mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-05-18 16:28:25 +00:00
83d70d3471
This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization) for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to identify the clock. These attributes are handled by `rtpjitterbuffer` on the consumer side. They MUST be part of the SDP offer. When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the clock's origin. Because the payloaders are not instantiated when the offer is sent to the consumer, the RTP offset is set to 0 and the payloader `timstamp-offset`s are set accordingly when they are created. The `webrtc-precise-sync` examples were updated to be able to start with an NTP (default), a PTP or the system clock (on the receiver only). The rtp jitter buffer will synchronize with the clock signalled in the SDP offer provided the sender is started with `--do-clock-signalling` & the receiver with `--expect-clock-signalling`. [RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500>
114 lines
3.3 KiB
TOML
114 lines
3.3 KiB
TOML
[package]
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name = "gst-plugin-webrtc"
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version.workspace = true
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edition.workspace = true
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authors = ["Mathieu Duponchelle <mathieu@centricular.com>", "Thibault Saunier <tsaunier@igalia.com>"]
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license = "MPL-2.0"
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description = "GStreamer plugin for high level WebRTC elements and a simple signaling server"
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repository.workspace = true
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rust-version.workspace = true
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[dependencies]
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gst = { workspace = true, features = ["v1_20", "serde"] }
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gst-app = { workspace = true, features = ["v1_20"] }
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gst-audio = { workspace = true, features = ["v1_20", "serde"] }
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gst-video = { workspace = true, features = ["v1_20", "serde"] }
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gst-net = { workspace = true, features = ["v1_20"] }
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gst-webrtc = { workspace = true, features = ["v1_20"] }
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gst-sdp = { workspace = true, features = ["v1_20"] }
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gst-rtp = { workspace = true, features = ["v1_20"] }
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gst-utils.workspace = true
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gst-base.workspace = true
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uuid = { version = "1", features = ["v4"] }
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anyhow = "1"
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thiserror = "1"
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futures = "0.3"
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tokio = { version = "1", features = ["fs", "macros", "rt-multi-thread", "time"] }
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tokio-native-tls = "0.3.0"
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tokio-stream = "0.1.11"
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async-tungstenite = { version = "0.25", features = ["tokio-runtime", "tokio-native-tls"] }
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serde = { version = "1", features = ["derive"] }
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serde_json = "1"
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fastrand = "2.0"
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gst_plugin_webrtc_protocol = { path="protocol", package = "gst-plugin-webrtc-signalling-protocol" }
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human_bytes = "0.4"
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url = "2"
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aws-config = "1.0"
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aws-types = "1.0"
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aws-credential-types = "1.0"
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aws-sigv4 = "1.0"
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aws-smithy-http = { version = "0.60", features = [ "rt-tokio" ] }
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aws-smithy-types = "1.0"
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aws-sdk-kinesisvideo = "1.0"
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aws-sdk-kinesisvideosignaling = "1.0"
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http = "1.0"
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chrono = "0.4"
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data-encoding = "2.3.3"
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url-escape = "0.1.1"
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regex = "1"
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reqwest = { version = "0.11", features = ["default-tls"] }
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parse_link_header = {version = "0.3", features = ["url"]}
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async-recursion = "1.0.0"
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livekit-protocol = { version = "0.3" }
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livekit-api = { version = "0.3", default-features = false, features = ["signal-client", "access-token", "native-tls"] }
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warp = "0.3"
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crossbeam-channel = "0.5"
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rand = "0.8"
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once_cell.workspace = true
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[dev-dependencies]
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gst-plugin-rtp = { path = "../rtp" }
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tracing = { version = "0.1", features = ["log"] }
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tracing-subscriber = { version = "0.3", features = ["registry", "env-filter"] }
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tracing-log = "0.2"
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clap = { version = "4", features = ["derive"] }
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[lib]
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name = "gstrswebrtc"
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crate-type = ["cdylib", "rlib"]
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path = "src/lib.rs"
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[build-dependencies]
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gst-plugin-version-helper.workspace = true
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[features]
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default = ["v1_22"]
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static = []
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capi = []
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v1_22 = ["gst/v1_22", "gst-app/v1_22", "gst-video/v1_22", "gst-webrtc/v1_22", "gst-sdp/v1_22", "gst-rtp/v1_22"]
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doc = []
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[package.metadata.capi]
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min_version = "0.9.21"
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[package.metadata.capi.header]
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enabled = false
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[package.metadata.capi.library]
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install_subdir = "gstreamer-1.0"
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versioning = false
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import_library = false
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[package.metadata.capi.pkg_config]
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requires_private = "gstreamer-rtp-1.0 >= 1.20, gstreamer-webrtc-1.0 >= 1.20, gstreamer-1.0 >= 1.20, gstreamer-app-1.0 >= 1.20, gstreamer-video-1.0 >= 1.20, gstreamer-sdp-1.0 >= 1.20, gobject-2.0, glib-2.0, gmodule-2.0"
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[[example]]
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name = "webrtcsink-stats-server"
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[[example]]
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name = "webrtcsink-high-quality-tune"
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[[example]]
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name = "webrtcsink-custom-signaller"
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[[example]]
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name = "webrtc-precise-sync-send"
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[[example]]
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name = "webrtc-precise-sync-recv"
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