAULT_MIN_BITRATE: Bitrate = 1000;
 const DEFAULT_ESTIMATED_BITRATE: Bitrate = 2_048_000;
@ -44,7 +45,7 @@ static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
 });

 // Table1. Time limit in milliseconds  between packet bursts which  identifies a group
-static BURST_TIME: Lazy<Duration> = Lazy::new(|| Duration::milliseconds(5));
+const BURST_TIME: Duration = Duration::milliseconds(5);

 // Table1. Coefficient used for the measured noise variance
 //  [0.1,0.001]
@ -55,13 +56,13 @@ const ONE_MINUS_CHI: f64 = 1. - CHI;
 const Q: f64 = 0.001;

 // Table1. Initial value for the adaptive threshold
-static INITIAL_DEL_VAR_TH: Lazy<Duration> = Lazy::new(|| Duration::microseconds(12500));
+const INITIAL_DEL_VAR_TH: Duration = Duration::microseconds(12500);

 // Table1. Initial value of the system error covariance
 const INITIAL_ERROR_COVARIANCE: f64 = 0.1;

 // Table1. Time required to trigger an overuse signal
-static OVERUSE_TIME_TH: Lazy<Duration> = Lazy::new(|| Duration::milliseconds(10));
+const OVERUSE_TIME_TH: Duration = Duration::milliseconds(10);

 // from 5.5 "beta is typically chosen to be in the interval [0.8, 0.95], 0.85 is the RECOMMENDED value."
 const BETA: f64 = 0.85;
@ -76,7 +77,7 @@ const MOVING_AVERAGE_SMOOTHING_FACTOR: f64 = 0.5;
 // `N(i)` is the number of packets received the past T seconds and `L(j)` is
 // the payload size of packet j.  A window between 0.5 and 1 second is
 // RECOMMENDED.
-static PACKETS_RECEIVED_WINDOW: Lazy<Duration> = Lazy::new(|| Duration::milliseconds(1000)); // ms
+const PACKETS_RECEIVED_WINDOW: Duration = Duration::milliseconds(1000); // ms

 // from "5.4 Over-use detector" ->
 // Moreover, del_var_th(i) SHOULD NOT be updated if this condition
@ -85,32 +86,32 @@ static PACKETS_RECEIVED_WINDOW: Lazy<Duration> = Lazy::new(|| Duration::millisec
 // ```
 // |m(i)| - del_var_th(i) > 15
 // ```
-static MAX_M_MINUS_DEL_VAR_TH: Lazy<Duration> = Lazy::new(|| Duration::milliseconds(15));
+const MAX_M_MINUS_DEL_VAR_TH: Duration = Duration::milliseconds(15);

 // from 5.4 "It is also RECOMMENDED to clamp del_var_th(i) to the range [6, 600]"
-static MIN_THRESHOLD: Lazy<Duration> = Lazy::new(|| Duration::milliseconds(6));
-static MAX_THRESHOLD: Lazy<Duration> = Lazy::new(|| Duration::milliseconds(600));
+const MIN_THRESHOLD: Duration = Duration::milliseconds(6);
+const MAX_THRESHOLD: Duration = Duration::milliseconds(600);

 // From 5.5 ""Close" is defined as three standard deviations around this average"
 const STANDARD_DEVIATION_CLOSE_NUM: f64 = 3.;

 // Minimal duration between 2 updates on the lost based rate controller
-static LOSS_UPDATE_INTERVAL: Lazy<time::Duration> = Lazy::new(|| time::Duration::from_millis(200));
-static LOSS_DECREASE_THRESHOLD: f64 = 0.1;
-static LOSS_INCREASE_THRESHOLD: f64 = 0.02;
-static LOSS_INCREASE_FACTOR: f64 = 1.05;
+const LOSS_UPDATE_INTERVAL: Duration = Duration::milliseconds(200);
+const LOSS_DECREASE_THRESHOLD: f64 = 0.1;
+const LOSS_INCREASE_THRESHOLD: f64 = 0.02;
+const LOSS_INCREASE_FACTOR: f64 = 1.05;

 // Minimal duration between 2 updates on the lost based rate controller
-static DELAY_UPDATE_INTERVAL: Lazy<time::Duration> = Lazy::new(|| time::Duration::from_millis(100));
+const DELAY_UPDATE_INTERVAL: Duration = Duration::milliseconds(100);

-static ROUND_TRIP_TIME_WINDOW_SIZE: usize = 100;
+const ROUND_TRIP_TIME_WINDOW_SIZE: usize = 100;

-fn ts2dur(t: gst::ClockTime) -> Duration {
+const fn ts2dur(t: gst::ClockTime) -> Duration {
    Duration::nanoseconds(t.nseconds() as i64)
 }

-fn dur2ts(t: Duration) -> gst::ClockTime {
-    gst::ClockTime::from_nseconds(t.num_nanoseconds().unwrap() as u64)
+const fn dur2ts(t: Duration) -> gst::ClockTime {
+    gst::ClockTime::from_nseconds(t.whole_nanoseconds() as u64)
 }

 #[derive(Debug)]
@ -118,7 +119,9 @@ enum BandwidthEstimationOp {
    /// Don't update target bitrate
    Hold,
    /// Decrease target bitrate
+    #[allow(unused)]
    Decrease(String /* reason */),
+    #[allow(unused)]
    Increase(String /* reason */),
 }

@ -160,7 +163,7 @@ impl Packet {
        let seqnum = structure.get::<u32>("seqnum").unwrap() as u64;
        if lost {
            return Some(Packet {
-                arrival: Duration::zero(),
+                arrival: Duration::ZERO,
                departure: ts2dur(departure),
                size: structure.get::<u32>("size").unwrap() as usize,
                seqnum,
@ -189,18 +192,12 @@ impl Default for PacketGroup {
    fn default() -> Self {
        Self {
            packets: Default::default(),
-            departure: Duration::zero(),
+            departure: Duration::ZERO,
            arrival: None,
        }
    }
 }

-fn pdur(d: &Duration) -> String {
-    let stdd = time::Duration::from_nanos(d.num_nanoseconds().unwrap().unsigned_abs());
-
-    format!("{}{stdd:?}", if d.lt(&Duration::zero()) { "-" } else { "" })
-}
-
 impl PacketGroup {
    fn add(&mut self, packet: Packet) {
        if self.departure.is_zero() {
@ -310,10 +307,10 @@ impl Debug for Detector {
            "Network Usage: {:?}. Effective bitrate: {}ps - Measure: {} Estimate: {} threshold {} - overuse_estimate {}",
            self.usage,
            human_kbits(self.effective_bitrate()),
-            pdur(&self.measure),
-            pdur(&self.estimate),
-            pdur(&self.threshold),
-            pdur(&self.last_overuse_estimate),
+            self.measure,
+            self.estimate,
+            self.threshold,
+            self.last_overuse_estimate,
        )
    }
 }
@ -323,7 +320,7 @@ impl Detector {
        Detector {
            group: Default::default(),
            prev_group: Default::default(),
-            measure: Duration::zero(),
+            measure: Duration::ZERO,

            /* Smallish value to hold PACKETS_RECEIVED_WINDOW packets */
            last_received_packets: BTreeMap::new(),
@ -334,16 +331,16 @@ impl Detector {
            gain: 0.,
            measurement_uncertainty: 0.,
            estimate_error: INITIAL_ERROR_COVARIANCE,
-            estimate: Duration::zero(),
+            estimate: Duration::ZERO,

-            threshold: *INITIAL_DEL_VAR_TH,
+            threshold: INITIAL_DEL_VAR_TH,
            last_threshold_update: None,
            num_deltas: 0,

            last_use_detector_update: time::Instant::now(),
            increasing_counter: 0,
-            last_overuse_estimate: Duration::zero(),
-            increasing_duration: Duration::zero(),
+            last_overuse_estimate: Duration::ZERO,
+            increasing_duration: Duration::ZERO,

            rtts: Default::default(),
            clock: gst::SystemClock::obtain(),
@ -371,7 +368,7 @@ impl Detector {
            .unwrap()
            .arrival;

-        while last_arrival - self.oldest_packet_in_window_ts() > *PACKETS_RECEIVED_WINDOW {
+        while last_arrival - self.oldest_packet_in_window_ts() > PACKETS_RECEIVED_WINDOW {
            let oldest_seqnum = *self.last_received_packets.iter().next().unwrap().0;
            self.last_received_packets.remove(&oldest_seqnum);
        }
@ -398,9 +395,8 @@ impl Detector {
            .sum::<f64>()
            * 8.;

-        (bits
-            / (duration.num_nanoseconds().unwrap() as f64
-                / gst::ClockTime::SECOND.nseconds() as f64)) as Bitrate
+        (bits / (duration.whole_nanoseconds() as f64 / gst::ClockTime::SECOND.nseconds() as f64))
+            as Bitrate
    }

    fn oldest_packet_in_window_ts(&self) -> Duration {
@ -425,7 +421,7 @@ impl Detector {
            (self
                .rtts
                .iter()
-                .map(|d| d.num_nanoseconds().unwrap() as f64)
+                .map(|d| d.whole_nanoseconds() as f64)
                .sum::<f64>()
                / self.rtts.len() as f64) as i64,
        )
@ -481,7 +477,7 @@ impl Detector {
            }

            if pkt.departure >= self.group.departure {
-                if self.group.inter_departure_time_pkt(pkt) < *BURST_TIME {
+                if self.group.inter_departure_time_pkt(pkt) < BURST_TIME {
                    self.group.add(*pkt);
                    continue;
                }
@ -491,8 +487,8 @@ impl Detector {
                // A Packet which has an inter-arrival time less than burst_time and
                // an inter-group delay variation d(i) less than 0 is considered
                // being part of the current group of packets.
-                if self.group.inter_arrival_time_pkt(pkt) < *BURST_TIME
-                    && self.group.inter_delay_variation_pkt(pkt) < Duration::zero()
+                if self.group.inter_arrival_time_pkt(pkt) < BURST_TIME
+                    && self.group.inter_delay_variation_pkt(pkt) < Duration::ZERO
                {
                    self.group.add(*pkt);
                    continue;
@ -502,7 +498,7 @@ impl Detector {
                gst::trace!(
                    CAT,
                    "Packet group done: {:?}",
-                    gst::ClockTime::from_nseconds(group.departure.num_nanoseconds().unwrap() as u64)
+                    gst::ClockTime::from_nseconds(group.departure.whole_nanoseconds() as u64)
                );
                if let Some(prev_group) = mem::replace(&mut self.prev_group, Some(group.clone())) {
                    // 5.3 Arrival-time filter
@ -514,7 +510,7 @@ impl Detector {
                gst::debug!(
                    CAT,
                    "Ignoring packet departed at {:?} as we got feedback too late",
-                    gst::ClockTime::from_nseconds(pkt.departure.num_nanoseconds().unwrap() as u64)
+                    gst::ClockTime::from_nseconds(pkt.departure.whole_nanoseconds() as u64)
                );
            }
        }
@ -527,10 +523,7 @@ impl Detector {

        if let Some(ref last_update) = self.last_loss_update {
            self.loss_average = loss_fraction
-                + (-Duration::from_std(now - *last_update)
-                    .unwrap()
-                    .num_milliseconds() as f64)
-                    .exp()
+                + (-(now - *last_update).whole_milliseconds() as f64).exp()
                    * (self.loss_average - loss_fraction);
        }

@ -541,7 +534,7 @@ impl Detector {
        self.measure = group.inter_delay_variation(prev_group);

        let z = self.measure - self.estimate;
-        let zms = z.num_microseconds().unwrap() as f64 / 1000.0;
+        let zms = z.whole_microseconds() as f64 / 1000.0;

        // This doesn't exactly follows the spec as we should compute and
        // use f_max here, no implementation we have found actually uses it.
@ -575,11 +568,11 @@ impl Detector {
        }

        let t = Duration::nanoseconds(
-            self.estimate.num_nanoseconds().unwrap() * i64::min(self.num_deltas, MAX_DELTAS),
+            self.estimate.whole_nanoseconds() as i64 * i64::min(self.num_deltas, MAX_DELTAS),
        );
        let usage = if t > self.threshold {
            NetworkUsage::Over
-        } else if t.num_nanoseconds().unwrap() < -self.threshold.num_nanoseconds().unwrap() {
+        } else if t.whole_nanoseconds() < -self.threshold.whole_nanoseconds() {
            NetworkUsage::Under
        } else {
            NetworkUsage::Normal
@ -593,15 +586,15 @@ impl Detector {
    fn update_threshold(&mut self, estimate: &Duration) {
        const K_U: f64 = 0.01; // Table1. Coefficient for the adaptive threshold
        const K_D: f64 = 0.00018; // Table1. Coefficient for the adaptive threshold
-        const MAX_TIME_DELTA: time::Duration = time::Duration::from_millis(100);
+        const MAX_TIME_DELTA: Duration = Duration::milliseconds(100);

        let now = time::Instant::now();
        if self.last_threshold_update.is_none() {
            self.last_threshold_update = Some(now);
        }

-        let abs_estimate = Duration::nanoseconds(estimate.num_nanoseconds().unwrap().abs());
-        if abs_estimate > self.threshold + *MAX_M_MINUS_DEL_VAR_TH {
+        let abs_estimate = estimate.abs();
+        if abs_estimate > self.threshold + MAX_M_MINUS_DEL_VAR_TH {
            self.last_threshold_update = Some(now);
            return;
        }
@ -611,14 +604,12 @@ impl Detector {
        } else {
            K_U
        };
-        let time_delta =
-            Duration::from_std((now - self.last_threshold_update.unwrap()).min(MAX_TIME_DELTA))
-                .unwrap();
+        let time_delta = (now - self.last_threshold_update.unwrap()).min(MAX_TIME_DELTA);
        let d = abs_estimate - self.threshold;
-        let add = k * d.num_milliseconds() as f64 * time_delta.num_milliseconds() as f64;
+        let add = k * d.whole_milliseconds() as f64 * time_delta.whole_milliseconds() as f64;

        self.threshold += Duration::nanoseconds((add * 100. * 1_000.) as i64);
-        self.threshold = self.threshold.clamp(*MIN_THRESHOLD, *MAX_THRESHOLD);
+        self.threshold = self.threshold.clamp(MIN_THRESHOLD, MAX_THRESHOLD);
        self.last_threshold_update = Some(now);
    }

@ -626,22 +617,22 @@ impl Detector {
        let (th_usage, estimate) = self.compare_threshold();

        let now = time::Instant::now();
-        let delta = Duration::from_std(now - self.last_use_detector_update).unwrap();
+        let delta = now - self.last_use_detector_update;
        self.last_use_detector_update = now;
        gst::log!(
            CAT,
            "{:?} - self.estimate {} - estimate: {} - th: {}",
            th_usage,
-            pdur(&self.estimate),
-            pdur(&estimate),
-            pdur(&self.threshold)
+            self.estimate,
+            estimate,
+            self.threshold
        );
        match th_usage {
            NetworkUsage::Over => {
                self.increasing_duration += delta;
                self.increasing_counter += 1;

