Skip to content

Commit

Permalink
add Unity instructions
Browse files Browse the repository at this point in the history
  • Loading branch information
@aler9
aler9 committed Oct 6, 2024
1 parent ca44582 commit cd406da
Showing 1 changed file with 260 additions and 11 deletions.
271 changes: 260 additions & 11 deletions README.md
View file
Original file line number Diff line number Diff line change
Expand Up @@ -22,7 +22,7 @@ Live streams can be published to the server with:
|--------|--------|------------|------------|
|[SRT clients](#srt-clients)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[SRT cameras and servers](#srt-cameras-and-servers)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[WebRTC clients](#webrtc-clients)|Browser-based, WHIP|AV1, VP9, VP8, H265, H264|Opus, G722, G711 (PCMA, PCMU)|
|[WebRTC clients](#webrtc-clients)|WHIP|AV1, VP9, VP8, H265, H264|Opus, G722, G711 (PCMA, PCMU)|
|[WebRTC servers](#webrtc-servers)|WHEP|AV1, VP9, VP8, H265, H264|Opus, G722, G711 (PCMA, PCMU)|
|[RTSP clients](#rtsp-clients)|UDP, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
|[RTSP cameras and servers](#rtsp-cameras-and-servers)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
Expand All @@ -32,17 +32,21 @@ Live streams can be published to the server with:
|[UDP/MPEG-TS](#udpmpeg-ts)|Unicast, broadcast, multicast|H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[Raspberry Pi Cameras](#raspberry-pi-cameras)||H264||

And can be read from the server with:
Instructions are provided for publishing through these protocols by using [FFmpeg](#ffmpeg), [GStreamer](#gstreamer), [OBS Studio](#obs-studio), [OpenCV](#opencv), [Unity](#unity), [Web browsers](#web-browsers).

Live streams can be read from the server with:

|protocol|variants|video codecs|audio codecs|
|--------|--------|------------|------------|
|[SRT](#srt)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[WebRTC](#webrtc)|Browser-based, WHEP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
|[WebRTC](#webrtc)|WHEP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
|[RTSP](#rtsp)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
|[RTMP](#rtmp)|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
|[HLS](#hls)|Low-Latency HLS, MP4-based HLS, legacy HLS|AV1, VP9, H265, H264|Opus, MPEG-4 Audio (AAC)|

And can be recorded and played back with:
Instructions are provided for reading through these protocols by using [FFmpeg](#ffmpeg-1), [GStreamer](#gstreamer-1), [VLC](#vlc), [Unity](#unity-1), [Web browsers](#web-browsers-1).

Live streams be recorded and played back with:

|format|video codecs|audio codecs|
|------|------------|------------|
Expand Down Expand Up @@ -83,6 +87,7 @@ _rtsp-simple-server_ has been rebranded as _MediaMTX_. The reason is pretty obvi
* [GStreamer](#gstreamer)
* [OBS Studio](#obs-studio)
* [OpenCV](#opencv)
* [Unity](#unity)
* [Web browsers](#web-browsers)
* [By device](#by-device)
* [Generic webcam](#generic-webcam)
Expand All @@ -103,6 +108,7 @@ _rtsp-simple-server_ has been rebranded as _MediaMTX_. The reason is pretty obvi
* [FFmpeg](#ffmpeg-1)
* [GStreamer](#gstreamer-1)
* [VLC](#vlc)
* [Unity](#unity-1)
* [Web browsers](#web-browsers-1)
* [By protocol](#by-protocol-1)
* [SRT](#srt)
Expand Down Expand Up @@ -347,7 +353,7 @@ The resulting stream will be available in path `/mystream`.

#### OpenCV

OpenCV can publish to the server through its GStreamer plugin, as a [RTSP client](#rtsp-clients). It must be compiled with GStreamer support, by following this procedure:
Software which uses the OpenCV library can publish to the server through its GStreamer plugin, as a [RTSP client](#rtsp-clients). It must be compiled with GStreamer support, by following this procedure:

```sh
sudo apt install -y libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev gstreamer1.0-plugins-ugly gstreamer1.0-rtsp python3-dev python3-numpy
Expand All @@ -367,7 +373,7 @@ python3 -c 'import cv2; print(cv2.getBuildInformation())'

Check that the output contains `GStreamer: YES`.

Videos can be published with `VideoWriter`:
Videos can be published with `cv2.VideoWriter`:

```python
from datetime import datetime
Expand Down Expand Up @@ -420,6 +426,117 @@ while True:

The resulting stream will be available in path `/mystream`.

#### Unity

Software written with the Unity Engine can publish a stream to the server by using the [WebRTC protocol](#webrtc).

Create a new Unity project or open an existing open.

Open _Window -> Package Manager_, click on the plus sign, _Add Package by name..._ and insert `com.unity.webrtc`. Wait for the package to be installed.

