mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-05-09 20:02:41 +00:00
105 lines
3.1 KiB
TOML
105 lines
3.1 KiB
TOML
[package]
|
|
name = "gst-plugin-webrtc"
|
|
version.workspace = true
|
|
edition.workspace = true
|
|
authors = ["Mathieu Duponchelle <mathieu@centricular.com>", "Thibault Saunier <tsaunier@igalia.com>"]
|
|
license = "MPL-2.0"
|
|
description = "GStreamer plugin for high level WebRTC elements and a simple signaling server"
|
|
repository.workspace = true
|
|
rust-version.workspace = true
|
|
|
|
[dependencies]
|
|
gst = { workspace = true, features = ["v1_20", "serde"] }
|
|
gst-app = { workspace = true, features = ["v1_20"] }
|
|
gst-video = { workspace = true, features = ["v1_20", "serde"] }
|
|
gst-webrtc = { workspace = true, features = ["v1_20"] }
|
|
gst-sdp = { workspace = true, features = ["v1_20"] }
|
|
gst-rtp = { workspace = true, features = ["v1_20"] }
|
|
gst-utils.workspace = true
|
|
gst-base.workspace = true
|
|
uuid = { version = "1", features = ["v4"] }
|
|
|
|
anyhow = "1"
|
|
thiserror = "1"
|
|
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"] }
|
|
serde = { version = "1", features = ["derive"] }
|
|
serde_json = "1"
|
|
fastrand = "2.0"
|
|
gst_plugin_webrtc_protocol = { path="protocol", package = "gst-plugin-webrtc-signalling-protocol" }
|
|
human_bytes = "0.4"
|
|
url = "2"
|
|
|
|
aws-config = "1.0"
|
|
aws-types = "1.0"
|
|
aws-credential-types = "1.0"
|
|
aws-sigv4 = "1.0"
|
|
aws-smithy-http = { version = "0.60", features = [ "rt-tokio" ] }
|
|
aws-smithy-types = "1.0"
|
|
aws-sdk-kinesisvideo = "1.0"
|
|
aws-sdk-kinesisvideosignaling = "1.0"
|
|
http = "1.0"
|
|
chrono = "0.4"
|
|
data-encoding = "2.3.3"
|
|
url-escape = "0.1.1"
|
|
regex = "1"
|
|
|
|
reqwest = { version = "0.11", features = ["default-tls"] }
|
|
parse_link_header = {version = "0.3", features = ["url"]}
|
|
async-recursion = "1.0.0"
|
|
|
|
livekit-protocol = { version = "0.3" }
|
|
livekit-api = { version = "0.3", default-features = false, features = ["signal-client", "access-token", "native-tls"] }
|
|
|
|
warp = "0.3"
|
|
crossbeam-channel = "0.5"
|
|
rand = "0.8"
|
|
once_cell.workspace = true
|
|
|
|
[dev-dependencies]
|
|
tracing = { version = "0.1", features = ["log"] }
|
|
tracing-subscriber = { version = "0.3", features = ["registry", "env-filter"] }
|
|
tracing-log = "0.2"
|
|
clap = { version = "4", features = ["derive"] }
|
|
|
|
[lib]
|
|
name = "gstrswebrtc"
|
|
crate-type = ["cdylib", "rlib"]
|
|
path = "src/lib.rs"
|
|
|
|
[build-dependencies]
|
|
gst-plugin-version-helper.workspace = true
|
|
|
|
[features]
|
|
default = ["v1_22"]
|
|
static = []
|
|
capi = []
|
|
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"]
|
|
doc = []
|
|
|
|
[package.metadata.capi]
|
|
min_version = "0.9.21"
|
|
|
|
[package.metadata.capi.header]
|
|
enabled = false
|
|
|
|
[package.metadata.capi.library]
|
|
install_subdir = "gstreamer-1.0"
|
|
versioning = false
|
|
import_library = 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"
|
|
|
|
[[example]]
|
|
name = "webrtcsink-stats-server"
|
|
|
|
[[example]]
|
|
name = "webrtcsink-high-quality-tune"
|
|
|
|
[[example]]
|
|
name = "webrtcsink-custom-signaller"
|