-                if self.increasing_duration > *OVERUSE_TIME_TH
+                if self.increasing_duration > OVERUSE_TIME_TH
                    && self.increasing_counter > 1
                    && estimate > self.last_overuse_estimate
                {
@ -649,7 +640,7 @@ impl Detector {
                }
            }
            NetworkUsage::Under | NetworkUsage::Normal => {
-                self.increasing_duration = Duration::zero();
+                self.increasing_duration = Duration::ZERO;
                self.increasing_counter = 0;

                self.usage = th_usage;
@ -759,10 +750,10 @@ impl Default for State {

 impl State {
    // 4. sending engine implementing a "leaky bucket"
-    fn create_buffer_list(&mut self, bwe: &super::BandwidthEstimator) -> gst::BufferList {
+    fn create_buffer_list(&mut self, bwe: &super::BandwidthEstimator) -> BufferList {
        let now = time::Instant::now();
-        let elapsed = Duration::from_std(now - self.last_push).unwrap();
-        let mut budget = (elapsed.num_nanoseconds().unwrap())
+        let elapsed = now - self.last_push;
+        let mut budget = (elapsed.whole_nanoseconds() as i64)
            .mul_div_round(
                self.estimated_bitrate as i64,
                gst::ClockTime::SECOND.nseconds() as i64,
@ -773,8 +764,8 @@ impl State {
        let mut remaining = self.buffers.iter().map(|b| b.size() as f64).sum::<f64>() * 8.;
        let total_size = remaining;

-        let mut list = gst::BufferList::new();
-        let mutlist = list.get_mut().unwrap();
+        let mut list_size = 0;
+        let mut list = BufferList::new();

        // Leak the bucket so it can hold at most 30ms of data
        let maximum_remaining_bits = 30. * self.estimated_bitrate as f64 / 1000.;
@ -784,7 +775,8 @@ impl State {
            let n_bits = buf.size() * 8;

            leaked = budget <= 0 && remaining > maximum_remaining_bits;
-            mutlist.add(buf);
+            list_size += buf.size();
+            list.push(buf);
            budget -= n_bits as i64;
            remaining -= n_bits as f64;
        }
@ -793,11 +785,11 @@ impl State {
            CAT,
            obj: bwe,
            "{} bitrate: {}ps budget: {}/{} sending: {} Remaining: {}/{}",
-            pdur(&elapsed),
+            elapsed,
            human_kbits(self.estimated_bitrate),
            human_kbits(budget as f64),
            human_kbits(total_budget as f64),
-            human_kbits(list.calculate_size() as f64 * 8.),
+            human_kbits(list_size as f64 * 8.),
            human_kbits(remaining),
            human_kbits(total_size)
        );
@ -815,11 +807,11 @@ impl State {
        let time_since_last_update_ms = match self.last_increase_on_delay {
            None => 0.,
            Some(prev) => {
-                if now - prev < *DELAY_UPDATE_INTERVAL {
+                if now - prev < DELAY_UPDATE_INTERVAL {
                    return None;
                }

-                (now - prev).as_millis() as f64
+                (now - prev).whole_milliseconds() as f64
            }
        };

@ -839,7 +831,7 @@ impl State {
            let packets_per_frame = f64::ceil(bits_per_frame / (1200. * 8.));
            let avg_packet_size_bits = bits_per_frame / packets_per_frame;

-            let rtt_ms = self.detector.rtt().num_milliseconds() as f64;
+            let rtt_ms = self.detector.rtt().whole_milliseconds() as f64;
            let response_time_ms = 100. + rtt_ms;
            let alpha = 0.5 * f64::min(time_since_last_update_ms / response_time_ms, 1.0);
            let threshold_on_effective_bitrate = 1.5 * effective_bitrate as f64;
@ -959,7 +951,7 @@ impl State {
        let now = time::Instant::now();

        if loss_ratio > LOSS_DECREASE_THRESHOLD
-            && (now - self.last_decrease_on_loss) > *LOSS_UPDATE_INTERVAL
+            && (now - self.last_decrease_on_loss) > LOSS_UPDATE_INTERVAL
        {
            let factor = 1. - (0.5 * loss_ratio);

@ -973,7 +965,7 @@ impl State {
                ControllerType::Loss,
            )
        } else if loss_ratio < LOSS_INCREASE_THRESHOLD
-            && (now - self.last_increase_on_loss) > *LOSS_UPDATE_INTERVAL
+            && (now - self.last_increase_on_loss) > LOSS_UPDATE_INTERVAL
        {
            self.last_control_op = BandwidthEstimationOp::Increase("Low loss".into());
            self.last_increase_on_loss = now;
@ -1000,7 +992,7 @@ impl State {
            },
            NetworkUsage::Over => {
                let now = time::Instant::now();
-                if now - self.last_decrease_on_delay > *DELAY_UPDATE_INTERVAL {
+                if now - self.last_decrease_on_delay > DELAY_UPDATE_INTERVAL {
                    let effective_bitrate = self.detector.effective_bitrate();
                    let target =
                        (self.estimated_bitrate as f64 * 0.95).min(BETA * effective_bitrate as f64);
@ -1037,8 +1029,14 @@ pub struct BandwidthEstimator {
 }

 impl BandwidthEstimator {
-    fn push_list(&self, list: gst::BufferList) -> Result<gst::FlowSuccess, gst::FlowError> {
-        let res = self.srcpad.push_list(list);
+    fn push_list(&self, list: BufferList) -> Result<gst::FlowSuccess, gst::FlowError> {
+        let mut res = Ok(gst::FlowSuccess::Ok);
+        for buf in list {
+            res = self.srcpad.push(buf);
+            if res.is_err() {
+                break;
+            }
+        }

        self.state.lock().unwrap().flow_return = res;

@ -1051,7 +1049,7 @@ impl BandwidthEstimator {
        let clock = gst::SystemClock::obtain();

        bwe.imp().state.lock().unwrap().clock_entry =
-            Some(clock.new_single_shot_id(clock.time().unwrap() + dur2ts(*BURST_TIME)));
+            Some(clock.new_single_shot_id(clock.time().unwrap() + dur2ts(BURST_TIME)));

        self.srcpad.start_task(move || {
            let pause = || {
@ -1088,10 +1086,7 @@ impl BandwidthEstimator {
            let list = {
                let mut state = lock_state();
                clock
-                    .single_shot_id_reinit(
-                        &clock_entry,
-                        clock.time().unwrap() + dur2ts(*BURST_TIME),
-                    )
+                    .single_shot_id_reinit(&clock_entry, clock.time().unwrap() + dur2ts(BURST_TIME))
                    .unwrap();
                state.clock_entry = Some(clock_entry);
                state.create_buffer_list(&bwe)
@ -1146,15 +1141,12 @@ impl ObjectSubclass for BandwidthEstimator {
    fn with_class(klass: &Self::Class) -> Self {
        let templ = klass.pad_template("sink").unwrap();
        let sinkpad = gst::Pad::builder_from_template(&templ)
-            .chain_function(|_pad, parent, mut buffer| {
+            .chain_function(|_pad, parent, buffer| {
                BandwidthEstimator::catch_panic_pad_function(
                    parent,
                    || Err(gst::FlowError::Error),
                    |this| {
                        let mut state = this.state.lock().unwrap();
-                        let mutbuf = buffer.make_mut();
-                        mutbuf.set_pts(None);
-                        mutbuf.set_dts(None);
                        state.buffers.push_front(buffer);

                        state.flow_return
--- a/net/webrtc/Cargo.toml
+++ b/net/webrtc/Cargo.toml
@ -25,11 +25,11 @@ futures = "0.3"
 tokio = { version = "1", features = ["fs", "macros", "rt-multi-thread", "time"] }
 tokio-native-tls = "0.3.0"
 tokio-stream = "0.1.11"
-async-tungstenite = { version = "0.24", features = ["tokio-runtime", "tokio-native-tls"] }
+async-tungstenite = { version = "0.25", features = ["tokio-runtime", "tokio-native-tls"] }
 serde = { version = "1", features = ["derive"] }
 serde_json = "1"
 fastrand = "2.0"
-gst_plugin_webrtc_protocol = { path="protocol", package = "gst-plugin-webrtc-signalling-protocol" }
+gst_plugin_webrtc_protocol = { path="protocol", package = "gst-plugin-webrtc-signalling-protocol", version = "0.12" }
 human_bytes = "0.4"
 url = "2"

--- a/net/webrtc/README.md
+++ b/net/webrtc/README.md
@ -350,6 +350,104 @@ gst-launch-1.0 -e uridecodebin uri=file:///home/meh/path/to/video/file ! \

 You should see a second video displayed in the videoroomtest web page.

+## Streaming from LiveKit using the livekitwebrtcsrc element
+
+First, publish a stream to the room using the following command:
+
+```shell
+gst-launch-1.0 livekitwebrtcsink name=sink \
+    signaller::ws-url=ws://127.0.0.1:7880 \
+    signaller::api-key=devkey \
+    signaller::secret-key=secret \
+    signaller::room-name=testroom \
+    signaller::identity=gst-producer \
+    signaller::participant-name=gst-producer \
+    video-caps='video/x-h264' \
+  videotestsrc is-live=1 \
+  ! video/x-raw,width=640,height=360,framerate=15/1 \
+  ! timeoverlay ! videoconvert ! queue ! sink.
+```
+
+Then play back the published stream:
+
+```shell
+gst-launch-1.0 livekitwebrtcsrc \
+    name=src \
+    signaller::ws-url=ws://127.0.0.1:7880 \
+    signaller::api-key=devkey \
+    signaller::secret-key=secret \
+    signaller::room-name=testroom \
+    signaller::identity=gst-consumer \
+    signaller::participant-name=gst-consumer \
+    signaller::producer-peer-id=gst-producer \
+    video-codecs='<H264>' \
+  src. ! queue ! videoconvert ! autovideosink
+```
+
+### Auto-subscribe with livekitwebrtcsrc element
+
+With the LiveKit source element, you can also subscribe to all the peers in
+your room, simply by not specifying any value for
+`signaller::producer-peer-id`. Unwanted peers can also be ignored by supplying
+an array of peer IDs to `signaller::excluded-producer-peer-ids`. Importantly,
+it is also necessary to add sinks for all the streams in the room that the
+source element has subscribed to.
+
+First, publish a few streams using different connections:
+
+```shell
+gst-launch-1.0 \
+  livekitwebrtcsink name=sinka \
+    signaller::ws-url=ws://127.0.0.1:7880 \
+    signaller::api-key=devkey \
+    signaller::secret-key=secret \
+    signaller::room-name=testroom \
+    signaller::identity=gst-producer-a \
+    signaller::participant-name=gst-producer-a \
+    video-caps='video/x-vp8' \
+  livekitwebrtcsink name=sinkb \
+    signaller::ws-url=ws://127.0.0.1:7880 \
+    signaller::api-key=devkey \
+    signaller::secret-key=secret \
+    signaller::room-name=testroom \
+    signaller::identity=gst-producer-b \
+    signaller::participant-name=gst-producer-b \
+    video-caps='video/x-vp8' \
+  livekitwebrtcsink name=sinkc \
+    signaller::ws-url=ws://127.0.0.1:7880 \
+    signaller::api-key=devkey \
+    signaller::secret-key=secret \
+    signaller::room-name=testroom \
+    signaller::identity=gst-producer-c \
+    signaller::participant-name=gst-producer-c \
+    video-caps='video/x-vp8' \
+  videotestsrc is-live=1 \
+  ! video/x-raw,width=640,height=360,framerate=15/1 \
+  ! timeoverlay ! videoconvert ! queue ! sinka. \
+  videotestsrc pattern=ball is-live=1 \
+  ! video/x-raw,width=320,height=180,framerate=15/1 \
+  ! timeoverlay ! videoconvert ! queue ! sinkb.
+  videotestsrc is-live=1 \
+  ! video/x-raw,width=320,height=180,framerate=15/1 \
+  ! timeoverlay ! videoconvert ! queue ! sinkc.
+```
+
+Then watch only streams A and B by excluding peer C:
+
+```shell
+gst-launch-1.0 livekitwebrtcsrc \
+  name=src \
+  signaller::ws-url=ws://127.0.0.1:7880 \
+  signaller::api-key=devkey \
+  signaller::secret-key=secret \
+  signaller::room-name=testroom \
+  signaller::identity=gst-consumer \
+  signaller::participant-name=gst-consumer \
+  signaller::excluded-producer-peer-ids='<gst-producer-c>' \
+  src. ! queue ! videoconvert ! autovideosink
+  src. ! queue ! videoconvert ! autovideosink
+```
+
 [LiveKit]: https://livekit.io/
 [janus]: https://github.com/meetecho/janus-gateway
 [simple whip server]: https://github.com/meetecho/simple-whip-server/
--- a/net/webrtc/signalling/Cargo.toml
+++ b/net/webrtc/signalling/Cargo.toml
@ -12,7 +12,7 @@ rust-version.workspace = true
 anyhow = "1"
 tokio = { version = "1", features = ["fs", "io-util", "macros", "rt-multi-thread", "time"] }
 tokio-native-tls = "0.3.0"
-async-tungstenite = { version = "0.24", features = ["tokio-runtime", "tokio-native-tls"] }
+async-tungstenite = { version = "0.25", features = ["tokio-runtime", "tokio-native-tls"] }
 serde = { version = "1", features = ["derive"] }
 serde_json = "1"
 clap = { version = "4", features = ["derive"] }
@ -24,7 +24,7 @@ uuid = { version = "1", features = ["v4"] }
 thiserror = "1"
 test-log = { version = "0.2", features = ["trace"], default-features = false }
 pin-project-lite = "0.2"
-gst_plugin_webrtc_protocol = { path="../protocol", package = "gst-plugin-webrtc-signalling-protocol" }
+gst_plugin_webrtc_protocol = { path="../protocol", package = "gst-plugin-webrtc-signalling-protocol", version = "0.12" }

 [[bin]]
 name = "gst-webrtc-signalling-server"
--- a/net/webrtc/src/janusvr_signaller/imp.rs
+++ b/net/webrtc/src/janusvr_signaller/imp.rs
@ -42,17 +42,37 @@ fn feed_id() -> u32 {
    thread_rng().gen()
 }

+#[derive(Serialize, Deserialize, Debug, PartialEq, Eq, Clone)]
+#[serde(untagged)]
+/// Ids are either u64 (default) or string in Janus, depending of the
+/// `string_ids` configuration in the videoroom plugin config file.
+enum JanusId {
+    Str(String),
+    Num(u64),
+}
+
+impl std::fmt::Display for JanusId {
+    fn fmt(&self, f: &mut std::fmt::Formatter<'_>) -> std::fmt::Result {
+        match self {
+            JanusId::Str(s) => write!(f, "{s}"),
+            JanusId::Num(n) => write!(f, "{n}"),
+        }
+    }
+}
+
 #[derive(Serialize, Deserialize, Debug, PartialEq, Eq)]
 struct KeepAliveMsg {
    janus: String,
    transaction: String,
    session_id: u64,
+    apisecret: Option<String>,
 }

 #[derive(Serialize, Deserialize, Debug, PartialEq, Eq)]
 struct CreateSessionMsg {
    janus: String,
    transaction: String,
+    apisecret: Option<String>,
 }

 #[derive(Serialize, Deserialize, Debug, PartialEq, Eq)]
@ -61,14 +81,15 @@ struct AttachPluginMsg {
    transaction: String,
    plugin: String,
    session_id: u64,
+    apisecret: Option<String>,
 }

