[package]
name = "gst-plugin-webrtc"
version = "0.9.13"
edition = "2021"
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 = "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
rust-version = "1.63"

[dependencies]
gst = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.19", version = "0.19.1", package = "gstreamer", features = ["v1_20", "serde"] }
gst-app = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.19", version = "0.19", package = "gstreamer-app", features = ["v1_20"] }
gst-video = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.19", version = "0.19", package = "gstreamer-video", features = ["v1_20", "serde"] }
gst-webrtc = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.19", version = "0.19", package = "gstreamer-webrtc", features = ["v1_20"] }
gst-sdp = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.19", version = "0.19", package = "gstreamer-sdp", features = ["v1_20"] }
gst-rtp = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.19", version = "0.19", package = "gstreamer-rtp", features = ["v1_20"] }
gst-utils = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.19", version = "0.19", package = "gstreamer-utils" }
gst-base = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", branch = "0.19", version = "0.19", package = "gstreamer-base" }
once_cell = "1.0"
anyhow = "1"
thiserror = "1"
futures = "0.3"
async-std = { version = "1", features = ["unstable"] }
async-native-tls = { version = "0.4.0" }
async-tungstenite = { version = "0.19", features = ["async-std-runtime", "async-native-tls"] }
serde = "1"
serde_json = "1"
fastrand = "2.0"
gst_plugin_webrtc_protocol = { version = "0.9", path="protocol", package = "gst-plugin-webrtc-signalling-protocol" }
human_bytes = "0.4"
url = "2"

[dev-dependencies]
tracing = { version = "0.1", features = ["log"] }
tracing-subscriber = { version = "0.3", features = ["registry", "env-filter"] }
tracing-log = "0.1"
uuid = { version = "1", features = ["v4"] }
clap = { version = "4", features = ["derive"] }

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

[build-dependencies]
gst-plugin-version-helper = { version = "0.7", path = "../../version-helper" }

[features]
static = []
capi = []
gst1_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 = "0.8.0"

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

[package.metadata.capi.library]
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"

[[example]]
name = "webrtcsink-stats-server"