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webrtc: Enhance documentation

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Thibault Saunier 2022-10-20 12:25:32 +02:00 committed by Sebastian Dröge
parent c0bf05d4bb
commit cbdd3a7f26
4 changed files with 32 additions and 39 deletions
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1
docs/meson.build
View file

@ -97,6 +97,7 @@ foreach plugin_name: list_plugin_res.stdout().split(':')
sitemap: 'plugins/sitemap.txt',
index: 'plugins/index.md',
gst_index: 'plugins/index.md',
include_paths: join_paths(meson.current_source_dir(), '..'),
gst_smart_index: true,
gst_c_sources: [
'../*/*/*/*.rs',

35
net/webrtc/README.md
View file

@ -64,27 +64,10 @@ expose more interfaces to guide and tune the heuristics it employs.
## Building
### Prerequisites
The element has only been tested for now against GStreamer main.
For testing, it is recommended to simply build GStreamer locally and run
in the uninstalled devenv.
> Make sure to install the development packages for some codec libraries
> beforehand, such as libx264, libvpx and libopusenc, exact names depend
> on your distribution.
```
git clone --depth 1 --single-branch --branch main https://gitlab.freedesktop.org/gstreamer/gstreamer
cd gstreamer
meson build
ninja -C build
ninja -C build devenv
```
### Compiling
``` shell
cargo build
```
@ -110,11 +93,11 @@ export GST_PLUGIN_PATH=$PWD/target/debug:$GST_PLUGIN_PATH
gst-launch-1.0 webrtcsink name=ws videotestsrc ! ws. audiotestsrc ! ws.
```
When the pipeline above is running succesfully, open a browser and
When the pipeline above is running successfully, open a browser and
point it to the http server:
``` shell
xdg-open http://127.0.0.1:8000
gio open http://127.0.0.1:8000
```
You should see an identifier listed in the left-hand panel, click on
@ -133,12 +116,6 @@ with gst-launch:
gst-launch-1.0 webrtcsink signaller::address="ws://127.0.0.1:8443" ..
```
The signaller object can not be inspected, refer to [the source code]
for the list of properties.
[the source code]: plugins/src/signaller/imp.rs
### Enable 'navigation' a.k.a user interactivity with the content
`webrtcsink` implements the [`GstNavigation`] interface which allows interacting
@ -152,7 +129,7 @@ As an example, the following pipeline allows you to navigate the GStreamer
documentation inside the video running within your web browser (in
http://127.0.0.1:8000 if you followed previous steps of that readme):
```
``` shell
gst-launch-1.0 wpesrc location=https://gstreamer.freedesktop.org/documentation/ ! webrtcsink enable-data-channel-navigation=true
```
@ -181,7 +158,7 @@ My testing procedure was:
* Run a `comcast` command on the server machine, for instance:
``` shell
/home/meh/go/bin/comcast --device=$SERVER_INTERFACE --target-bw 3000 --target-addr=$CLIENT_IP --target-port=1:65535 --target-proto=udp
$HOME/go/bin/comcast --device=$SERVER_INTERFACE --target-bw 3000 --target-addr=$CLIENT_IP --target-port=1:65535 --target-proto=udp
```
* Observe the bitrate sharply decreasing, playback should slow down briefly
@ -191,7 +168,7 @@ My testing procedure was:
back to a maximum:
``` shell
/home/meh/go/bin/comcast --device=$SERVER_INTERFACE --stop
$HOME/go/bin/comcast --device=$SERVER_INTERFACE --stop
```
For comparison, the congestion control property can be set to disabled on
@ -210,7 +187,7 @@ gst-launch-1.0 webrtcsink congestion-control=disabled
An example server / client application for monitoring per-consumer stats
can be found [here].
[here]: plugins/examples/README.md
[here]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/tree/main/net/webrtc/examples
## License

29
net/webrtc/src/gcc/imp.rs
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@ -1,13 +1,22 @@
///! Implements https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02
///!
///! This element implements the pacing as describe in the spec by running its
///! own streaming thread on its srcpad. It implements the mathematic as closely
///! to the specs as possible and sets the #rtpgccbwe:estimated-bitrate property
///! each time a new estimate is produced. User should connect to the
///! `rtpgccbwe::notify::estimated-bitrate` signal to make the encoders target
///! that new estimated bitrate (the overall target bitrate of the potentially
///! multiple encore should match that target bitrate, the application is
///! responsible for determining what bitrate to give to each encode)
/**
* element-rtpgccbwe:
*
* Implements the [Google Congestion Control algorithm](https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02).
*
* This element should always be placed right before a `rtpsession` and will only work
* when [twcc](https://datatracker.ietf.org/doc/html/draft-holmer-rmcat-transport-wide-cc-extensions-01) is enabled
* as the bandwidth estimation relies on it.
*
* This element implements the pacing as describe in the spec by running its
* own streaming thread on its srcpad. It implements the mathematic as closely
* to the specs as possible and sets the #rtpgccbwe:estimated-bitrate property
* each time a new estimate is produced. User should connect to the
* `rtpgccbwe::notify::estimated-bitrate` signal to make the encoders target
* that new estimated bitrate (the overall target bitrate of the potentially
* multiple encoders should match that target bitrate, the application is
* responsible for determining what bitrate to give to each encode)
*
*/
use chrono::Duration;
use gst::{
glib::{self},

6
net/webrtc/src/webrtcsink/mod.rs
View file

@ -1,5 +1,11 @@
// SPDX-License-Identifier: MPL-2.0
/**
* element-webrtcsink:
*
* {{ net/webrtc/README.md[0:190] }}
*
*/
use gst::glib;
use gst::prelude::*;
use gst::subclass::prelude::*;