 #[derive(Serialize, Deserialize, Debug, PartialEq, Eq)]
 struct RoomRequestBody {
    request: String,
    ptype: String,
-    room: u64,
-    id: u32,
+    room: JanusId,
+    id: JanusId,
    #[serde(skip_serializing_if = "Option::is_none")]
    display: Option<String>,
 }
@ -79,14 +100,13 @@ struct RoomRequestMsg {
    transaction: String,
    session_id: u64,
    handle_id: u64,
+    apisecret: Option<String>,
    body: RoomRequestBody,
 }

 #[derive(Serialize, Deserialize, Debug, PartialEq, Eq)]
 struct PublishBody {
    request: String,
-    audio: bool,
-    video: bool,
 }

 #[derive(Serialize, Deserialize, Debug, PartialEq, Eq)]
@ -102,6 +122,7 @@ struct PublishMsg {
    transaction: String,
    session_id: u64,
    handle_id: u64,
+    apisecret: Option<String>,
    body: PublishBody,
    jsep: Jsep,
 }
@ -119,6 +140,7 @@ struct TrickleMsg {
    transaction: String,
    session_id: u64,
    handle_id: u64,
+    apisecret: Option<String>,
    candidate: Candidate,
 }

@ -139,25 +161,39 @@ struct InnerError {
    reason: String,
 }

+#[derive(Serialize, Deserialize, Debug)]
+struct InnerHangup {
+    session_id: JanusId,
+    sender: JanusId,
+    reason: String,
+}
+
 #[derive(Serialize, Deserialize, Debug)]
 struct RoomJoined {
-    room: Option<u64>,
+    room: JanusId,
 }

 #[derive(Serialize, Deserialize, Debug)]
 struct RoomEvent {
-    room: Option<u64>,
+    room: Option<JanusId>,
    error_code: Option<i32>,
    error: Option<String>,
 }

 #[derive(Serialize, Deserialize, Debug)]
 #[serde(tag = "videoroom")]
+struct RoomDestroyed {
+    room: JanusId,
+}
+
+#[derive(Serialize, Deserialize, Debug)]
+#[serde(tag = "videoroom", rename_all = "kebab-case")]
 enum VideoRoomData {
    #[serde(rename = "joined")]
    Joined(RoomJoined),
    #[serde(rename = "event")]
    Event(RoomEvent),
+    Destroyed(RoomDestroyed),
 }

 #[derive(Serialize, Deserialize, Debug)]
@ -189,20 +225,15 @@ struct EventMsg {

 // IncomingMessage
 #[derive(Serialize, Deserialize, Debug)]
-#[serde(tag = "janus")]
+#[serde(tag = "janus", rename_all = "lowercase")]
 enum JsonReply {
-    #[serde(rename = "ack")]
    Ack,
-    #[serde(rename = "success")]
    Success(SuccessMsg),
-    #[serde(rename = "event")]
    Event(EventMsg),
-    #[serde(rename = "webrtcup")]
    WebRTCUp,
-    #[serde(rename = "media")]
    Media,
-    #[serde(rename = "error")]
    Error(InnerError),
+    HangUp(InnerHangup),
 }

 #[derive(Default)]
@ -213,14 +244,18 @@ struct State {
    session_id: Option<u64>,
    handle_id: Option<u64>,
    transaction_id: Option<String>,
+    room_id: Option<JanusId>,
+    feed_id: Option<JanusId>,
 }

 #[derive(Clone)]
 struct Settings {
    janus_endpoint: String,
    room_id: Option<String>,
-    feed_id: u32,
+    feed_id: String,
    display_name: Option<String>,
+    secret_key: Option<String>,
+    string_ids: bool,
 }

 impl Default for Settings {
@ -228,8 +263,10 @@ impl Default for Settings {
        Self {
            janus_endpoint: "ws://127.0.0.1:8188".to_string(),
            room_id: None,
-            feed_id: feed_id(),
+            feed_id: feed_id().to_string(),
            display_name: None,
+            secret_key: None,
+            string_ids: false,
        }
    }
 }
@ -240,8 +277,10 @@ pub struct Signaller {
    state: Mutex<State>,
    #[property(name="janus-endpoint", get, set, type = String, member = janus_endpoint, blurb = "The Janus server endpoint to POST SDP offer to")]
    #[property(name="room-id", get, set, type = String, member = room_id, blurb = "The Janus Room ID that will be joined to")]
-    #[property(name="feed-id", get, set, type = u32, member = feed_id, blurb = "The Janus Feed ID to identify where the track is coming from")]
+    #[property(name="feed-id", get, set, type = String, member = feed_id, blurb = "The Janus Feed ID to identify where the track is coming from")]
    #[property(name="display-name", get, set, type = String, member = display_name, blurb = "The name of the publisher in the Janus Video Room")]
+    #[property(name="secret-key", get, set, type = String, member = secret_key, blurb = "The secret API key to communicate with Janus server")]
+    #[property(name="string-ids", get, set, type = bool, member = string_ids, blurb = "Force passing room-id and feed-id as string even if they can be parsed into an integer")]
    settings: Mutex<Settings>,
 }

@ -292,15 +331,20 @@ impl Signaller {
                        },
                        _ = tokio::time::sleep(Duration::from_secs(10)) => {
                            if let Some(ref this) = this {
-                                let (transaction, session_id) = {
+                                let (transaction, session_id, apisecret) = {
                                    let state = this.state.lock().unwrap();
-                                    (state.transaction_id.clone().unwrap(),
-                                    state.session_id.unwrap())
+                                    let settings = this.settings.lock().unwrap();
+                                    (
+                                        state.transaction_id.clone().unwrap(),
+                                        state.session_id.unwrap(),
+                                        settings.secret_key.clone(),
+                                    )
                                };
                                let msg = OutgoingMessage::KeepAlive(KeepAliveMsg {
                                    janus: "keepalive".to_string(),
                                    transaction,
                                    session_id,
+                                    apisecret,
                                });
                                res = ws_sink
                                    .send(WsMessage::Text(serde_json::to_string(&msg).unwrap()))
@ -401,10 +445,8 @@ impl Signaller {
                if let Some(PluginData::VideoRoom { data: plugindata }) = event.plugindata {
                    match plugindata {
                        VideoRoomData::Joined(joined) => {
-                            if let Some(room) = joined.room {
-                                gst::trace!(CAT, imp: self, "Joined room {} successfully", room);
-                                self.session_requested();
-                            }
+                            gst::trace!(CAT, imp: self, "Joined room {:?} successfully", joined.room);
+                            self.session_requested();
                        }
                        VideoRoomData::Event(room_event) => {
                            if room_event.error_code.is_some() && room_event.error.is_some() {
@ -423,12 +465,21 @@ impl Signaller {
                                }
                            }
                        }
+                        VideoRoomData::Destroyed(room_destroyed) => {
+                            gst::trace!(CAT, imp: self, "Room {} has been destroyed", room_destroyed.room);
+
+                            self.raise_error(format!(
+                                "room {} has been destroyed",
+                                room_destroyed.room
+                            ));
+                        }
                    }
                }
            }
            JsonReply::Error(error) => {
                self.raise_error(format!("code: {}, reason: {}", error.code, error.reason))
            }
+            JsonReply::HangUp(hangup) => self.raise_error(format!("hangup: {}", hangup.reason)),
            // ignore for now
            JsonReply::Ack | JsonReply::Media => {}
        }
@ -466,9 +517,12 @@ impl Signaller {
    fn create_session(&self) {
        let transaction = transaction_id();
        self.set_transaction_id(transaction.clone());
+        let settings = self.settings.lock().unwrap();
+        let apisecret = settings.secret_key.clone();
        self.send(OutgoingMessage::CreateSession(CreateSessionMsg {
            janus: "create".to_string(),
            transaction,
+            apisecret,
        }));
    }

@ -481,12 +535,14 @@ impl Signaller {
    }

    fn attach_plugin(&self) {
-        let (transaction, session_id) = {
+        let (transaction, session_id, apisecret) = {
            let state = self.state.lock().unwrap();
+            let settings = self.settings.lock().unwrap();

            (
                state.transaction_id.clone().unwrap(),
                state.session_id.unwrap(),
+                settings.secret_key.clone(),
            )
        };
        self.send(OutgoingMessage::AttachPlugin(AttachPluginMsg {
@ -494,12 +550,13 @@ impl Signaller {
            transaction,
            plugin: "janus.plugin.videoroom".to_string(),
            session_id,
+            apisecret,
        }));
    }

    fn join_room(&self) {
-        let (transaction, session_id, handle_id, room, feed_id, display) = {
-            let state = self.state.lock().unwrap();
+        let (transaction, session_id, handle_id, room, feed_id, display, apisecret) = {
+            let mut state = self.state.lock().unwrap();
            let settings = self.settings.lock().unwrap();

            if settings.room_id.is_none() {
@ -507,13 +564,38 @@ impl Signaller {
                return;
            }

+            /* room_id and feed_id can be either a string or integer depending
+             * on server configuration. The property is always a string, if we
+             * can parse it to integer then assume that's what the server expects,
+             * unless string-ids=true is set to force usage of strings.
+             * Save parsed value in state to not have to parse it again for future
+             * API calls.
+             */
+            if settings.string_ids {
+                state.room_id = Some(JanusId::Str(settings.room_id.clone().unwrap()));
+                state.feed_id = Some(JanusId::Str(settings.feed_id.clone()));
+            } else {
+                let room_id_str = settings.room_id.as_ref().unwrap();
+                match room_id_str.parse() {
+                    Ok(n) => {
+                        state.room_id = Some(JanusId::Num(n));
+                        state.feed_id = Some(JanusId::Num(settings.feed_id.parse().unwrap()));
+                    }
+                    Err(_) => {
+                        state.room_id = Some(JanusId::Str(room_id_str.clone()));
+                        state.feed_id = Some(JanusId::Str(settings.feed_id.clone()));
+                    }
+                };
+            }
+
            (
                state.transaction_id.clone().unwrap(),
                state.session_id.unwrap(),
                state.handle_id.unwrap(),
-                settings.room_id.as_ref().unwrap().parse().unwrap(),
-                settings.feed_id,
+                state.room_id.clone().unwrap(),
+                state.feed_id.clone().unwrap(),
                settings.display_name.clone(),
+                settings.secret_key.clone(),
            )
        };
        self.send(OutgoingMessage::RoomRequest(RoomRequestMsg {
@ -521,6 +603,7 @@ impl Signaller {
            transaction,
            session_id,
            handle_id,
+            apisecret,
            body: RoomRequestBody {
                request: "join".to_string(),
                ptype: "publisher".to_string(),
@ -532,7 +615,7 @@ impl Signaller {
    }

    fn leave_room(&self) {
-        let (transaction, session_id, handle_id, room, feed_id, display) = {
+        let (transaction, session_id, handle_id, room, feed_id, display, apisecret) = {
            let state = self.state.lock().unwrap();
            let settings = self.settings.lock().unwrap();

@ -545,9 +628,10 @@ impl Signaller {
                state.transaction_id.clone().unwrap(),
                state.session_id.unwrap(),
                state.handle_id.unwrap(),
-                settings.room_id.as_ref().unwrap().parse().unwrap(),
-                settings.feed_id,
+                state.room_id.clone().unwrap(),
+                state.feed_id.clone().unwrap(),
                settings.display_name.clone(),
+                settings.secret_key.clone(),
            )
        };
        self.send_blocking(OutgoingMessage::RoomRequest(RoomRequestMsg {
@ -555,6 +639,7 @@ impl Signaller {
            transaction,
            session_id,
            handle_id,
+            apisecret,
            body: RoomRequestBody {
                request: "leave".to_string(),
                ptype: "publisher".to_string(),
@ -566,7 +651,7 @@ impl Signaller {
    }

    fn publish(&self, offer: &gst_webrtc::WebRTCSessionDescription) {
-        let (transaction, session_id, handle_id) = {
+        let (transaction, session_id, handle_id, apisecret) = {
            let state = self.state.lock().unwrap();
            let settings = self.settings.lock().unwrap();

@ -579,6 +664,7 @@ impl Signaller {
                state.transaction_id.clone().unwrap(),
                state.session_id.unwrap(),
                state.handle_id.unwrap(),
+                settings.secret_key.clone(),
            )
        };
        let sdp_data = offer.sdp().as_text().unwrap();
@ -587,10 +673,9 @@ impl Signaller {
            transaction,
            session_id,
            handle_id,
+            apisecret,
            body: PublishBody {
                request: "publish".to_string(),
-                audio: true,
-                video: true,
            },
            jsep: Jsep {
                sdp: sdp_data,
@ -601,7 +686,7 @@ impl Signaller {
    }

    fn trickle(&self, candidate: &str, sdp_m_line_index: u32) {
-        let (transaction, session_id, handle_id) = {
+        let (transaction, session_id, handle_id, apisecret) = {
            let state = self.state.lock().unwrap();
            let settings = self.settings.lock().unwrap();

@ -614,6 +699,7 @@ impl Signaller {
                state.transaction_id.clone().unwrap(),
                state.session_id.unwrap(),
                state.handle_id.unwrap(),
+                settings.secret_key.clone(),
            )
        };
        self.send(OutgoingMessage::Trickle(TrickleMsg {
@ -621,6 +707,7 @@ impl Signaller {
            transaction,
            session_id,
            handle_id,
+            apisecret,
            candidate: Candidate {
                candidate: candidate.to_string(),
                sdp_m_line_index,
--- a/net/webrtc/src/livekit_signaller/imp.rs
+++ b/net/webrtc/src/livekit_signaller/imp.rs
@ -1,6 +1,6 @@
 // SPDX-License-Identifier: MPL-2.0

-use crate::signaller::{Signallable, SignallableImpl};
+use crate::signaller::{Signallable, SignallableImpl, WebRTCSignallerRole};

 use crate::utils::{wait_async, WaitError};
 use crate::RUNTIME;
@ -13,6 +13,7 @@ use gst::subclass::prelude::*;
 use once_cell::sync::Lazy;
 use serde::{Deserialize, Serialize};
 use std::collections::HashMap;
+use std::str::FromStr;
 use std::sync::{Arc, Mutex};
 use tokio::sync::oneshot;
 use tokio::task::JoinHandle;
@ -40,6 +41,9 @@ struct Settings {
    identity: Option<String>,
    room_name: Option<String>,
    auth_token: Option<String>,
+    role: WebRTCSignallerRole,
+    producer_peer_id: Option<String>,
+    excluded_produder_peer_ids: Vec<String>,
    timeout: u32,
 }

@ -53,6 +57,9 @@ impl Default for Settings {
            identity: Some("gstreamer".to_string()),
            room_name: None,
            auth_token: None,
+            role: WebRTCSignallerRole::default(),
+            producer_peer_id: None,
+            excluded_produder_peer_ids: vec![],
            timeout: DEFAULT_TRACK_PUBLISH_TIMEOUT,
        }
    }
@ -93,6 +100,99 @@ impl Signaller {
            .emit_by_name::<()>("error", &[&format!("Error: {msg}")]);
    }