In the _Project_ window, under `Assets`, create a new C# Script called `WebRTCPublisher.cs` with this content:

```cs
using System.Collections;
using UnityEngine;
using Unity.WebRTC;
using UnityEngine.Networking;

public class WebRTCPublisher : MonoBehaviour
{
public string url = "http://localhost:8889/unity/whip";
public int videoWidth = 1280;
public int videoHeight = 720;

private RTCPeerConnection pc;
private MediaStream videoStream;

void Start()
{
pc = new RTCPeerConnection();
Camera sourceCamera = gameObject.GetComponent<Camera>();
videoStream = sourceCamera.CaptureStream(videoWidth, videoHeight);
foreach (var track in videoStream.GetTracks())
{
pc.AddTrack(track);
}

StartCoroutine(WebRTC.Update());
StartCoroutine(createOffer());
}

private IEnumerator createOffer()
{
var op = pc.CreateOffer();
yield return op;
if (op.IsError) {
Debug.LogError("CreateOffer() failed");
yield break;
}

yield return setLocalDescription(op.Desc);
}

private IEnumerator setLocalDescription(RTCSessionDescription offer)
{
var op = pc.SetLocalDescription(ref offer);
yield return op;
if (op.IsError) {
Debug.LogError("SetLocalDescription() failed");
yield break;
}

yield return postOffer(offer);
}

private IEnumerator postOffer(RTCSessionDescription offer)
{
var content = new System.Net.Http.StringContent(offer.sdp);
content.Headers.ContentType = new System.Net.Http.Headers.MediaTypeHeaderValue("application/sdp");
var client = new System.Net.Http.HttpClient();

var task = System.Threading.Tasks.Task.Run(async () => {
var res = await client.PostAsync(new System.UriBuilder(url).Uri, content);
res.EnsureSuccessStatusCode();
return await res.Content.ReadAsStringAsync();
});
yield return new WaitUntil(() => task.IsCompleted);
if (task.Exception != null) {
Debug.LogError(task.Exception);
yield break;
}

yield return setRemoteDescription(task.Result);
}

private IEnumerator setRemoteDescription(string answer)
{
RTCSessionDescription desc = new RTCSessionDescription();
desc.type = RTCSdpType.Answer;
desc.sdp = answer;
var op = pc.SetRemoteDescription(ref desc);
yield return op;
if (op.IsError) {
Debug.LogError("SetRemoteDescription() failed");
yield break;
}

yield break;
}

void OnDestroy()
{
pc?.Close();
pc?.Dispose();
videoStream?.Dispose();
}
}
```

In the _Hierarchy_ window, find or create a scene and a camera, then add the `WebRTCPublisher.cs` script as component of the camera, by dragging it inside the _Inspector_ window. then Press the _Play_ button at the top of the page.

The resulting stream will be available in path `/unity`.

#### Web browsers

Web browsers can publish a stream to the server by using the [WebRTC protocol](#webrtc). Start the server and open the web page:
Expand All @@ -442,7 +559,7 @@ For more advanced setups, you can create and serve a custom web page by starting

#### Generic webcam

If the OS is Linux-based, edit `mediamtx.yml` and replace everything inside section `paths` with the following content:
If the operating system is Linux-based, edit `mediamtx.yml` and replace everything inside section `paths` with the following content:

```yml
paths:
Expand All @@ -451,7 +568,7 @@ paths:
runOnInitRestart: yes
```
If the OS is Windows:
If the operating system is Windows:
```yml
paths:
Expand All @@ -470,7 +587,7 @@ The resulting stream will be available in path `/cam`.

#### Raspberry Pi Cameras

_MediaMTX_ natively supports the Raspberry Pi Camera, enabling high-quality and low-latency video streaming from the camera to any user, for any purpose. There are a couple of requirements:
_MediaMTX_ natively supports most of the Raspberry Pi Camera models, enabling high-quality and low-latency video streaming from the camera to any user, for any purpose. There are a couple of requirements:

1. The server must run on a Raspberry Pi, with one of the following operating systems:

Expand Down Expand Up @@ -617,7 +734,7 @@ Regarding authentication, read [Authenticating with WHIP/WHEP](#authenticating-w

Depending on the network it may be difficult to establish a connection between server and clients, read [Solving WebRTC connectivity issues](#solving-webrtc-connectivity-issues).

Known clients that can publish with WebRTC and WHIP are [FFmpeg](#ffmpeg), [GStreamer](#gstreamer), [OBS Studio](#obs-studio).
Known clients that can publish with WebRTC and WHIP are [FFmpeg](#ffmpeg), [GStreamer](#gstreamer), [OBS Studio](#obs-studio), [Unity](#unity) and [Web browsers](#web-browsers).

#### WebRTC servers

Expand Down Expand Up @@ -813,6 +930,138 @@ snap install vlc

At the moment VLC doesn't support reading encrypted RTSP streams. However, you can use a proxy like [stunnel](https://www.stunnel.org) or [nginx](https://nginx.org/) or a local _MediaMTX_ instance to decrypt streams before reading them.

#### Unity

Software written with the Unity Engine can read a stream from the server by using the [WebRTC protocol](#webrtc).

Create a new Unity project or open an existing open.

Open _Window -> Package Manager_, click on the plus sign, _Add Package by name..._ and insert `com.unity.webrtc`. Wait for the package to be installed.

In the _Project_ window, under `Assets`, create a new C# Script called `WebRTCReader.cs` with this content:

