gst-plugin-webrtc 0.13.4

GStreamer plugin for high level WebRTC elements and a simple signaling server
[build-dependencies.gst-plugin-version-helper]
version = "0.8"

[dependencies.anyhow]
version = "1"

[dependencies.async-recursion]
optional = true
version = "1.0.0"

[dependencies.async-tungstenite]
features = ["tokio-runtime", "tokio-native-tls", "url"]
version = "0.28"

[dependencies.aws-config]
optional = true
version = "1.0"

[dependencies.aws-credential-types]
optional = true
version = "1.0"

[dependencies.aws-sdk-kinesisvideo]
optional = true
version = "1.0"

[dependencies.aws-sdk-kinesisvideosignaling]
optional = true
version = "1.0"

[dependencies.aws-sigv4]
optional = true
version = "1.0"

[dependencies.aws-smithy-http]
features = ["rt-tokio"]
optional = true
version = "0.60"

[dependencies.aws-smithy-types]
optional = true
version = "1.0"

[dependencies.aws-types]
optional = true
version = "1.0"

[dependencies.chrono]
version = "0.4"

[dependencies.ctrlc]
optional = true
version = "3.4.0"

[dependencies.data-encoding]
optional = true
version = "2.3.3"

[dependencies.fastrand]
version = "2.0"

[dependencies.futures]
version = "0.3"

[dependencies.gst]
features = ["v1_20", "serde"]
package = "gstreamer"
version = "0.23"

[dependencies.gst-app]
features = ["v1_20"]
package = "gstreamer-app"
version = "0.23"

[dependencies.gst-audio]
features = ["v1_20", "serde"]
package = "gstreamer-audio"
version = "0.23"

[dependencies.gst-base]
package = "gstreamer-base"
version = "0.23"

[dependencies.gst-net]
features = ["v1_20"]
package = "gstreamer-net"
version = "0.23"

[dependencies.gst-rtp]
features = ["v1_20"]
package = "gstreamer-rtp"
version = "0.23"

[dependencies.gst-sdp]
features = ["v1_20"]
package = "gstreamer-sdp"
version = "0.23"

[dependencies.gst-utils]
package = "gstreamer-utils"
version = "0.23"

[dependencies.gst-video]
features = ["v1_20", "serde"]
package = "gstreamer-video"
version = "0.23"

[dependencies.gst-webrtc]
features = ["v1_20"]
package = "gstreamer-webrtc"
version = "0.23"

[dependencies.gst_plugin_webrtc_protocol]
package = "gst-plugin-webrtc-signalling-protocol"
version = "0.13"

[dependencies.http]
optional = true
version = "1.0"

[dependencies.human_bytes]
version = "0.4"

[dependencies.livekit-api]
default-features = false
features = ["signal-client", "access-token", "native-tls"]
optional = true
version = "0.3"

[dependencies.livekit-protocol]
optional = true
version = "0.3, < 0.3.4"

[dependencies.once_cell]
version = "1"

[dependencies.parse_link_header]
features = ["url"]
version = "0.3"

[dependencies.rand]
version = "0.8"

[dependencies.reqwest]
features = ["default-tls"]
optional = true
version = "0.11"

[dependencies.serde]
features = ["derive"]
version = "1"

[dependencies.serde_json]
version = "1"

[dependencies.thiserror]
version = "1"

[dependencies.tokio]
features = ["fs", "macros", "rt-multi-thread", "time"]
version = "1"

[dependencies.tokio-native-tls]
version = "0.3.0"

[dependencies.tokio-stream]
version = "0.1.11"

[dependencies.url]
version = "2"

[dependencies.url-escape]
optional = true
version = "0.1.1"

[dependencies.uuid]
features = ["v4"]
version = "1"

[dependencies.warp]
optional = true
version = "0.3"

[dev-dependencies.clap]
features = ["derive"]
version = "4"

[dev-dependencies.regex]
version = "1"

[dev-dependencies.tokio]
features = ["signal"]
version = "1"

[dev-dependencies.tracing]
features = ["log"]
version = "0.1"

[dev-dependencies.tracing-log]
version = "0.2"

[dev-dependencies.tracing-subscriber]
features = ["registry", "env-filter"]
version = "0.3"

[[example]]
name = "webrtc-precise-sync-recv"
path = "examples/webrtc-precise-sync-recv.rs"

[[example]]
name = "webrtc-precise-sync-send"
path = "examples/webrtc-precise-sync-send.rs"

[[example]]
name = "webrtcsink-custom-signaller"
path = "examples/webrtcsink-custom-signaller/main.rs"

[[example]]
name = "webrtcsink-high-quality-tune"
path = "examples/webrtcsink-high-quality-tune.rs"

[[example]]
name = "webrtcsink-stats-server"
path = "examples/webrtcsink-stats-server.rs"

[[example]]
name = "whipserver"
path = "examples/whipserver.rs"
required-features = ["whip"]

[features]
aws = ["dep:aws-config", "dep:aws-types", "dep:aws-credential-types", "dep:aws-sigv4", "dep:aws-smithy-http", "dep:aws-smithy-types", "dep:aws-sdk-kinesisvideo", "dep:aws-sdk-kinesisvideosignaling", "dep:data-encoding", "dep:http", "dep:url-escape"]
capi = []
default = ["v1_22", "aws", "janus", "livekit", "whip"]
doc = []
janus = ["dep:http"]
livekit = ["dep:livekit-protocol", "dep:livekit-api"]
static = []
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"]
whip = ["dep:async-recursion", "dep:reqwest", "dep:warp", "dep:ctrlc"]

[lib]
crate-type = ["cdylib", "rlib"]
name = "gstrswebrtc"
path = "src/lib.rs"

[package]
authors = ["Mathieu Duponchelle <mathieu@centricular.com>", "Thibault Saunier <tsaunier@igalia.com>"]
autobenches = false
autobins = false
autoexamples = false
autolib = false
autotests = false
build = "build.rs"
description = "GStreamer plugin for high level WebRTC elements and a simple signaling server"
edition = "2021"
license = "MPL-2.0"
name = "gst-plugin-webrtc"
readme = "README.md"
repository = "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
rust-version = "1.71"
version = "0.13.4"

[package.metadata.capi]
min_version = "0.9.21"

[package.metadata.capi.header]
enabled = false

[package.metadata.capi.library]
import_library = false
install_subdir = "gstreamer-1.0"
versioning = false

[package.metadata.capi.pkg_config]
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"

[package.metadata.gstreamer]
release_date = "2024-12-24"