+    fn role(&self) -> Option<WebRTCSignallerRole> {
+        self.settings.lock().map(|s| s.role).ok()
+    }
+
+    fn is_subscriber(&self) -> bool {
+        matches!(self.role(), Some(WebRTCSignallerRole::Consumer))
+    }
+
+    fn producer_peer_id(&self) -> Option<String> {
+        assert!(self.is_subscriber());
+        let settings = self.settings.lock().ok()?;
+        settings.producer_peer_id.clone()
+    }
+
+    fn auto_subscribe(&self) -> bool {
+        self.is_subscriber()
+            && self.producer_peer_id().is_none()
+            && self.excluded_producer_peer_ids_is_empty()
+    }
+
+    fn signal_target(&self) -> Option<proto::SignalTarget> {
+        match self.role()? {
+            WebRTCSignallerRole::Consumer => Some(proto::SignalTarget::Subscriber),
+            WebRTCSignallerRole::Producer => Some(proto::SignalTarget::Publisher),
+            _ => None,
+        }
+    }
+
+    fn excluded_producer_peer_ids_is_empty(&self) -> bool {
+        assert!(self.is_subscriber());
+        self.settings
+            .lock()
+            .unwrap()
+            .excluded_produder_peer_ids
+            .is_empty()
+    }
+
+    fn is_peer_excluded(&self, peer_id: &str) -> bool {
+        self.settings
+            .lock()
+            .unwrap()
+            .excluded_produder_peer_ids
+            .iter()
+            .any(|id| id == peer_id)
+    }
+
+    fn signal_client(&self) -> Option<Arc<signal_client::SignalClient>> {
+        let connection = self.connection.lock().unwrap();
+        Some(connection.as_ref()?.signal_client.clone())
+    }
+
+    fn require_signal_client(&self) -> Arc<signal_client::SignalClient> {
+        self.signal_client().unwrap()
+    }
+
+    async fn send_trickle_request(&self, candidate_init: &str) {
+        let Some(signal_client) = self.signal_client() else {
+            return;
+        };
+        let Some(target) = self.signal_target() else {
+            return;
+        };
+        signal_client
+            .send(proto::signal_request::Message::Trickle(
+                proto::TrickleRequest {
+                    candidate_init: candidate_init.to_string(),
+                    target: target as i32,
+                },
+            ))
+            .await;
+    }
+
+    async fn send_delayed_ice_candidates(&self) {
+        let Some(mut early_candidates) = self
+            .connection
+            .lock()
+            .unwrap()
+            .as_mut()
+            .and_then(|c| c.early_candidates.take())
+        else {
+            return;
+        };
+
+        while let Some(candidate_str) = early_candidates.pop() {
+            gst::debug!(
+                CAT,
+                imp: self,
+                "Sending delayed ice candidate {candidate_str:?}"
+            );
+            self.send_trickle_request(&candidate_str).await;
+        }
+    }
+
    async fn signal_task(&self, mut signal_events: signal_client::SignalEvents) {
        loop {
            match wait_async(&self.signal_task_canceller, signal_events.recv(), 0).await {
@ -136,10 +236,36 @@ impl Signaller {
                self.obj()
                    .emit_by_name::<()>("session-description", &[&"unique", &answer]);
            }
+
+            proto::signal_response::Message::Offer(offer) => {
+                if !self.is_subscriber() {
+                    gst::warning!(CAT, imp: self, "Ignoring subscriber offer in non-subscriber mode: {:?}", offer);
+                    return;
+                }
+                gst::debug!(CAT, imp: self, "Received subscriber offer: {:?}", offer);
+                let sdp = match gst_sdp::SDPMessage::parse_buffer(offer.sdp.as_bytes()) {
+                    Ok(sdp) => sdp,
+                    Err(_) => {
+                        self.raise_error("Couldn't parse Offer SDP".to_string());
+                        return;
+                    }
+                };
+                let offer = gst_webrtc::WebRTCSessionDescription::new(
+                    gst_webrtc::WebRTCSDPType::Offer,
+                    sdp,
+                );
+                self.obj()
+                    .emit_by_name::<()>("session-description", &[&"unique", &offer]);
+            }
+
            proto::signal_response::Message::Trickle(trickle) => {
                gst::debug!(CAT, imp: self, "Received ice_candidate {:?}", trickle);

-                if trickle.target() == proto::SignalTarget::Publisher {
+                let Some(target) = self.signal_target() else {
+                    return;
+                };
+
+                if target == trickle.target() {
                    if let Ok(json) =
                        serde_json::from_str::<IceCandidateJson>(&trickle.candidate_init)
                    {
@ -165,6 +291,17 @@ impl Signaller {
                }
            }

+            proto::signal_response::Message::Update(update) => {
+                if !self.is_subscriber() {
+                    gst::trace!(CAT, imp: self, "Ignoring update in non-subscriber mode: {:?}", update);
+                    return;
+                }
+                gst::debug!(CAT, imp: self, "Update: {:?}", update);
+                for participant in update.participants {
+                    self.on_participant(&participant, true)
+                }
+            }
+
            proto::signal_response::Message::Leave(leave) => {
                gst::debug!(CAT, imp: self, "Leave: {:?}", leave);
            }
@ -172,182 +309,36 @@ impl Signaller {
            _ => {}
        }
    }
-}
-
-impl SignallableImpl for Signaller {
-    fn start(&self) {
-        gst::debug!(CAT, imp: self, "Connecting");
-
-        let wsurl = if let Some(wsurl) = &self.settings.lock().unwrap().wsurl {
-            wsurl.clone()
-        } else {
-            self.raise_error("WebSocket URL must be set".to_string());
-            return;
-        };
-
-        let auth_token = {
-            let settings = self.settings.lock().unwrap();
-
-            if let Some(auth_token) = &settings.auth_token {
-                auth_token.clone()
-            } else if let (
-                Some(api_key),
-                Some(secret_key),
-                Some(identity),
-                Some(participant_name),
-                Some(room_name),
-            ) = (
-                &settings.api_key,
-                &settings.secret_key,
-                &settings.identity,
-                &settings.participant_name,
-                &settings.room_name,
-            ) {
-                let grants = VideoGrants {
-                    room_join: true,
-                    can_subscribe: false,
-                    room: room_name.clone(),
-                    ..Default::default()
-                };
-                let access_token = AccessToken::with_api_key(api_key, secret_key)
-                    .with_name(participant_name)
-                    .with_identity(identity)
-                    .with_grants(grants);
-                match access_token.to_jwt() {
-                    Ok(token) => token,
-                    Err(err) => {
-                        self.raise_error(format!(
-                            "{:?}",
-                            anyhow!("Could not create auth token {err}")
-                        ));
-                        return;
-                    }
-                }
-            } else {
-                self.raise_error("Either auth-token or (api-key and secret-key and identity and room-name) must be set".to_string());
-                return;
-            }
-        };
-
-        gst::debug!(CAT, imp: self, "We have an authentication token");

+    fn send_sdp_answer(&self, _session_id: &str, sessdesc: &gst_webrtc::WebRTCSessionDescription) {
        let weak_imp = self.downgrade();
+        let sessdesc = sessdesc.clone();
+
        RUNTIME.spawn(async move {
-            let imp = if let Some(imp) = weak_imp.upgrade() {
-                imp
-            } else {
-                return;
-            };
-
-            let options = signal_client::SignalOptions::default();
-            gst::debug!(CAT, imp: imp, "Connecting to {}", wsurl);
-
-            let res = signal_client::SignalClient::connect(&wsurl, &auth_token, options).await;
-            let (signal_client, join_response, signal_events) = match res {
-                Err(err) => {
-                    imp.obj()
-                        .emit_by_name::<()>("error", &[&format!("{:?}", anyhow!("Error: {err}"))]);
-                    return;
-                }
-                Ok(ok) => ok,
-            };
-            let signal_client = Arc::new(signal_client);
-
-            gst::debug!(
-                CAT,
-                imp: imp,
-                "Connected with JoinResponse: {:?}",
-                join_response
-            );
-
-            let weak_imp = imp.downgrade();
-            let signal_task = RUNTIME.spawn(async move {
-                if let Some(imp) = weak_imp.upgrade() {
-                    imp.signal_task(signal_events).await;
-                }
-            });
-
-            let weak_imp = imp.downgrade();
-            imp.obj().connect_closure(
-                "webrtcbin-ready",
-                false,
-                glib::closure!(|_signaler: &super::LiveKitSignaller,
-                                _consumer_identifier: &str,
-                                webrtcbin: &gst::Element| {
-                    gst::info!(CAT, "Adding data channels");
-                    let reliable_channel = webrtcbin.emit_by_name::<gst_webrtc::WebRTCDataChannel>(
-                        "create-data-channel",
-                        &[
-                            &"_reliable",
-                            &gst::Structure::builder("config")
-                                .field("ordered", true)
-                                .build(),
-                        ],
-                    );
-                    let lossy_channel = webrtcbin.emit_by_name::<gst_webrtc::WebRTCDataChannel>(
-                        "create-data-channel",
-                        &[
-                            &"_lossy",
-                            &gst::Structure::builder("config")
-                                .field("ordered", true)
-                                .field("max-retransmits", 0)
-                                .build(),
-                        ],
-                    );
-
-                    if let Some(imp) = weak_imp.upgrade() {
-                        let mut connection = imp.connection.lock().unwrap();
-                        if let Some(connection) = connection.as_mut() {
-                            connection.channels = Some(Channels {
-                                reliable_channel,
-                                lossy_channel,
-                            });
-                        }
-                    }
-                }),
-            );
-
-            let connection = Connection {
-                signal_client,
-                signal_task,
-                pending_tracks: Default::default(),
-                early_candidates: Some(Vec::new()),
-                channels: None,
-            };
-
-            if let Ok(mut sc) = imp.connection.lock() {
-                *sc = Some(connection);
+            if let Some(imp) = weak_imp.upgrade() {
+                let sdp = sessdesc.sdp();
+                gst::debug!(CAT, imp: imp, "Sending SDP {:?} now", &sdp);
+                let signal_client = imp.require_signal_client();
+                signal_client
+                    .send(proto::signal_request::Message::Answer(
+                        proto::SessionDescription {
+                            r#type: "answer".to_string(),
+                            sdp: sdp.to_string(),
+                        },
+                    ))
+                    .await;
+                imp.send_delayed_ice_candidates().await;
            }
-
-            imp.obj().emit_by_name::<()>(
-                "session-requested",
-                &[
-                    &"unique",
-                    &"unique",
-                    &None::<gst_webrtc::WebRTCSessionDescription>,
-                ],
-            );
        });
    }

-    fn send_sdp(&self, _session_id: &str, sessdesc: &gst_webrtc::WebRTCSessionDescription) {
-        gst::debug!(CAT, imp: self, "Created offer SDP {:#?}", sessdesc.sdp());
-
-        assert!(sessdesc.type_() == gst_webrtc::WebRTCSDPType::Offer);
-
+    fn send_sdp_offer(&self, _session_id: &str, sessdesc: &gst_webrtc::WebRTCSessionDescription) {
        let weak_imp = self.downgrade();
        let sessdesc = sessdesc.clone();
        RUNTIME.spawn(async move {
            if let Some(imp) = weak_imp.upgrade() {
                let sdp = sessdesc.sdp();
-                let signal_client = imp
-                    .connection
-                    .lock()
-                    .unwrap()
-                    .as_ref()
-                    .unwrap()
-                    .signal_client
-                    .clone();
+                let signal_client = imp.require_signal_client();
                let timeout = imp.settings.lock().unwrap().timeout;

                for media in sdp.medias() {
@ -457,35 +448,271 @@ impl SignallableImpl for Signaller {
                    .await;

                if let Some(imp) = weak_imp.upgrade() {
-                    let early_candidates =
-                        if let Some(connection) = &mut *imp.connection.lock().unwrap() {
-                            connection.early_candidates.take()
-                        } else {
-                            None
-                        };
-
-                    if let Some(mut early_candidates) = early_candidates {
-                        while let Some(candidate_str) = early_candidates.pop() {
-                            gst::debug!(
-                                CAT,
-                                imp: imp,
-                                "Sending delayed ice candidate {candidate_str:?}"
-                            );
-                            signal_client
-                                .send(proto::signal_request::Message::Trickle(
-                                    proto::TrickleRequest {
-                                        candidate_init: candidate_str,
-                                        target: proto::SignalTarget::Publisher as i32,
-                                    },
-                                ))
-                                .await;
-                        }
-                    }
+                    imp.send_delayed_ice_candidates().await;
                }
            }
        });
    }