```cs
using System.Collections;
using UnityEngine;
using Unity.WebRTC;
public class WebRTCReader : MonoBehaviour
{
public string url = "http://localhost:8889/stream/whep";
private RTCPeerConnection pc;
private MediaStream receiveStream;
void Start()
{
UnityEngine.UI.RawImage rawImage = gameObject.GetComponentInChildren<UnityEngine.UI.RawImage>();
AudioSource audioSource = gameObject.GetComponentInChildren<AudioSource>();
pc = new RTCPeerConnection();
receiveStream = new MediaStream();
pc.OnTrack = e =>
{
receiveStream.AddTrack(e.Track);
};
receiveStream.OnAddTrack = e =>
{
if (e.Track is VideoStreamTrack videoTrack)
{
videoTrack.OnVideoReceived += (tex) =>
{
rawImage.texture = tex;
};
}
else if (e.Track is AudioStreamTrack audioTrack)
{
audioSource.SetTrack(audioTrack);
audioSource.loop = true;
audioSource.Play();
}
};
RTCRtpTransceiverInit init = new RTCRtpTransceiverInit();
init.direction = RTCRtpTransceiverDirection.RecvOnly;
pc.AddTransceiver(TrackKind.Audio, init);
pc.AddTransceiver(TrackKind.Video, init);
StartCoroutine(WebRTC.Update());
StartCoroutine(createOffer());
}
private IEnumerator createOffer()
{
var op = pc.CreateOffer();
yield return op;
if (op.IsError) {
Debug.LogError("CreateOffer() failed");
yield break;
}
yield return setLocalDescription(op.Desc);
}
private IEnumerator setLocalDescription(RTCSessionDescription offer)
{
var op = pc.SetLocalDescription(ref offer);
yield return op;
if (op.IsError) {
Debug.LogError("SetLocalDescription() failed");
yield break;
}
yield return postOffer(offer);
}
private IEnumerator postOffer(RTCSessionDescription offer)
{
var content = new System.Net.Http.StringContent(offer.sdp);
content.Headers.ContentType = new System.Net.Http.Headers.MediaTypeHeaderValue("application/sdp");
var client = new System.Net.Http.HttpClient();
var task = System.Threading.Tasks.Task.Run(async () => {
var res = await client.PostAsync(new System.UriBuilder(url).Uri, content);
res.EnsureSuccessStatusCode();
return await res.Content.ReadAsStringAsync();
});
yield return new WaitUntil(() => task.IsCompleted);
if (task.Exception != null) {
Debug.LogError(task.Exception);
yield break;
}
yield return setRemoteDescription(task.Result);
}
private IEnumerator setRemoteDescription(string answer)
{
RTCSessionDescription desc = new RTCSessionDescription();
desc.type = RTCSdpType.Answer;
desc.sdp = answer;
var op = pc.SetRemoteDescription(ref desc);
yield return op;
if (op.IsError) {
Debug.LogError("SetRemoteDescription() failed");
yield break;
}
yield break;
}
void OnDestroy()
{
pc?.Close();
pc?.Dispose();
receiveStream?.Dispose();
}
}
```

Edit the `url` variable according to your needs.

In the _Hierarchy_ window, find or create a scene. Inside the scene, add a Canvas. Inside the Canvas, add a _Raw Image_ and an _Audio Source_. Then add the `WebRTCReader.cs` script as component of the canvas, by dragging it inside the _Inspector_ window. then Press the _Play_ button at the top of the page.

#### Web browsers

Web browsers can read a stream from the server in multiple ways (WebRTC or HLS).
Expand Down Expand Up @@ -885,7 +1134,7 @@ Regarding authentication, read [Authenticating with WHIP/WHEP](#authenticating-w

Depending on the network it may be difficult to establish a connection between server and clients, read [Solving WebRTC connectivity issues](#solving-webrtc-connectivity-issues).

Known clients that can read with WebRTC and WHEP are [FFmpeg](#ffmpeg-1), [GStreamer](#gstreamer-1) and [web browsers](#web-browsers-1).
Known clients that can read with WebRTC and WHEP are [FFmpeg](#ffmpeg-1), [GStreamer](#gstreamer-1), [Unity](#unity-1) and [web browsers](#web-browsers-1).

#### RTSP

Expand Down

0 comments on commit cd406da

Please sign in to comment.