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gst-plugins-rs/plugins/Cargo.toml

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Initial commit
2021-10-05 21:28:05 +00:00
[package]
name = "webrtcsink"
version = "0.1.0"
edition = "2018"
authors = ["Mathieu Duponchelle <mathieu@centricular.com>"]
license = "MIT"
description = "GStreamer WebRTC sink"
repository = "https://github.com/centricular/webrtcsink/"
build = "build.rs"
[dependencies]
change ser_de to serde because gstreamer-rs rename ser_de feature to serde
2022-09-27 09:42:36 +00:00
gst = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer", features = ["v1_20", "serde"] }
Depend on gstreamer-rs main to add a feature for v1_22
2022-04-14 12:50:33 +00:00
gst-app = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-app", features = ["v1_20"] }
change ser_de to serde because gstreamer-rs rename ser_de feature to serde
2022-09-27 09:42:36 +00:00
gst-video = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-video", features = ["v1_20", "serde"] }
Depend on gstreamer-rs main to add a feature for v1_22
2022-04-14 12:50:33 +00:00
gst-webrtc = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-webrtc", features = ["v1_20"] }
gst-sdp = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-sdp", features = ["v1_20"] }
gst-rtp = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-rtp", features = ["v1_20"] }
Use StreamConsumer from gstreamer itself Merged in https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1022
2022-05-11 14:41:11 +00:00
gst-utils = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-utils" }
Initial commit
2021-10-05 21:28:05 +00:00
once_cell = "1.0"
plugin: Implement bandwidth estimator based on the Google Congestion Control algorithm See https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02 This commit implements the bandwidth estimation as a GStreamer element that is then used in webrtcbin through the new `request-bandwidth-estimator` signal. This keeps our Homegrown congestion controller but removes the possibility to switch CC algorithm at runtime.
2022-05-11 20:58:53 +00:00
chrono = { version = "0.4", default-features = false }
Initial commit
2021-10-05 21:28:05 +00:00
smallvec = "1"
anyhow = "1"
API: refactor error signalling * Expect Box<dyn Error> from custom signaller * Use thiserror to define the error codes returned to custom signallers
2021-12-21 22:37:29 +00:00
thiserror = "1"
Initial commit
2021-10-05 21:28:05 +00:00
futures = "0.3"
webrtcsink: Initial congestion control implementation Naive heuristic lifted from an earlier proof of concept, augmented with logic from https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02#section-5.5 A property is exposed to disable congestion control for testing purposes, it can be extended in the future to allow selecting a different congestion control scheme. + Update the documentation
2021-11-04 17:26:50 +00:00
async-std = { version = "1", features = ["unstable"] }
Add property to specify certificate authority Under certain circumstances one would like to use a self signed certificate without the obstacles many browsers implement to safeguard their users. One approach is to create a certificate authority. One can now add this to the OS certificate store, or simply add it to the browser. In the latter case one also needs to specify the new authority on the webrtcsink. This commit enables this.
2022-03-30 15:31:41 +00:00
async-native-tls = { version = "0.4.0" }
signalling: update protocol, reimplement in rust
2022-03-03 03:30:44 +00:00
async-tungstenite = { version = "0.17", features = ["async-std-runtime", "async-native-tls"] }
Initial commit
2021-10-05 21:28:05 +00:00
serde = "1"
serde_json = "1"
fastrand = "1.0"
signalling: update protocol, reimplement in rust
2022-03-03 03:30:44 +00:00
webrtcsink-protocol = { version = "0.1", path="../protocol" }
Make our "loss based control" algorithm closer to what is defined in [GCC] As specified in Google Congestion Control we should run the packet loss estimation algorithm "every time feedback from the receiver is received". And, also as defined by GCC, we now have 2 different estimated bitrates, one for the delay-based controller value and one for the loss-based one, and we use the minimum value between those 2 as our current estimation. [GCC]: https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02
2022-03-30 15:39:19 +00:00
human_bytes = "0.3.1"
Initial commit
2021-10-05 21:28:05 +00:00
examples: implement webrtcsink-stat-server
2021-11-30 21:49:33 +00:00
[dev-dependencies]
tracing = { version = "0.1", features = ["log"] }
Update to gstreamer-rs 0.18 instead of using it from git And update all other dependencies.
2022-01-25 14:24:22 +00:00
tracing-subscriber = { version = "0.3", features = ["registry", "env-filter"] }
examples: implement webrtcsink-stat-server
2021-11-30 21:49:33 +00:00
tracing-log = "0.1"
uuid = { version = "0.8", features = ["v4"] }
Implement support for FEC / retransmission
2021-12-09 23:06:46 +00:00
clap = { version = "3.0.0-rc.1", features = ["derive"] }
examples: implement webrtcsink-stat-server
2021-11-30 21:49:33 +00:00
Initial commit
2021-10-05 21:28:05 +00:00
[lib]
name = "webrtcsink"
crate-type = ["cdylib", "rlib"]
path = "src/lib.rs"
[build-dependencies]
gst-plugin-version-helper = "0.7"
[features]
static = []
capi = []
Depend on gstreamer-rs main to add a feature for v1_22
2022-04-14 12:50:33 +00:00
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"]
Initial commit
2021-10-05 21:28:05 +00:00
[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]
webrtcsink: Initial congestion control implementation Naive heuristic lifted from an earlier proof of concept, augmented with logic from https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02#section-5.5 A property is exposed to disable congestion control for testing purposes, it can be extended in the future to allow selecting a different congestion control scheme. + Update the documentation
2021-11-04 17:26:50 +00:00
requires_private = "gstreamer-rtp >= 1.20, gstreamer-webrtc >= 1.20, gstreamer-1.0 >= 1.20, gstreamer-app >= 1.20, gstreamer-video >= 1.20, gstreamer-sdp >= 1.20, gobject-2.0, glib-2.0, gmodule-2.0"
examples: implement webrtcsink-stat-server
2021-11-30 21:49:33 +00:00
[[example]]
name = "webrtcsink-stats-server"
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