+    fn on_participant(&self, participant: &proto::ParticipantInfo, new_connection: bool) {
+        gst::debug!(CAT, imp: self, "{:?}", participant);
+        if !participant.is_publisher {
+            return;
+        }
+        let peer_sid = &participant.sid;
+        let peer_identity = &participant.identity;
+        match self.producer_peer_id() {
+            Some(id) if id == *peer_sid => {
+                gst::debug!(CAT, imp: self, "matching peer sid {id:?}");
+            }
+            Some(id) if id == *peer_identity => {
+                gst::debug!(CAT, imp: self, "matching peer identity {id:?}");
+            }
+            None => {
+                if self.is_peer_excluded(peer_sid) || self.is_peer_excluded(peer_identity) {
+                    gst::debug!(CAT, imp: self, "ignoring excluded peer {participant:?}");
+                    return;
+                }
+                gst::debug!(CAT, imp: self, "catch-all mode, matching {participant:?}");
+            }
+            _ => return,
+        }
+        let meta = Some(&participant.metadata)
+            .filter(|meta| !meta.is_empty())
+            .and_then(|meta| gst::Structure::from_str(meta).ok());
+        match participant.state {
+            x if x == proto::participant_info::State::Active as i32 => {
+                let track_sids = participant
+                    .tracks
+                    .iter()
+                    .filter(|t| !t.muted)
+                    .map(|t| t.sid.clone())
+                    .collect::<Vec<_>>();
+                let update = proto::UpdateSubscription {
+                    track_sids: track_sids.clone(),
+                    subscribe: true,
+                    participant_tracks: vec![proto::ParticipantTracks {
+                        participant_sid: participant.sid.clone(),
+                        track_sids: track_sids.clone(),
+                    }],
+                };
+                let update = proto::signal_request::Message::Subscription(update);
+                let weak_imp = self.downgrade();
+                let peer_sid = peer_sid.clone();
+                RUNTIME.spawn(async move {
+                    let imp = match weak_imp.upgrade() {
+                        Some(imp) => imp,
+                        None => return,
+                    };
+                    let signal_client = imp.require_signal_client();
+                    signal_client.send(update).await;
+                    imp.obj()
+                        .emit_by_name::<()>("producer-added", &[&peer_sid, &meta, &new_connection]);
+                });
+            }
+            _ => {
+                self.obj()
+                    .emit_by_name::<()>("producer-removed", &[&peer_sid, &meta]);
+            }
+        }
+    }
+
+    async fn close_signal_client(signal_client: &signal_client::SignalClient) {
+        signal_client
+            .send(proto::signal_request::Message::Leave(proto::LeaveRequest {
+                can_reconnect: false,
+                reason: proto::DisconnectReason::ClientInitiated as i32,
+                ..Default::default()
+            }))
+            .await;
+        signal_client.close().await;
+    }
+}
+
+impl SignallableImpl for Signaller {
+    fn start(&self) {
+        gst::debug!(CAT, imp: self, "Connecting");
+
+        let wsurl = if let Some(wsurl) = &self.settings.lock().unwrap().wsurl {
+            wsurl.clone()
+        } else {
+            self.raise_error("WebSocket URL must be set".to_string());
+            return;
+        };
+
+        let auth_token = {
+            let settings = self.settings.lock().unwrap();
+            let role = settings.role;
+
+            if let Some(auth_token) = &settings.auth_token {
+                auth_token.clone()
+            } else if let (
+                Some(api_key),
+                Some(secret_key),
+                Some(identity),
+                Some(participant_name),
+                Some(room_name),
+            ) = (
+                &settings.api_key,
+                &settings.secret_key,
+                &settings.identity,
+                &settings.participant_name,
+                &settings.room_name,
+            ) {
+                let grants = VideoGrants {
+                    room_join: true,
+                    can_subscribe: role == WebRTCSignallerRole::Consumer,
+                    room: room_name.clone(),
+                    ..Default::default()
+                };
+                let access_token = AccessToken::with_api_key(api_key, secret_key)
+                    .with_name(participant_name)
+                    .with_identity(identity)
+                    .with_grants(grants);
+                match access_token.to_jwt() {
+                    Ok(token) => token,
+                    Err(err) => {
+                        self.raise_error(format!(
+                            "{:?}",
+                            anyhow!("Could not create auth token {err}")
+                        ));
+                        return;
+                    }
+                }
+            } else {
+                self.raise_error("Either auth-token or (api-key and secret-key and identity and room-name) must be set".to_string());
+                return;
+            }
+        };
+
+        gst::debug!(CAT, imp: self, "We have an authentication token");
+
+        let weak_imp = self.downgrade();
+        RUNTIME.spawn(async move {
+            let imp = if let Some(imp) = weak_imp.upgrade() {
+                imp
+            } else {
+                return;
+            };
+
+            let options = signal_client::SignalOptions {
+                auto_subscribe: imp.auto_subscribe(),
+                ..Default::default()
+            };
+            gst::debug!(CAT, imp: imp, "Connecting to {}", wsurl);
+
+            let res = signal_client::SignalClient::connect(&wsurl, &auth_token, options).await;
+            let (signal_client, join_response, signal_events) = match res {
+                Err(err) => {
+                    imp.obj()
+                        .emit_by_name::<()>("error", &[&format!("{:?}", anyhow!("Error: {err}"))]);
+                    return;
+                }
+                Ok(ok) => ok,
+            };
+            let signal_client = Arc::new(signal_client);
+
+            gst::debug!(
+                CAT,
+                imp: imp,
+                "Connected with JoinResponse: {:?}",
+                join_response
+            );
+
+            let weak_imp = imp.downgrade();
+            let signal_task = RUNTIME.spawn(async move {
+                if let Some(imp) = weak_imp.upgrade() {
+                    imp.signal_task(signal_events).await;
+                }
+            });
+
+            if imp.is_subscriber() {
+                imp.obj()
+                    .emit_by_name::<()>("session-started", &[&"unique", &"unique"]);
+                for participant in &join_response.other_participants {
+                    imp.on_participant(participant, false)
+                }
+            }
+
+            let weak_imp = imp.downgrade();
+            imp.obj().connect_closure(
+                "webrtcbin-ready",
+                false,
+                glib::closure!(|_signaller: &super::LiveKitSignaller,
+                                _consumer_identifier: &str,
+                                webrtcbin: &gst::Element| {
+                    gst::info!(CAT, "Adding data channels");
+                    let reliable_channel = webrtcbin.emit_by_name::<gst_webrtc::WebRTCDataChannel>(
+                        "create-data-channel",
+                        &[
+                            &"_reliable",
+                            &gst::Structure::builder("config")
+                                .field("ordered", true)
+                                .build(),
+                        ],
+                    );
+                    let lossy_channel = webrtcbin.emit_by_name::<gst_webrtc::WebRTCDataChannel>(
+                        "create-data-channel",
+                        &[
+                            &"_lossy",
+                            &gst::Structure::builder("config")
+                                .field("ordered", true)
+                                .field("max-retransmits", 0)
+                                .build(),
+                        ],
+                    );
+
+                    if let Some(imp) = weak_imp.upgrade() {
+                        let mut connection = imp.connection.lock().unwrap();
+                        if let Some(connection) = connection.as_mut() {
+                            connection.channels = Some(Channels {
+                                reliable_channel,
+                                lossy_channel,
+                            });
+                        }
+                    }
+                }),
+            );
+
+            let connection = Connection {
+                signal_client,
+                signal_task,
+                pending_tracks: Default::default(),
+                early_candidates: Some(Vec::new()),
+                channels: None,
+            };
+
+            if let Ok(mut sc) = imp.connection.lock() {
+                *sc = Some(connection);
+            }
+
+            imp.obj().emit_by_name::<()>(
+                "session-requested",
+                &[
+                    &"unique",
+                    &"unique",
+                    &None::<gst_webrtc::WebRTCSessionDescription>,
+                ],
+            );
+        });
+    }
+
+    fn send_sdp(&self, session_id: &str, sessdesc: &gst_webrtc::WebRTCSessionDescription) {
+        gst::debug!(CAT, imp: self, "Created SDP {:?}", sessdesc.sdp());
+
+        match sessdesc.type_() {
+            gst_webrtc::WebRTCSDPType::Offer => {
+                self.send_sdp_offer(session_id, sessdesc);
+            }
+            gst_webrtc::WebRTCSDPType::Answer => {
+                self.send_sdp_answer(session_id, sessdesc);
+            }
+            _ => {
+                gst::debug!(CAT, imp: self, "Ignoring SDP {:?}", sessdesc.sdp());
+            }
+        }
+    }
+
    fn add_ice(
        &self,
        _session_id: &str,
@ -514,20 +741,7 @@ impl SignallableImpl for Signaller {
        let imp = self.downgrade();
        RUNTIME.spawn(async move {
            if let Some(imp) = imp.upgrade() {
-                let signal_client = if let Some(connection) = &mut *imp.connection.lock().unwrap() {
-                    connection.signal_client.clone()
-                } else {
-                    return;
-                };
-
-                signal_client
-                    .send(proto::signal_request::Message::Trickle(
-                        proto::TrickleRequest {
-                            candidate_init: candidate_str,
-                            target: proto::SignalTarget::Publisher as i32,
-                        },
-                    ))
-                    .await;
+                imp.send_trickle_request(&candidate_str).await;
            };
        });
    }
@ -542,7 +756,7 @@ impl SignallableImpl for Signaller {

        if let Some(connection) = self.connection.lock().unwrap().take() {
            block_on(connection.signal_task).unwrap();
-            block_on(connection.signal_client.close());
+            block_on(Self::close_signal_client(&connection.signal_client));
        }
    }

@ -615,6 +829,22 @@ impl ObjectImpl for Signaller {
                    .blurb("Lossy Data Channel object.")
                    .flags(glib::ParamFlags::READABLE)
                    .build(),
+                glib::ParamSpecEnum::builder_with_default("role", WebRTCSignallerRole::default())
+                    .nick("Sigaller Role")
+                    .blurb("Whether this signaller acts as either a Consumer or Producer. Listener is not currently supported.")
+                    .flags(glib::ParamFlags::READWRITE)
+                    .build(),
+                glib::ParamSpecString::builder("producer-peer-id")
+                    .nick("Producer Peer ID")
+                    .blurb("When in Consumer Role, the signaller will subscribe to this peer's tracks.")
+                    .flags(glib::ParamFlags::READWRITE)
+                    .build(),
+                gst::ParamSpecArray::builder("excluded-producer-peer-ids")
+                    .nick("Excluded Producer Peer IDs")
+                    .blurb("When in Consumer Role, the signaller will not subscribe to these peers' tracks.")
+                    .flags(glib::ParamFlags::READWRITE)
+                    .element_spec(&glib::ParamSpecString::builder("producer-peer-id").build())
+                    .build(),
            ]
        });

@ -648,6 +878,18 @@ impl ObjectImpl for Signaller {
            "timeout" => {
                settings.timeout = value.get().unwrap();
            }
+            "role" => settings.role = value.get().unwrap(),
+            "producer-peer-id" => settings.producer_peer_id = value.get().unwrap(),
+            "excluded-producer-peer-ids" => {
+                settings.excluded_produder_peer_ids = value
+                    .get::<gst::ArrayRef>()
+                    .expect("type checked upstream")
+                    .as_slice()
+                    .iter()
+                    .filter_map(|id| id.get::<&str>().ok())
+                    .map(|id| id.to_string())
+                    .collect::<Vec<String>>()
+            }
            _ => unimplemented!(),
        }
    }
@ -679,6 +921,11 @@ impl ObjectImpl for Signaller {
                };
                channel.to_value()
            }
+            "role" => settings.role.to_value(),
+            "producer-peer-id" => settings.producer_peer_id.to_value(),
+            "excluded-producer-peer-ids" => {
+                gst::Array::new(&settings.excluded_produder_peer_ids).to_value()
+            }
            _ => unimplemented!(),
        }
    }
--- a/net/webrtc/src/livekit_signaller/mod.rs
+++ b/net/webrtc/src/livekit_signaller/mod.rs
@ -1,6 +1,6 @@
 // SPDX-License-Identifier: MPL-2.0

-use crate::signaller::Signallable;
+use crate::signaller::{Signallable, WebRTCSignallerRole};
 use gst::glib;

 mod imp;
@ -9,6 +9,20 @@ glib::wrapper! {
    pub struct LiveKitSignaller(ObjectSubclass<imp::Signaller>) @implements Signallable;
 }

+impl LiveKitSignaller {
+    fn new(role: WebRTCSignallerRole) -> Self {
+        glib::Object::builder().property("role", role).build()
+    }
+
+    pub fn new_consumer() -> Self {
+        Self::new(WebRTCSignallerRole::Consumer)
+    }
+
+    pub fn new_producer() -> Self {
+        Self::new(WebRTCSignallerRole::Producer)
+    }
+}
+
 impl Default for LiveKitSignaller {
    fn default() -> Self {
        glib::Object::new()
--- a/net/webrtc/src/webrtcsink/imp.rs
+++ b/net/webrtc/src/webrtcsink/imp.rs
@ -20,7 +20,9 @@ use std::ops::Mul;
 use std::sync::{mpsc, Arc, Condvar, Mutex};

 use super::homegrown_cc::CongestionController;
-use super::{WebRTCSinkCongestionControl, WebRTCSinkError, WebRTCSinkMitigationMode};
+use super::{
+    WebRTCSinkCongestionControl, WebRTCSinkError, WebRTCSinkMitigationMode, WebRTCSinkPad,
+};
 use crate::aws_kvs_signaller::AwsKvsSignaller;
 use crate::janusvr_signaller::JanusVRSignaller;
 use crate::livekit_signaller::LiveKitSignaller;
@ -186,7 +188,7 @@ impl futures::stream::FusedStream for CustomBusStream {
 /// Wrapper around our sink pads
 #[derive(Debug, Clone)]
 struct InputStream {
-    sink_pad: gst::GhostPad,
+    sink_pad: WebRTCSinkPad,
    producer: Option<StreamProducer>,
    /// The (fixed) caps coming in
    in_caps: Option<gst::Caps>,
@ -233,6 +235,8 @@ pub struct VideoEncoder {
    session_id: String,
    mitigation_mode: WebRTCSinkMitigationMode,
    pub transceiver: gst_webrtc::WebRTCRTPTransceiver,
+    /// name of the sink pad feeding this encoder
+    stream_name: String,
 }

 struct Session {
@ -554,8 +558,8 @@ fn make_converter_for_video_caps(caps: &gst::Caps, codec: &Codec) -> Result<gst:
                // NVIDIA V4L2 encoders require NVMM memory as input and that requires using the
                // corresponding converter
                || codec
-                    .encoder_factory()
-                    .map_or(false, |factory| factory.name().starts_with("nvv4l2"))
+                .encoder_factory()
+                .map_or(false, |factory| factory.name().starts_with("nvv4l2"))
            {
                let queue = make_element("queue", None)?;
                let nvconvert = if let Ok(nvconvert) = make_element("nvvideoconvert", None) {
@ -918,6 +922,7 @@ impl VideoEncoder {
        session_id: &str,
        codec_name: &str,
        transceiver: gst_webrtc::WebRTCRTPTransceiver,
+        stream_name: String,
    ) -> Option<Self> {
        let halved_framerate = video_info.fps().mul(gst::Fraction::new(1, 2));
        Some(Self {
@ -936,6 +941,7 @@ impl VideoEncoder {
            session_id: session_id.to_string(),
            mitigation_mode: WebRTCSinkMitigationMode::NONE,
            transceiver,
+            stream_name,
        })
    }

@ -1320,6 +1326,7 @@ impl Session {
                &self.id,
                codec.caps.structure(0).unwrap().name(),
                transceiver,
+                stream_name.clone(),
            ) {
                match self.cc_info.heuristic {
                    WebRTCSinkCongestionControl::Disabled => {
@ -1434,6 +1441,10 @@ impl InputStream {
            ),
        )
    }
+
+    fn msid(&self) -> Option<String> {
+        self.sink_pad.property("msid")
+    }
 }

 impl NavigationEventHandler {
@ -1537,12 +1548,13 @@ impl BaseWebRTCSink {
        match extension_configuration_type {
            ExtensionConfigurationType::Auto => {
                // GstRTPBasePayload::extensions property is only available since GStreamer 1.24
-                if !payloader.has_property("extensions", Some(gst::Array::static_type()))
-                    && self.has_connected_payloader_setup_slots()
-                {
-                    gst::warning!(CAT, "'extensions' property is not available: TWCC extension ID will default to 1. \
+                if !payloader.has_property("extensions", Some(gst::Array::static_type())) {
+                    if self.has_connected_payloader_setup_slots() {
+                        gst::warning!(CAT, "'extensions' property is not available: TWCC extension ID will default to 1. \
        Application code must ensure to pick non-conflicting IDs for any additionally configured extensions. \
        Please consider updating GStreamer to 1.24.");
+                    }
+
                    return Some(1);
                }

@ -1738,6 +1750,11 @@ impl BaseWebRTCSink {
                return;
            };

+            if let Some(msid) = stream.msid() {
+                gst::trace!(CAT, obj: element, "forwarding msid={msid:?} to webrtcbin sinkpad");
+                pad.set_property("msid", &msid);
+            }
+
            let transceiver = pad.property::<gst_webrtc::WebRTCRTPTransceiver>("transceiver");

            transceiver.set_property(
@ -1867,7 +1884,7 @@ impl BaseWebRTCSink {
                        gst::StreamError::Failed,
                        ["Signalling error: {}", error]
                    );
-                })
+                }),
            ),

            request_meta: signaler.connect_closure(
@ -1877,7 +1894,7 @@ impl BaseWebRTCSink {
                    let meta = instance.imp().settings.lock().unwrap().meta.clone();

                    meta
-                })
+                }),
            ),

            session_requested: signaler.connect_closure(
@ -1887,7 +1904,7 @@ impl BaseWebRTCSink {
                    if let Err(err) = instance.imp().start_session(session_id, peer_id, offer) {
                        gst::warning!(CAT, "{}", err);
                    }
-                })
+                }),
            ),

            session_description: signaler.connect_closure(
@ -1908,9 +1925,9 @@ impl BaseWebRTCSink {
            ),

            handle_ice: signaler.connect_closure(
-                    "handle-ice",
-                    false,
-                    glib::closure!(@watch instance => move |
+                "handle-ice",
+                false,
+                glib::closure!(@watch instance => move |
                        _signaler: glib::Object,
                        session_id: &str,
                        sdp_m_line_index: u32,
@ -1920,7 +1937,7 @@ impl BaseWebRTCSink {
                            .imp()
                            .handle_ice(session_id, Some(sdp_m_line_index), None, candidate);
                    }),
-                ),
+            ),

            session_ended: signaler.connect_closure(
                "session-ended",
@ -1930,7 +1947,7 @@ impl BaseWebRTCSink {
                        gst::warning!(CAT, "{}", err);
                    }
                    false
-                })
+                }),
            ),

            shutdown: signaler.connect_closure(
@ -1938,7 +1955,7 @@ impl BaseWebRTCSink {
                false,
                glib::closure!(@watch instance => move |_signaler: glib::Object|{
                    instance.imp().shutdown(instance);
-                })
+                }),
            ),
        });
    }
@ -2585,7 +2602,7 @@ impl BaseWebRTCSink {
        if session.congestion_controller.is_some() {
            let session_id_str = session_id.to_string();
            rtpbin.connect_closure("on-new-ssrc", true,
-                glib::closure!(@weak-allow-none element,
+                                   glib::closure!(@weak-allow-none element,
                                => move |rtpbin: gst::Object, session_id: u32, _src: u32| {
                        let rtp_session = rtpbin.emit_by_name::<gst::Element>("get-session", &[&session_id]);

