gst-plugins-rs/net/webrtc/examples/README.md

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# webrtcsink examples
Collection of webrtcsink examples
## webrtcsink-stats-server
A simple application that instantiates a webrtcsink and serves stats
over websockets.
The application expects a signalling server to be running at `ws://localhost:8443`,
similar to the usage example in the main README.
``` shell
cargo run --example webrtcsink-stats-server
```
Once it is running, follow the instruction in the webrtcsink-stats folder to
run an example client.
## webrtcsink-custom-signaller
An example of custom signaller implementation, see the corresponding
[README](webrtcsink-custom-signaller/README.md) for more details on code and usage.
## WebRTC precise synchronization example
This example demonstrates a sender / receiver setup which ensures precise
synchronization of multiple streams in a single session.
[RFC 6051]-style rapid synchronization of RTP streams is available as an option.
Se the [Instantaneous RTP synchronization...] blog post for details about this
mode and an example based on RTSP instead of WebRTC.
The examples can also be used for [RFC 7273] NTP or PTP clock signalling and
synchronization.
[RFC 6051]: https://datatracker.ietf.org/doc/html/rfc6051
[RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273
[Instantaneous RTP synchronization...]: https://coaxion.net/blog/2022/05/instantaneous-rtp-synchronization-retrieval-of-absolute-sender-clock-times-with-gstreamer/
### Signaller
The example uses the default WebRTC signaller. Launch it using the following
command:
```shell
cargo run --bin gst-webrtc-signalling-server
```
### Receiver
The receiver awaits for new audio & video stream publishers and render the
streams using auto sink elements. Launch it using the following command:
```shell
cargo r --example webrtc-precise-sync-recv
```
The default configuration should work for a local test. For a multi-host setup,
see the available options:
```shell
cargo r --example webrtc-precise-sync-recv -- --help
```
E.g.: the following will force `avdec_h264` over hardware decoders, activate
debug logs for the receiver and connect to the signalling server at the
specified address:
```shell
GST_PLUGIN_FEATURE_RANK=avdec_h264:MAX \
WEBRTC_PRECISE_SYNC_RECV_LOG=debug \
cargo r --example webrtc-precise-sync-recv -- --server 192.168.1.22
```
### Sender
The sender publishes audio & video test streams. Launch it using the following
command:
```shell
cargo r --example webrtc-precise-sync-send
```
The default configuration should work for a local test. For a multi-host setup,
to set the number of audio / video streams, to enable rapid synchronization or
to force the video encoder, see the available options:
```shell
cargo r --example webrtc-precise-sync-send -- --help
```
E.g.: the following will force H264 and `x264enc` over hardware encoders,
activate debug logs for the sender and connect to the signalling server at the
specified address:
```shell
GST_PLUGIN_FEATURE_RANK=264enc:MAX \
WEBRTC_PRECISE_SYNC_SEND_LOG=debug \
cargo r --example webrtc-precise-sync-send -- \
--server 192.168.1.22 --video-caps video/x-h264
```
### The pipeline latency
The `--pipeline-latency` argument configures a static latency of 1s by default.
This needs to be higher than the sum of the sender latency and the receiver
latency of the receiver with the highest latency. As this can't be known
automatically and depends on many factors, this has to be known for the overall
system and configured accordingly.
The default configuration is on the safe side and favors synchronization over
low latency. Depending on the use case, shorter or larger values should be used.
### RFC 7273 NTP or PTP clock signalling and synchronization
For [RFC 7273] NTP or PTP clock signalling and synchronization, you can use
commands such as:
#### Receiver
```shell
cargo r --example webrtc-precise-sync-recv -- --expect-clock-signalling
```
#### Sender
```shell
cargo r --example webrtc-precise-sync-send -- --clock ntp --do-clock-signalling \
--video-streams 0 --audio-streams 2
```
webrtc: add android webrtcsrc example This commit adds an Android `webrtcsrc` based example with the following features: * A first view allows retrieving the producer list from the signaller (peer ids are uuids which are too long to tap, especially using an onscreen keyboard). * Selecting a producer opens a second view. The first available video stream is rendered on a native Surface. All the audio streams are rendered using `autoaudiosink`. Available Settings: * Signaller URI. * A toggle to prefer hardware decoding for OPUS, otherwise the app defaults to raising `opusdec`'s rank. Hardware decoding was moved aside since it was found to crash the app on all tested devices (2 smartphones, 1 tv). **Warning**: in order to ease testing, this demonstration application enables unencrypted network communication. See `AndroidManifest.xml`. The application uses the technologies currenlty proposed by Android Studio when creating a new project: * Kotlin as the default language, which is fully interoperable with Java and uses the same SDK. * gradle 8.6. * kotlin dialect for gradle. The structure is mostly the same as the previously preferred dialect, for which examples can be found online readily. * However, JNI code generation still uses Makefiles (instead of CMake) due to the need to call [`gstreamer-1.0.mk`] for `gstreamer_android` generation. Note: on-going work on that front: - https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/1466 - https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6794 Current limitations: * x86 support is currently discarded as `gstreamer_android` libs generation fails (observed with `gstreamer-1.0-android-universal-1.24.3`). * A selector could be added to let the user chose the video streams and possibly decide whether to render all audio streams or just select one. Nice to have: * Support for the synchronization features of the `webrtc-precise-sync-recv` example (NTP clock, RFC 7273). * It could be nice to use Rust for the specific native code. [`gstreamer-1.0.mk`]: https://gitlab.freedesktop.org/gstreamer/cerbero/-/blob/main/data/ndk-build/gstreamer-1.0.mk Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1578>
2024-05-24 15:33:46 +00:00
## Android
### `webrtcsrc` based Android application
An Android demonstration application which retrieves available producers from
the signaller and renders audio and video streams.
**Important**: in order to ease testing, this demonstration application enables
unencrypted network communication. See `app/src/main/AndroidManifest.xml` for
details.
#### Build the application
* Download the latest Android prebuilt binaries from:
https://gstreamer.freedesktop.org/download/
* Uncompress / untar the package, e.g. under `/opt/android/`.
* Define the `GSTREAMER_ROOT_ANDROID` environment variable with the
directory chosen at previous step.
* Install a recent version of Android Studio (tested with 2023.3.1.18).
* Open the project from the folder `android/webrtcsrc`.
* Have Android Studio download and install the required SDK & NDK.
* Click the build button or build and run on the target device.
* The resulting `apk` is generated under:
`android/webrtcsrc/app/build/outputs/apk/debug`.
For more details, refer to:
* https://gstreamer.freedesktop.org/documentation/installing/for-android-development.html
Once the SDK & NDK are installed, you can use `gradlew` to build and install
the apk (make sure the device is visible from adb):
```shell
# From the android/webrtcsrc directory
./gradlew installDebug
```
#### Install the application
Prerequisites: activate developer mode on the target device.
There are several ways to install the application:
* The easiest is to click the run button in Android Studio.
* You can also install the `apk` using `adb`.
Depending on your host OS, you might need to define `udev` rules. See:
https://github.com/M0Rf30/android-udev-rules
#### Setup
1. Run the Signaller from the `gst-plugins-rs` root directory:
```shell
cargo run --bin gst-webrtc-signalling-server
```
2. In the Android app, tap the 3 dots button -> Settings and edit the Signaller
URI.
3. Add a producer, e.g. using `gst-launch` & `webrtcsink` or run:
```shell
cargo r --example webrtc-precise-sync-send
```
4. Click the `Refresh` button on the Producer List view of the app.