mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-11-25 13:01:07 +00:00
87c6719e1d
When congestion control is used for a session with multiple encoders, the default implementation simply divides the overall bitrate equally between encoders. This is not always desirable, and this patch exposes a new signal that users can register to, with two arguments: * The overall bitrate to allocate * A structure with an encoder.stream_name -> bitrate mapping Handlers should return a similar structure with a custom mapping. An example is also provided. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1792>
132 lines
4.6 KiB
TOML
132 lines
4.6 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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chrono = "0.4"
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thiserror = "1"
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futures = "0.3"
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itertools = "0.12"
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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 = "0.28", features = ["tokio-runtime", "tokio-native-tls", "url"] }
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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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gst_plugin_webrtc_signalling = { path="signalling", package = "gst-plugin-webrtc-signalling" }
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human_bytes = "0.4"
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once_cell.workspace = true
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rand = "0.8"
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url = "2"
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aws-config = { version = "1.0", optional = true }
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aws-types = { version = "1.0", optional = true }
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aws-credential-types = { version = "1.0", optional = true }
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aws-sigv4 = { version = "1.0", optional = true }
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aws-smithy-http = { version = "0.60", features = [ "rt-tokio" ], optional = true }
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aws-smithy-types = { version = "1.0", optional = true }
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aws-sdk-kinesisvideo = { version = "1.0", optional = true }
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aws-sdk-kinesisvideosignaling = { version = "1.0", optional = true }
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http = { version = "1.0", optional = true }
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data-encoding = {version = "2.3.3", optional = true }
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url-escape = { version = "0.1.1", optional = true }
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reqwest = { version = "0.11", features = ["default-tls"], optional = true }
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parse_link_header = {version = "0.3", features = ["url"]}
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async-recursion = { version = "1.0.0", optional = true }
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livekit-protocol = { version = "0.3, < 0.3.4", optional = true }
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livekit-api = { version = "0.3", default-features = false, features = ["signal-client", "access-token", "native-tls"], optional = true }
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warp = {version = "0.3", optional = true, features = ["tls"] }
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ctrlc = {version = "3.4.0", optional = true }
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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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[dev-dependencies]
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gst-plugin-rtp = { path = "../rtp" }
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tokio = { version = "1", features = ["signal"] }
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clap = { version = "4", features = ["derive"] }
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regex = "1"
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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", "aws", "janus", "livekit", "whip", "web_server"]
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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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aws = ["dep:aws-config", "dep:aws-types", "dep:aws-credential-types", "dep:aws-sigv4",
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"dep:aws-smithy-http", "dep:aws-smithy-types", "dep:aws-sdk-kinesisvideo",
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"dep:aws-sdk-kinesisvideosignaling", "dep:data-encoding", "dep:http", "dep:url-escape"]
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janus = ["dep:http"]
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livekit = ["dep:livekit-protocol", "dep:livekit-api"]
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whip = ["dep:async-recursion", "dep:reqwest", "dep:warp", "dep:ctrlc"]
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web_server = ["dep:warp"]
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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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[[example]]
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name = "whipserver"
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required-features = [ "whip" ]
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[[example]]
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name = "webrtcsink-define-encoder-bitrates"
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