@ -2606,8 +2623,8 @@ impl BaseWebRTCSink {
                                ));
                            }
                        }
-                    })
-                );
+                    }),
+            );
        }

        let clock = element.clock();
@ -3447,6 +3464,37 @@ impl BaseWebRTCSink {
        )
    }

+    /// Check if the caps of a sink pad can be changed from `current` to `new` without requiring a WebRTC renegotiation
+    fn input_caps_change_allowed(&self, current: &gst::CapsRef, new: &gst::CapsRef) -> bool {
+        let Some(current) = current.structure(0) else {
+            return false;
+        };
+        let Some(new) = new.structure(0) else {
+            return false;
+        };
+
+        if current.name() != new.name() {
+            return false;
+        }
+
+        let mut current = current.to_owned();
+        let mut new = new.to_owned();
+
+        // Allow changes of fields which should not be part of the SDP, and so can be updated without requiring
+        // a renegotiation.
+        let caps_type = current.name();
+        if caps_type.starts_with("video/") {
+            const VIDEO_ALLOWED_CHANGES: &[&str] = &["width", "height", "framerate"];
+
+            current.remove_fields(VIDEO_ALLOWED_CHANGES.iter().copied());
+            new.remove_fields(VIDEO_ALLOWED_CHANGES.iter().copied());
+        } else if caps_type.starts_with("audio/") {
+            // TODO
+        }
+
+        current == new
+    }
+
    fn sink_event(
        &self,
        pad: &gst::Pad,
@ -3457,10 +3505,7 @@ impl BaseWebRTCSink {

        if let EventView::Caps(e) = event.view() {
            if let Some(caps) = pad.current_caps() {
-                if caps.is_strictly_equal(e.caps()) {
-                    // Nothing changed
-                    return true;
-                } else {
+                if !self.input_caps_change_allowed(&caps, e.caps()) {
                    gst::error!(
                        CAT,
                        obj: pad,
@ -3470,30 +3515,44 @@ impl BaseWebRTCSink {
                    );
                    return false;
                }
-            } else {
-                gst::info!(CAT, obj: pad, "Received caps event {:?}", e);
+            }
+            gst::info!(CAT, obj: pad, "Received caps event {:?}", e);

-                self.state
-                    .lock()
-                    .unwrap()
-                    .streams
-                    .iter_mut()
-                    .for_each(|(_, stream)| {
-                        if stream.sink_pad.upcast_ref::<gst::Pad>() == pad {
-                            // We do not want VideoInfo to consider max-framerate
-                            // when computing fps, so we strip it away here
-                            let mut caps = e.caps().to_owned();
-                            {
-                                let mut_caps = caps.get_mut().unwrap();
-                                if let Some(s) = mut_caps.structure_mut(0) {
-                                    if s.has_name("video/x-raw") {
-                                        s.remove_field("max-framerate");
-                                    }
-                                }
+            let mut state = self.state.lock().unwrap();
+
+            state.streams.iter_mut().for_each(|(_, stream)| {
+                if stream.sink_pad.upcast_ref::<gst::Pad>() == pad {
+                    // We do not want VideoInfo to consider max-framerate
+                    // when computing fps, so we strip it away here
+                    let mut caps = e.caps().to_owned();
+                    {
+                        let mut_caps = caps.get_mut().unwrap();
+                        if let Some(s) = mut_caps.structure_mut(0) {
+                            if s.has_name("video/x-raw") {
+                                s.remove_field("max-framerate");
                            }
-                            stream.in_caps = Some(caps.to_owned());
                        }
-                    });
+                    }
+                    stream.in_caps = Some(caps.to_owned());
+                }
+            });
+
+            if e.caps().structure(0).unwrap().name().starts_with("video/") {
+                if let Ok(video_info) = gst_video::VideoInfo::from_caps(e.caps()) {
+                    // update video encoder info used when downscaling/downsampling the input
+                    let stream_name = pad.name().to_string();
+
+                    state
+                        .sessions
+                        .values_mut()
+                        .flat_map(|session| session.unwrap_mut().encoders.iter_mut())
+                        .filter(|encoder| encoder.stream_name == stream_name)
+                        .for_each(|encoder| {
+                            encoder.halved_framerate =
+                                video_info.fps().mul(gst::Fraction::new(1, 2));
+                            encoder.video_info = video_info.clone();
+                        });
+                }
            }
        }

@ -3718,7 +3777,7 @@ impl ObjectImpl for BaseWebRTCSink {
                    .mutable_ready()
                    .build(),
                glib::ParamSpecObject::builder::<Signallable>("signaller")
-                    .flags(glib::ParamFlags::READABLE  | gst::PARAM_FLAG_MUTABLE_READY)
+                    .flags(glib::ParamFlags::READABLE | gst::PARAM_FLAG_MUTABLE_READY)
                    .blurb("The Signallable object to use to handle WebRTC Signalling")
                    .build(),
            ]
@ -4071,11 +4130,12 @@ impl ElementImpl for BaseWebRTCSink {
                caps_builder = caps_builder.structure(codec.caps.structure(0).unwrap().to_owned());
            }

-            let video_pad_template = gst::PadTemplate::new(
+            let video_pad_template = gst::PadTemplate::with_gtype(
                "video_%u",
                gst::PadDirection::Sink,
                gst::PadPresence::Request,
                &caps_builder.build(),
+                WebRTCSinkPad::static_type(),
            )
            .unwrap();

@ -4084,11 +4144,12 @@ impl ElementImpl for BaseWebRTCSink {
            for codec in Codecs::audio_codecs() {
                caps_builder = caps_builder.structure(codec.caps.structure(0).unwrap().to_owned());
            }
-            let audio_pad_template = gst::PadTemplate::new(
+            let audio_pad_template = gst::PadTemplate::with_gtype(
                "audio_%u",
                gst::PadDirection::Sink,
                gst::PadPresence::Request,
                &caps_builder.build(),
+                WebRTCSinkPad::static_type(),
            )
            .unwrap();

@ -4127,13 +4188,13 @@ impl ElementImpl for BaseWebRTCSink {
            (name, false)
        };

-        let sink_pad = gst::GhostPad::builder_from_template(templ)
+        let sink_pad = gst::PadBuilder::<WebRTCSinkPad>::from_template(templ)
            .name(name.as_str())
            .chain_function(|pad, parent, buffer| {
                BaseWebRTCSink::catch_panic_pad_function(
                    parent,
                    || Err(gst::FlowError::Error),
-                    |this| this.chain(pad, buffer),
+                    |this| this.chain(pad.upcast_ref(), buffer),
                )
            })
            .event_function(|pad, parent, event| {
@ -4250,7 +4311,7 @@ impl ChildProxyImpl for BaseWebRTCSink {
    fn child_by_name(&self, name: &str) -> Option<glib::Object> {
        match name {
            "signaller" => Some(self.settings.lock().unwrap().signaller.clone().upcast()),
-            _ => None,
+            _ => self.obj().static_pad(name).map(|pad| pad.upcast()),
        }
    }
 }
@ -4393,7 +4454,7 @@ impl ObjectImpl for LiveKitWebRTCSink {
        let element = self.obj();
        let ws = element.upcast_ref::<super::BaseWebRTCSink>().imp();

-        let _ = ws.set_signaller(LiveKitSignaller::default().upcast());
+        let _ = ws.set_signaller(LiveKitSignaller::new_producer().upcast());
    }
 }

--- a/net/webrtc/src/webrtcsink/mod.rs
+++ b/net/webrtc/src/webrtcsink/mod.rs
@ -39,11 +39,16 @@ use gst::subclass::prelude::*;
 mod homegrown_cc;

 mod imp;
+mod pad;

 glib::wrapper! {
    pub struct BaseWebRTCSink(ObjectSubclass<imp::BaseWebRTCSink>) @extends gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
 }

+glib::wrapper! {
+    pub struct WebRTCSinkPad(ObjectSubclass<pad::WebRTCSinkPad>) @extends gst::GhostPad, gst::ProxyPad, gst::Pad, gst::Object;
+}
+
 glib::wrapper! {
    pub struct WebRTCSink(ObjectSubclass<imp::WebRTCSink>) @extends BaseWebRTCSink, gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
 }
@ -124,6 +129,7 @@ enum WebRTCSinkMitigationMode {
 }

 pub fn register(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
+    WebRTCSinkPad::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
    BaseWebRTCSink::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
    WebRTCSinkCongestionControl::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
    gst::Element::register(
--- a/net/webrtc/src/webrtcsink/pad.rs
+++ b/net/webrtc/src/webrtcsink/pad.rs
@ -0,0 +1,57 @@
+// SPDX-License-Identifier: MPL-2.0
+
+use gst::{glib, prelude::*, subclass::prelude::*};
+use once_cell::sync::Lazy;
+use std::sync::Mutex;
+
+#[derive(Default)]
+pub struct WebRTCSinkPad {
+    settings: Mutex<Settings>,
+}
+
+#[derive(Debug, Default)]
+struct Settings {
+    msid: Option<String>,
+}
+
+#[glib::object_subclass]
+impl ObjectSubclass for WebRTCSinkPad {
+    const NAME: &'static str = "GstWebRTCSinkPad";
+    type Type = super::WebRTCSinkPad;
+    type ParentType = gst::GhostPad;
+}
+
+impl ObjectImpl for WebRTCSinkPad {
+    fn properties() -> &'static [glib::ParamSpec] {
+        static PROPS: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
+            vec![glib::ParamSpecString::builder("msid")
+                .flags(glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY)
+                .blurb("Remote MediaStream ID in use for this pad")
+                .build()]
+        });
+        PROPS.as_ref()
+    }
+    fn set_property(&self, _id: usize, value: &glib::Value, pspec: &glib::ParamSpec) {
+        let mut settings = self.settings.lock().unwrap();
+        match pspec.name() {
+            "msid" => {
+                settings.msid = value
+                    .get::<Option<String>>()
+                    .expect("type checked upstream")
+            }
+            name => panic!("no writable property {name:?}"),
+        }
+    }
+    fn property(&self, _id: usize, pspec: &glib::ParamSpec) -> glib::Value {
+        let settings = self.settings.lock().unwrap();
+        match pspec.name() {
+            "msid" => settings.msid.to_value(),
+            name => panic!("no readable property {name:?}"),
+        }
+    }
+}
+
+impl GstObjectImpl for WebRTCSinkPad {}
+impl PadImpl for WebRTCSinkPad {}
+impl ProxyPadImpl for WebRTCSinkPad {}
+impl GhostPadImpl for WebRTCSinkPad {}
--- a/net/webrtc/src/webrtcsrc/imp.rs
+++ b/net/webrtc/src/webrtcsrc/imp.rs
@ -2,6 +2,7 @@

 use gst::prelude::*;

+use crate::livekit_signaller::LiveKitSignaller;
 use crate::signaller::{prelude::*, Signallable, Signaller};
 use crate::utils::{Codec, Codecs, NavigationEvent, AUDIO_CAPS, RTP_CAPS, VIDEO_CAPS};
 use crate::webrtcsrc::WebRTCSrcPad;
@ -21,6 +22,7 @@ use url::Url;

 const DEFAULT_STUN_SERVER: Option<&str> = Some("stun://stun.l.google.com:19302");
 const DEFAULT_ENABLE_DATA_CHANNEL_NAVIGATION: bool = false;
+const DEFAULT_DO_RETRANSMISSION: bool = true;

 static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
    gst::DebugCategory::new(
@ -38,6 +40,7 @@ struct Settings {
    video_codecs: Vec<Codec>,
    audio_codecs: Vec<Codec>,
    enable_data_channel_navigation: bool,
+    do_retransmission: bool,
 }

 #[derive(Default)]
@ -112,7 +115,13 @@ impl ObjectImpl for BaseWebRTCSrc {
                    .default_value(DEFAULT_ENABLE_DATA_CHANNEL_NAVIGATION)
                    .mutable_ready()
                    .build(),
-            ]
+                glib::ParamSpecBoolean::builder("do-retransmission")
+                    .nick("Enable retransmission")
+                    .blurb("Send retransmission events upstream when a packet is late")
+                    .default_value(DEFAULT_DO_RETRANSMISSION)
+                    .mutable_ready()
+                    .build(),
+             ]
        });

        PROPS.as_ref()
@ -170,6 +179,10 @@ impl ObjectImpl for BaseWebRTCSrc {
                let mut settings = self.settings.lock().unwrap();
                settings.enable_data_channel_navigation = value.get::<bool>().unwrap();
            }
+            "do-retransmission" => {
+                let mut settings = self.settings.lock().unwrap();
+                settings.do_retransmission = value.get::<bool>().unwrap();
+            }
            _ => unimplemented!(),
        }
    }
@ -202,6 +215,7 @@ impl ObjectImpl for BaseWebRTCSrc {
                let settings = self.settings.lock().unwrap();
                settings.enable_data_channel_navigation.to_value()
            }
+            "do-retransmission" => self.settings.lock().unwrap().do_retransmission.to_value(),
            name => panic!("{} getter not implemented", name),
        }
    }
@ -211,7 +225,7 @@ impl ObjectImpl for BaseWebRTCSrc {
            vec![
                /**
                 * BaseWebRTCSrc::request-encoded-filter:
-                 * @producer_id: Identifier of the producer
+                 * @producer_id: (nullable): Identifier of the producer
                 * @pad_name: The name of the output pad
                 * @allowed_caps: the allowed caps for the output pad
                 *
@ -269,6 +283,7 @@ impl Default for Settings {
                .filter(|codec| codec.has_decoder())
                .collect(),
            enable_data_channel_navigation: DEFAULT_ENABLE_DATA_CHANNEL_NAVIGATION,
+            do_retransmission: DEFAULT_DO_RETRANSMISSION,
        }
    }
 }
@ -404,7 +419,11 @@ impl BaseWebRTCSrc {
            .expect("Adding ghostpad to the bin should always work");

        if let Some(srcpad) = srcpad {
-            let producer_id = self.signaller().property::<String>("producer-peer-id");
+            let producer_id = self
+                .signaller()
+                .property::<Option<String>>("producer-peer-id")
+                .or_else(|| pad.property("msid"));
+
            let encoded_filter = self.obj().emit_by_name::<Option<gst::Element>>(
                "request-encoded-filter",
                &[&producer_id, &srcpad.name(), &srcpad.allowed_caps()],
@ -760,14 +779,17 @@ impl BaseWebRTCSrc {
        let direction = gst_webrtc::WebRTCRTPTransceiverDirection::Recvonly;
        let webrtcbin = self.webrtcbin();
        for (i, media) in sdp.medias().enumerate() {
-            let codec_names = {
+            let (codec_names, do_retransmission) = {
                let settings = self.settings.lock().unwrap();
-                settings
-                    .video_codecs
-                    .iter()
-                    .chain(settings.audio_codecs.iter())
-                    .map(|codec| codec.name.clone())
-                    .collect::<HashSet<String>>()
+                (
+                    settings
+                        .video_codecs
+                        .iter()
+                        .chain(settings.audio_codecs.iter())
+                        .map(|codec| codec.name.clone())
+                        .collect::<HashSet<String>>(),
+                    settings.do_retransmission,
+                )
            };
            let caps = media
                .formats()
@ -828,7 +850,7 @@ impl BaseWebRTCSrc {
                        &[&direction, &caps],
                    );

