mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-06-12 19:29:24 +00:00
ab1ec12698
This is a first step where we try to replicate encoding conditions from the input stream into the discovery pipeline. A second patch will implement using input buffers in the discovery pipelines. This moves discovery to using input buffers directly. Instead of trying to replicate buffers that `webrtcsink` is getting as input with testsrc, directly run discovery based on the real buffers. This way we are sure we work with the exact right stream type and we don't need encoders to support encoding streams inputs. We use the same logic for both encoded and raw input to avoid having several code paths and makes it all more correct in any case. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
88 lines
3.2 KiB
TOML
88 lines
3.2 KiB
TOML
[package]
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name = "gst-plugin-webrtc"
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version = "0.11.0-alpha.1"
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edition = "2021"
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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 = "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
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rust-version = "1.66"
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[dependencies]
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gst = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer", features = ["v1_20", "serde"] }
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gst-app = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-app", features = ["v1_20"] }
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gst-video = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-video", features = ["v1_20", "serde"] }
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gst-webrtc = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-webrtc", features = ["v1_20"] }
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gst-sdp = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-sdp", features = ["v1_20"] }
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gst-rtp = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-rtp", features = ["v1_20"] }
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gst-utils = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-utils" }
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gst-base = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-base" }
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uuid = { version = "1", features = ["v4"] }
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once_cell = "1.0"
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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 = "0.22", features = ["tokio-runtime", "tokio-native-tls"] }
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serde = "1"
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serde_json = "1"
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fastrand = "1.0"
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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 = "0.55.0"
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aws-types = "0.55.0"
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aws-credential-types = "0.55.0"
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aws-sig-auth = "0.55.0"
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aws-smithy-http = { version = "0.55.0", features = [ "rt-tokio" ] }
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aws-smithy-types = "0.55.0"
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aws-sdk-kinesisvideo = "0.28.0"
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aws-sdk-kinesisvideosignaling = "0.28.0"
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http = "0.2.7"
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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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[dev-dependencies]
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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 = "0.1"
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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 = { path = "../../version-helper" }
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[features]
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static = []
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capi = []
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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"]
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doc = []
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[package.metadata.capi]
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min_version = "0.8.0"
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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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[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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