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webrtc: Specify to use playbin3 instead of playbin

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1995>
This commit is contained in:
Sebastian Dröge 2024-12-17 16:49:01 +02:00 committed by GStreamer Marge Bot
parent 248b7ac059
commit 7d4ddc7eb9
3 changed files with 5 additions and 5 deletions
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4
net/webrtc/README.md
View file

@ -136,11 +136,11 @@ Then open a new terminal and run:
``` shell
export GST_PLUGIN_PATH=<path-to-gst-plugins-rs>/target/debug:$GST_PLUGIN_PATH
gst-launch-1.0 playbin uri=gstwebrtc://127.0.0.1:8443?peer-id=[Client ID]
gst-launch-1.0 playbin3 uri=gstwebrtc://127.0.0.1:8443?peer-id=[Client ID]
```
Replacing the "peer-id" value with the previously copied "Client ID" value. You
should see the playbin element opening a window and showing you the content
should see the playbin3 element opening a window and showing you the content
produced by the web page.
## Configuration

4
net/webrtc/gstwebrtc-api/README.md
View file

@ -101,12 +101,12 @@ that appears on the webpage.
Then launch the following GStreamer pipeline:
```shell
$ gst-launch-1.0 playbin uri=gstwebrtc://[signalling server]?peer-id=[client ID of the producer]
$ gst-launch-1.0 playbin3 uri=gstwebrtc://[signalling server]?peer-id=[client ID of the producer]
```
Using the local signalling server, it will look like this:
```shell
$ gst-launch-1.0 playbin uri=gstwebrtc://127.0.0.1:8443?peer-id=e54e5d6b-f597-4e8f-bc96-2cc3765b6567
$ gst-launch-1.0 playbin3 uri=gstwebrtc://127.0.0.1:8443?peer-id=e54e5d6b-f597-4e8f-bc96-2cc3765b6567
```
The underlying *uridecodebin* element recognizes the *gstwebrtc://* scheme as a WebRTC stream compatible with the

2
net/webrtc/src/webrtcsrc/mod.rs
View file

@ -16,7 +16,7 @@
* gst-launch-1.0 webrtcsrc signaller::producer-peer-id=<webrtcsink-peer-id> ! videoconvert ! autovideosink
* ```
*
* or directly using `playbin`:
* or directly using `playbin3`:
*
* ``` bash
* gst-launch-1.0 playbin3 uri="gstwebrtc://localhost:8443?peer-id=<webrtcsink-peer-id>"