-                    transceiver.set_property("do_nack", true);
+                    transceiver.set_property("do-nack", do_retransmission);
                    transceiver.set_property("fec-type", gst_webrtc::WebRTCFECType::UlpRed);
                }
            } else {
@ -1244,11 +1266,6 @@ impl ObjectImpl for WhipServerSrc {

        let _ = ws.set_signaller(WhipServerSignaller::default().upcast());

-        let obj = &*self.obj();
-
-        obj.set_suppressed_flags(gst::ElementFlags::SINK | gst::ElementFlags::SOURCE);
-        obj.set_element_flags(gst::ElementFlags::SOURCE);
-
        let settings = ws.settings.lock().unwrap();
        element
            .bind_property("stun-server", &settings.signaller, "stun-server")
@ -1286,3 +1303,44 @@ impl ObjectSubclass for WhipServerSrc {
    type Type = super::WhipServerSrc;
    type ParentType = super::BaseWebRTCSrc;
 }
+
+#[derive(Default)]
+pub struct LiveKitWebRTCSrc;
+
+impl ObjectImpl for LiveKitWebRTCSrc {
+    fn constructed(&self) {
+        self.parent_constructed();
+        let element = self.obj();
+        let ws = element.upcast_ref::<super::BaseWebRTCSrc>().imp();
+
+        let _ = ws.set_signaller(LiveKitSignaller::new_consumer().upcast());
+    }
+}
+
+impl GstObjectImpl for LiveKitWebRTCSrc {}
+
+impl BinImpl for LiveKitWebRTCSrc {}
+
+impl ElementImpl for LiveKitWebRTCSrc {
+    fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
+        static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
+            gst::subclass::ElementMetadata::new(
+                "LiveKitWebRTCSrc",
+                "Source/Network/WebRTC",
+                "WebRTC source with LiveKit signaller",
+                "Jordan Yelloz <jordan.yelloz@collabora.com>",
+            )
+        });
+
+        Some(&*ELEMENT_METADATA)
+    }
+}
+
+impl BaseWebRTCSrcImpl for LiveKitWebRTCSrc {}
+
+#[glib::object_subclass]
+impl ObjectSubclass for LiveKitWebRTCSrc {
+    const NAME: &'static str = "GstLiveKitWebRTCSrc";
+    type Type = super::LiveKitWebRTCSrc;
+    type ParentType = super::BaseWebRTCSrc;
+}
--- a/net/webrtc/src/webrtcsrc/mod.rs
+++ b/net/webrtc/src/webrtcsrc/mod.rs
@ -53,6 +53,10 @@ glib::wrapper! {
    pub struct WhipServerSrc(ObjectSubclass<imp::WhipServerSrc>) @extends BaseWebRTCSrc, gst::Bin, gst::Element, gst::Object, @implements gst::URIHandler, gst::ChildProxy;
 }

+glib::wrapper! {
+    pub struct LiveKitWebRTCSrc(ObjectSubclass<imp::LiveKitWebRTCSrc>) @extends BaseWebRTCSrc, gst::Bin, gst::Element, gst::Object, gst::ChildProxy;
+}
+
 glib::wrapper! {
    pub struct WebRTCSrcPad(ObjectSubclass<pad::WebRTCSrcPad>) @extends gst::GhostPad, gst::ProxyPad, gst::Pad, gst::Object;
 }
@ -76,5 +80,60 @@ pub fn register(plugin: Option<&gst::Plugin>) -> Result<(), glib::BoolError> {
        WhipServerSrc::static_type(),
    )?;

+    /**
+     * element-livekitwebrtcsrc:
+     *
+     * The `livekitwebrtcsrc` plays streams from a LiveKit room.
+     *
+     * The element can either subscribe to the streams published by a single
+     * peer in the room using the same `signaller::producer-peer-id` child
+     * property that other webrtcsrc elements use or auto-subscribe to all peers
+     * in a room by not specifying anything for that property. When in
+     * auto-subscribe mode, you can use
+     * `signaller::excluded-producer-peer-ids=<a,b,c>` to ignore peers `a`, `b`,
+     * and `c` while subscribing to all other members of the room.
+     *
+     * ## Sample Pipeline
+     *
+     * First, start the livekit server with the `--dev` flag to enable the test credentials.
+     *
+     * Next, publish a stream:
+     *
+     * ```shell
+     * gst-launch-1.0 \
+     * videotestsrc is-live=1 \
+     * ! video/x-raw,width=640,height=360,framerate=15/1 \
+     * ! timeoverlay ! videoconvert ! queue \
+     * ! livekitwebrtcsink name=sink \
+     *     signaller::ws-url=ws://127.0.0.1:7880 \
+     *     signaller::api-key=devkey \
+     *     signaller::secret-key=secret \
+     *     signaller::room-name=testroom \
+     *     signaller::identity=gst-producer \
+     *     signaller::participant-name=gst-producer \
+     *     video-caps='video/x-vp8'
+     * ```
+     *
+     * Finally, watch the stream:
+     *
+     * ```shell
+     * gst-launch-1.0 \
+     * livekitwebrtcsrc \
+     *   signaller::ws-url=ws://127.0.0.1:7880 \
+     *   signaller::api-key=devkey \
+     *   signaller::secret-key=secret \
+     *   signaller::room-name=testroom \
+     *   signaller::identity=gst-consumer \
+     *   signaller::participant-name=gst-consumer \
+     * ! queue ! videoconvert ! autovideosink
+     * ```
+     */
+    gst::Element::register(
+        plugin,
+        "livekitwebrtcsrc",
+        gst::Rank::NONE,
+        LiveKitWebRTCSrc::static_type(),
+    )?;
+
    Ok(())
 }
--- a/text/regex/src/gstregex/imp.rs
+++ b/text/regex/src/gstregex/imp.rs
@ -10,7 +10,7 @@ use gst::glib;
 use gst::prelude::*;
 use gst::subclass::prelude::*;

-use regex::Regex;
+use regex::{Regex, RegexBuilder};
 use std::default::Default;
 use std::sync::Mutex;

@ -193,7 +193,33 @@ impl ObjectImpl for RegEx {
                        }
                    };

-                    let regex = match Regex::new(&pattern) {
+                    let mut builder = RegexBuilder::new(&pattern);
+                    builder
+                        .unicode(s.get::<bool>("unicode").unwrap_or(true))
+                        .case_insensitive(s.get::<bool>("case-insensitive").unwrap_or(false))
+                        .multi_line(s.get::<bool>("multi-line").unwrap_or(false))
+                        .dot_matches_new_line(
+                            s.get::<bool>("dot-matches-new-line").unwrap_or(false),
+                        )
+                        .crlf(s.get::<bool>("crlf").unwrap_or(false))
+                        .line_terminator(s.get::<u8>("line-terminator").unwrap_or(b'\n'))
+                        .swap_greed(s.get::<bool>("swap-greed").unwrap_or(false))
+                        .ignore_whitespace(s.get::<bool>("ignore-whitespace").unwrap_or(false))
+                        .octal(s.get::<bool>("octal").unwrap_or(false));
+
+                    if let Ok(limit) = s.get::<u64>("size-limit") {
+                        builder.size_limit(limit as usize);
+                    }
+
+                    if let Ok(limit) = s.get::<u64>("dfa-size-limit") {
+                        builder.dfa_size_limit(limit as usize);
+                    }
+
+                    if let Ok(limit) = s.get::<u32>("nest-limit") {
+                        builder.nest_limit(limit);
+                    }
+
+                    let regex = match builder.build() {
                        Ok(regex) => regex,
                        Err(err) => {
                            gst::error!(CAT, imp: self, "Failed to compile regex: {:?}", err);
--- a/text/wrap/src/gsttextwrap/imp.rs
+++ b/text/wrap/src/gsttextwrap/imp.rs
@ -174,7 +174,7 @@ impl TextWrap {
                    CAT,
                    imp: self,
                    "Outputting contents {}, ts: {}, duration: {}",
-                    drained.to_string(),
+                    drained,
                    state.start_ts.display(),
                    duration.display(),
                );
@ -199,7 +199,7 @@ impl TextWrap {

            state.end_ts = buffer.pts();

-            let words = data.split_whitespace();
+            let words = data.split_ascii_whitespace();
            let mut current_text = state.current_text.to_string();

            for word in words {
--- a/utils/fallbackswitch/Cargo.toml
+++ b/utils/fallbackswitch/Cargo.toml
@ -13,7 +13,7 @@ gst.workspace = true
 gst-base.workspace = true
 gst-audio.workspace = true
 gst-video.workspace = true
-gst-plugin-gtk4 = { path = "../../video/gtk4", optional = true }
+gst-plugin-gtk4 = { path = "../../video/gtk4", optional = true, version = "0.12" }
 gtk = { workspace = true, optional = true }
 gio = { workspace = true, optional = true }
 parking_lot = "0.12"
--- a/utils/fallbackswitch/src/fallbacksrc/imp.rs
+++ b/utils/fallbackswitch/src/fallbacksrc/imp.rs
@ -534,7 +534,7 @@ impl ObjectImpl for FallbackSrc {

    // Called whenever a value of a property is read. It can be called
    // at any time from any thread.
-    #[allow(clippy::blocks_in_if_conditions)]
+    #[allow(clippy::block_in_conditions)]
    fn property(&self, _id: usize, pspec: &glib::ParamSpec) -> glib::Value {
        match pspec.name() {
            "enable-audio" => {
@ -3237,7 +3237,7 @@ impl FallbackSrc {
        });
    }

-    #[allow(clippy::blocks_in_if_conditions)]
+    #[allow(clippy::block_in_conditions)]
    fn schedule_source_restart_timeout(
        &self,
        state: &mut State,
@ -3400,7 +3400,7 @@ impl FallbackSrc {
        source.restart_timeout = Some(timeout);
    }

-    #[allow(clippy::blocks_in_if_conditions)]
+    #[allow(clippy::block_in_conditions)]
    fn have_fallback_activated(&self, state: &State) -> bool {
        let mut have_audio = false;
        let mut have_video = false;
--- a/utils/fallbackswitch/src/fallbackswitch/imp.rs
+++ b/utils/fallbackswitch/src/fallbackswitch/imp.rs
@ -771,7 +771,7 @@ impl FallbackSwitch {
            is_active
        );

-        #[allow(clippy::blocks_in_if_conditions)]
+        #[allow(clippy::block_in_conditions)]
        let output_clockid = if is_active {
            pad_state.schedule_clock(
                self,
--- a/utils/livesync/Cargo.toml
+++ b/utils/livesync/Cargo.toml
@ -12,7 +12,7 @@ rust-version.workspace = true
 gio = { workspace = true, optional = true }
 gst.workspace = true
 gst-audio.workspace = true
-gst-plugin-gtk4 = { path = "../../video/gtk4", optional = true }
+gst-plugin-gtk4 = { path = "../../video/gtk4", optional = true, version = "0.12" }
 gtk = { workspace = true, optional = true }
 num-rational = { version = "0.4", default-features = false, features = [] }
 once_cell.workspace = true
--- a/utils/livesync/tests/livesync.rs
+++ b/utils/livesync/tests/livesync.rs
@ -75,16 +75,22 @@ fn test_video_singlesegment() {
 }

 #[test]
+// FIXME: racy: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/328
+#[ignore]
 fn test_audio_singlesegment() {
    test_audio(true);
 }

 #[test]
+// FIXME: racy: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/357
+#[ignore]
 fn test_video_nonsinglesegment() {
    test_video(false);
 }

 #[test]
+// FIXME: racy: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/328
+#[ignore]
 fn test_audio_nonsinglesegment() {
    test_audio(false);
 }
--- a/utils/togglerecord/Cargo.toml
+++ b/utils/togglerecord/Cargo.toml
@ -12,7 +12,7 @@ rust-version.workspace = true
 gst.workspace = true
 gst-audio.workspace = true
 gst-video.workspace = true
-gst-plugin-gtk4 = { path = "../../video/gtk4", optional = true }
+gst-plugin-gtk4 = { path = "../../video/gtk4", optional = true, version = "0.12" }
 gtk = { workspace = true, optional = true }
 gio = { workspace = true, optional = true }
 parking_lot = "0.12"
--- a/utils/togglerecord/src/togglerecord/imp.rs
+++ b/utils/togglerecord/src/togglerecord/imp.rs
@ -693,7 +693,7 @@ impl ToggleRecord {
        }
    }

-    #[allow(clippy::blocks_in_if_conditions)]
+    #[allow(clippy::block_in_conditions)]
    fn handle_secondary_stream<T: HandleData>(
        &self,
        pad: &gst::Pad,
--- a/utils/togglerecord/tests/tests.rs
+++ b/utils/togglerecord/tests/tests.rs
@ -979,6 +979,8 @@ fn test_two_stream_close_open_nonlivein_nonliveout() {
 }

 #[test]
+// FIXME: racy: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/513
+#[ignore]
 fn test_two_stream_close_open_nonlivein_liveout() {
    init();

--- a/version-helper/Cargo.toml
+++ b/version-helper/Cargo.toml
@ -1,6 +1,6 @@
 [package]
 name = "gst-plugin-version-helper"
-version = "0.8.0"
+version = "0.8.2"
 authors = ["Sajeer Ahamed <ahamedsajeer.15.15@cse.mrt.ac.lk>",
           "Sebastian Dröge <sebastian@centricular.com>"]
 categories = ["development-tools"]
@ -14,4 +14,4 @@ rust-version = "1.69"

 [dependencies]
 chrono = { version = "0.4.23", default-features = false, features = ["std", "clock"] }
-toml_edit = { version = "0.22", default-features = false, features = ["parse"] }
+toml_edit = { version = "0.22.8", default-features = false, features = ["parse"] }
--- a/version-helper/src/lib.rs
+++ b/version-helper/src/lib.rs
@ -95,7 +95,7 @@ fn cargo_metadata_release_date(crate_dir: &path::Path) -> Option<chrono::DateTim
    let mut contents = String::new();
    file.read_to_string(&mut contents).ok()?;

-    let doc = contents.parse::<toml_edit::Document>().ok()?;
+    let doc = contents.parse::<toml_edit::DocumentMut>().ok()?;
    let release_date = doc
        .get("package")
        .and_then(|package| package.as_table_like())
--- a/video/closedcaption/src/scc_parse/imp.rs
+++ b/video/closedcaption/src/scc_parse/imp.rs
@ -378,6 +378,10 @@ impl SccParse {

            // Still need to scan lines to find the first buffer
            if state.seeking {
+                // Remember this timecode in order to fallback to this one
+                // if invalid timecode is detected during scanning
+                state.last_timecode = Some(timecode);
+
                drop(state);
                return Ok(self.state.lock().unwrap());
            }
--- a/video/gtk4/src/sink/imp.rs
+++ b/video/gtk4/src/sink/imp.rs
@ -61,6 +61,7 @@ pub(crate) static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
 #[derive(Default)]
 pub struct PaintableSink {
    paintable: Mutex<Option<ThreadGuard<Paintable>>>,
+    window: Mutex<Option<ThreadGuard<gtk::Window>>>,
    info: Mutex<Option<gst_video::VideoInfo>>,
    sender: Mutex<Option<async_channel::Sender<SinkEvent>>>,
    pending_frame: Mutex<Option<Frame>>,
@ -231,6 +232,18 @@ impl ElementImpl for PaintableSink {
    ) -> Result<gst::StateChangeSuccess, gst::StateChangeError> {
        match transition {
            gst::StateChange::NullToReady => {
+                let create_window = glib::program_name().as_deref() == Some("gst-launch-1.0")
+                    || std::env::var("GST_GTK4_WINDOW").as_deref() == Ok("1");
+
+                if create_window {
+                    let res = utils::invoke_on_main_thread(gtk::init);
+
+                    if let Err(err) = res {
+                        gst::error!(CAT, imp: self, "Failed to create initialize GTK: {err}");
+                        return Err(gst::StateChangeError);
+                    }
+                }
+
                let mut paintable = self.paintable.lock().unwrap();

                if paintable.is_none() {
@ -273,6 +286,10 @@ impl ElementImpl for PaintableSink {
                        );
                    }
                }
+
+                if create_window {
+                    self.create_window();
+                }
            }
            _ => (),
        }
@ -294,6 +311,17 @@ impl ElementImpl for PaintableSink {
                    }
                });
            }
+            gst::StateChange::ReadyToNull => {
+                let mut window_guard = self.window.lock().unwrap();
+                if let Some(window) = window_guard.take() {
+                    drop(window_guard);
+
+                    glib::MainContext::default().invoke(move || {
+                        let window = window.get_ref();
+                        window.close();
+                    });
+                }
+            }
            _ => (),
        }

@ -520,6 +548,46 @@ impl PaintableSink {
            .replace(tmp_caps);
    }

+    fn create_window(&self) {
+        let self_ = self.to_owned();
+        glib::MainContext::default().invoke(move || {
+            let mut window_guard = self_.window.lock().unwrap();
+            if window_guard.is_some() {
+                return;
+            }
+
+            let paintable = match &*self_.paintable.lock().unwrap() {
+                Some(paintable) => paintable.get_ref().clone(),
+                None => return,
+            };
+
+            let window = gtk::Window::new();
+            let picture = gtk::Picture::new();
+            picture.set_paintable(Some(&paintable));
+            window.set_child(Some(&picture));
+            window.set_default_size(640, 480);
+
+            window.connect_close_request({
+                let self_ = self_.clone();
+                move |_window| {
+                    if self_.window.lock().unwrap().is_some() {
+                        gst::element_imp_error!(
+                            self_,
+                            gst::ResourceError::NotFound,
+                            ("Output window was closed")
+                        );
+                    }
+
+                    glib::Propagation::Proceed
+                }
+            });
+
+            window.show();
+
+            *window_guard = Some(ThreadGuard::new(window));
+        });
+    }
+
    fn create_paintable(&self, paintable_storage: &mut MutexGuard<Option<ThreadGuard<Paintable>>>) {
        #[cfg(any(target_os = "macos", target_os = "windows", feature = "gst-gl"))]
        {
--- a/video/gtk4/src/sink/paintable/imp.rs
+++ b/video/gtk4/src/sink/paintable/imp.rs
@ -220,7 +220,9 @@ impl PaintableImpl for Paintable {

                let texture_width = *paintable_width * scale_x as f32;
                let texture_height = *paintable_height * scale_y as f32;
-                let bounds = graphene::Rect::new(*x, *y, texture_width, texture_height);
+                let x = *x * scale_x as f32;
+                let y = *y * scale_y as f32;
+                let bounds = graphene::Rect::new(x, y, texture_width, texture_height);

                // Only premultiply GL textures that expect to be in premultiplied RGBA format.
                //
Author	SHA1	Message	Date
Sebastian Dröge	5f6ed181c2	Update version to 0.12.3	2024-03-21 18:53:12 +02:00
Sebastian Dröge	10d7324d90	Update CHANGELOG.md for 0.12.3	2024-03-21 18:47:31 +02:00
Sebastian Dröge	03166bcc35	Update Cargo.lock Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-21 13:30:20 +02:00
François Laignel	fd154d272c	webrtcsrc: add do-retransmission property Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-21 13:30:20 +02:00
Sebastian Dröge	77e91e96ff	livesync: Ignore another racy test Same problem as https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/328 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-21 13:30:20 +02:00
François Laignel	ede354d7a5	webrtcsink: prevent video-info error log for audio streams The following error is logged when `webrtcsink` is feeded with an audio stream: > ERROR video-info video-info.c:540:gst_video_info_from_caps: > wrong name 'audio/x-raw', expected video/ or image/ This commit bypasses `VideoInfo::from_caps` for audio streams. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-21 13:30:20 +02:00
François Laignel	158fe80779	rtp: gccbwe: don't break downstream assumptions pushing buffer lists Some elements in the RTP stack assume all buffers in a `gst::BufferList` correspond to the same timestamp. See in [`rtpsession`] for instance. This also had the effect that `rtpsession` did not create correct RTCP as it only saw some of the SSRCs in the stream. `rtpgccbwe` formed a packet group by gathering buffers in a `gst::BufferList`, regardless of whether they corresponded to the same timestamp, which broke synchronization under certain circonstances. This commit makes `rtpgccbwe` push the buffers as they were received: one by one. [`rtpsession`]: `bc858976db/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c (L2462)` Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-21 13:30:20 +02:00
Guillaume Desmottes	a502dba6d5	webrtc: janus: handle 'hangup' messages from Janus Fix error about this message not being handled: { "janus": "hangup", "session_id": 4758817463851315, "sender": 4126342934227009, "reason": "Close PC" } Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-21 13:30:20 +02:00
Guillaume Desmottes	be055f6dfa	webrtc: janus: handle 'destroyed' messages from Janus Fix this error when the room is destroyed: ERROR webrtc-janusvr-signaller imp.rs:413:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x55b166a3fe40> Unknown message from server: { "janus": "event", "session_id": 6667171862739941, "sender": 1964690595468240, "plugindata": { "plugin": "janus.plugin.videoroom", "data": { "videoroom": "destroyed", "room": 8320333573294267 } } } Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-21 13:30:20 +02:00
Sebastian Dröge	c982db73a7	rtp: Switch from chrono to time Which allows to simplify quite a bit of code and avoids us having to handle some API deprecations. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-21 13:30:20 +02:00
Sebastian Dröge	4def418b45	fmp4mux: Move away from deprecated chrono function Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Sebastian Dröge	dab1ff7dd9	version-helper: Use non-deprecated type alias from toml_edit Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Sebastian Dröge	c0970c6cf4	deny: Add override for heck 0.4 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Guillaume Desmottes	21f59c65da	webrtc: allow resolution and framerate input changes Some changes do not require a WebRTC renegotiation so we can allow those. Fix #515 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Mathieu Duponchelle	2374147b0b	gstregex: add support for switches exposed by RegexBuilder The builder allows for instance for switching off case-sensitiveness for the entire pattern, instead of having to do so inline with `(?i)`. All the options exposed by the builder at <https://docs.rs/regex/latest/regex/struct.RegexBuilder.html> can now be passed as fields of invidual commands, snake-cased. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Guillaume Desmottes	3f0c6e654b	gtk4: scale texture position Fix regression in 0.12 introduced by `3423d05f77` Code from Ivan Molodetskikh suggested on Matrix. Fix #519 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Nirbheek Chauhan	ce2aa0aecf	meson: Disable docs completely when the option is disabled Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Guillaume Desmottes	6b7e41e551	threadshare: disable racy tests https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/250 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Guillaume Desmottes	40402f8e58	livesync: disable racy tests https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/328 https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/357 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Guillaume Desmottes	14ed7b0db1	togglerecord: disable racy test_two_stream_close_open_nonlivein_liveout test See https://gitlab.freedesktop.org/gdesmott/gst-plugins-rs/-/jobs/56183085 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Guillaume Desmottes	2d350252c0	spotify: document how to use with non Facebook accounts See discussion on #203. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Seungha Yang	2cb76f2ff2	sccparse: Ignore invalid timecode during seek as well sccparse holds last timecode in order to ignore invalid timecode and fallback to the previous timecode. That should happen when sccparse is handling seek event too. Otherwise single invalid timecode before the target seek position will cause flow error. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Jordan Yelloz	7a90e96332	livekit_signaller: Added missing getter for excluded-producer-peer-ids Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Jordan Yelloz	283d1568b4	webrtcsrc: Removed incorrect URIHandler from LiveKit source Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Sebastian Dröge	12533a4c0d	Remove empty line from the CHANGELOG.md that confuses the GitLab renderer Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Jordan Yelloz	51287705ce	livekit_signaller: Improved shutdown behavior Without sending a Leave request to the server before disconnecting, the disconnected client will appear present and stuck in the room for a little while until the server removes it due to inactivity. After this change, the disconnecting client will immediately leave the room. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Jordan Yelloz	e9edee131b	webrtcsrc: Removed flag setup from WhipServerSrc It's already done in the base class Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Jordan Yelloz	feb01510f9	webrtcsrc: Updated readme for LiveKit source Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Jordan Yelloz	32e13f0a10	webrtcsrc: Added LiveKit source element Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Jordan Yelloz	c8dcd50846	webrtcsink: Updated livekitwebrtcsink for new signaller constructor Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Jordan Yelloz	59ee2721bf	livekit_signaller: Added dual-role support Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:24:55 +01:00
Guillaume Desmottes	133b527391	webrtc: janus: rename RoomId to JanusId Those weird ids are used in multiple places, not only for the room id, so best to have a more generic name. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:54:45 +02:00
Guillaume Desmottes	7f460c2db8	webrtc: janus: room id not optional in 'joined' message Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:54:17 +02:00
Guillaume Desmottes	8361471fcc	webrtc: janus: remove 'audio' and 'video' from publish messages Those are deprecated and no longer used. See https://janus.conf.meetecho.com/docs/videoroom and https://github.com/meetecho/janus-gateway/blob/master/src/plugins/janus_videoroom.c#L9894 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:54:02 +02:00
Guillaume Desmottes	b9ea05a14a	webrtc: janus: numerical room ids are u64 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:53:51 +02:00
Jordan Yelloz	048d51d9d9	webrtcsrc: Made producer-peer-id optional It may be necessary for some signalling clients but the source element doesn't need to depend on it. Also, the value will fall back to the pad's MSID for the first argument to the request-encoded-filter gobject signal when it isn't available from the signalling client. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>	2024-03-20 16:53:22 +02:00
Sebastian Dröge	55b74c9a9a	Update Cargo.lock	2024-02-26 14:56:22 +02:00
Sebastian Dröge	c6841e1e74	Update versions to 0.12.2	2024-02-26 14:55:51 +02:00
Sebastian Dröge	3a50489dac	Update CHANGELOG.md for 0.12.2	2024-02-26 14:55:29 +02:00
Sebastian Dröge	078c76c260	deny: Add zerocopy 0.6 duplicate override for librespot Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:26:36 +02:00
Sebastian Dröge	3ac010bfb6	Update Cargo.lock Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:25:33 +02:00
Xavier Claessens	7edf94f98b	janusvr: Add string-ids property It forces usage of strings even if it can be parsed into an integer. This allows joining room `"133"` in a server configured with string room ids. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:24:29 +02:00
Xavier Claessens	ea59544c71	janusvr: Room IDs can be strings Sponsored-by: Netflix Inc. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:24:25 +02:00
Sebastian Dröge	0b96457395	fmp4mux: Update to dash-mpd 0.15 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:24:17 +02:00
Xavier Claessens	5888f5aa5f	meson: Fix error when default_library=both Skip duplicated plugin_name when we have both the static and shared plugin in the plugins list. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:24:08 +02:00
Maksym Khomenko	4e86b0f3c8	webrtcsink: extensions: separate API and signal checks Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:24:02 +02:00
Maksym Khomenko	98411e97f1	webrtcsink: apply rustfmt Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:23:56 +02:00
Mathieu Duponchelle	8211c253a8	textwrap: don't split on all whitespaces .. but only on ASCII whitespaces, as we want to honor non-breaking whitespaces (\u{a0}) Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:23:50 +02:00
Xavier Claessens	3a0f30be96	janusvr: Add secret-key property Every API calls have an optional "apisecret" argument. Sponsored-by: Netflix Inc. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:23:45 +02:00
Sebastian Dröge	de7e4806e5	deny: Add winnow 0.5 override Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:23:38 +02:00
Sebastian Dröge	27d806ae85	Remove Cargo.lock from .gitignore Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:21:00 +02:00
Sebastian Dröge	3679db5740	rtpgccbwe: Don't reset PTS/DTS to `None` The element is usually placed before `rtpsession`, and `rtpsession` needs the PTS/DTS for correctly determining the running time. The running time is then used to produce correct RTCP SR, and to potentially update an NTP-64 RTP header extension if existing on the packets. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/496 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>	2024-02-26 14:20:53 +02:00
Sebastian Dröge	2f2aac55a3	Update version to 0.12.1	2024-02-13 13:02:27 +02:00
Sebastian Dröge	31dfcd0a78	Update CHANGELOG.md for 0.12.1	2024-02-13 13:01:46 +02:00
Sebastian Dröge	b3e233f0c5	Update Cargo.lock Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>	2024-02-13 12:37:23 +02:00
Sebastian Dröge	58a065caf3	textwrap: Remove unnecessary `to_string()` in debug output of a string Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>	2024-02-13 12:35:40 +02:00
Jordan Yelloz	606352d7cf	webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>	2024-02-12 18:11:42 +02:00
Sebastian Dröge	aa2d056ea1	Update to async-tungstenite 0.25 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>	2024-02-12 18:11:31 +02:00
Sebastian Dröge	3f9d5cf2f0	gtk4: Create a window if running from gst-launch-1.0 or GST_GTK4_WINDOW=1 is set Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/482 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>	2024-02-12 18:11:25 +02:00
Sebastian Dröge	149eff08b7	utils: Update for renamed clippy lint in 1.76 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>	2024-02-12 18:11:19 +02:00
Sebastian Dröge	c4e3fff2a2	Update Cargo.lock Downgrade clap_derive to 4.4.7 to not require Rust 1.74 or newer.	2024-02-08 20:52:14 +02:00
Sebastian Dröge	16e001e3f2	Update dependency versions for gtk-rs-core / gtk4-rs / gstreamer-rs and local crates	2024-02-08 19:40:08 +02:00
Sebastian Dröge	af694e8bc1	ci: Use 0.22 branch of gstreamer-rs images templates	2024-02-08 19:35:05 +02:00
Sebastian Dröge	66f2969eb9	Update Cargo.lock	2024-02-08 19:33:32 +02:00
Sebastian Dröge	50efdf6a64	Update version to 0.12.0	2024-02-08 19:33:09 +02:00