## π₯ Stream Chat and Voice & Video Calling SDK
__[Stream Video SDK for Compose](https://getstream.io/video/docs/android?utm_source=Github&utm_medium=Jaewoong_OSS&utm_content=Developer&utm_campaign=Github_Feb2023_Jaewoong_StreamWebRTCAndroid&utm_term=DevRelOss)__ is the official Android SDK for [Stream Video](https://getstream.io/video/?utm_source=Github&utm_medium=Github_Repo_Content_Ad&utm_content=Developer&utm_campaign=Github_Feb2023_Jaewoong_StreamWebRTCAndroid&utm_term=DevRelOss), a service for building video calls, audio rooms, and live-streaming applications. Stream's versatile Video SDK has been built with this **webrtc-android** library, and you can check out the tutorials below if you want to get more information.
- [Stream Video SDK for Android on GitHub](https://github.com/getstream/stream-video-android?utm_source=Github&utm_medium=Jaewoong_OSS&utm_content=Developer&utm_campaign=Github_Dec2023_Jaewoong_TwitchCompose&utm_term=DevRelOss)
- [Video Call Tutorial](https://getstream.io/video/docs/android/tutorials/video-calling?utm_source=Github&utm_medium=Jaewoong_OSS&utm_content=Developer&utm_campaign=Github_Dec2023_Jaewoong_TwitchCompose&utm_term=DevRelOss)
- [Audio Room Tutorial](https://getstream.io/video/docs/android/tutorials/audio-room?utm_source=Github&utm_medium=Jaewoong_OSS&utm_content=Developer&utm_campaign=Github_Dec2023_Jaewoong_TwitchCompose&utm_term=DevRelOss)
- [Livestream Tutorial](https://getstream.io/video/docs/android/tutorials/livestream?utm_source=Github&utm_medium=Jaewoong_OSS&utm_content=Developer&utm_campaign=Github_Dec2023_Jaewoong_TwitchCompose&utm_term=DevRelOss)
## Download
[](https://search.maven.org/search?q=g:%22io.getstream%22%20AND%20a:%22stream-webrtc-android%22)
### Gradle
Add the below dependency to your **module**'s `build.gradle` file:
```gradle
dependencies {
implementation "io.getstream:stream-webrtc-android:1.1.1"
}
```
## SNAPSHOT
## WebRTC for UI Components
[](https://search.maven.org/search?q=g:%22io.getstream%22%20AND%20a:%22stream-webrtc-android-ui%22)
**Stream WebRTC Android** supports some useful UI components for WebRTC, such as `VideoTextureViewRenderer`. First, add the dependency below to your **module's** `build.gradle` file:
```gradle
dependencies {
implementation "io.getstream:stream-webrtc-android-ui:$version"
}
```
### VideoTextureViewRenderer
`VideoTextureViewRenderer` is a custom [TextureView](https://developer.android.com/reference/android/view/TextureView) that implements [VideoSink](https://getstream.github.io/webrtc-android/stream-webrtc-android/org.webrtc/-video-sink/index.html?query=interface%20VideoSink) and [SurfaceTextureListener](https://developer.android.com/reference/kotlin/android/view/TextureView.SurfaceTextureListener.html).
Usually, you can use [SurfaceViewRenderer](https://getstream.github.io/webrtc-android/stream-webrtc-android/org.webrtc/-surface-view-renderer/index.html?query=open%20class%20SurfaceViewRenderer%20:%20SurfaceView,%20SurfaceHolder.Callback,%20VideoSink,%20RendererCommon.RendererEvents) to display real-time video streams on a layout if you need a simple video call screen without overlaying video frames over another one. However, it might not work well as you expect if you suppose to need to design a complex video call screen, such as one video call layout should overlay another video call layout, such as the example below:
For this case, we'd recommend you use `VideoTextureViewRenderer` like the example below:
```xml
## WebRTC for Jetpack Compose
[](https://search.maven.org/search?q=g:%22io.getstream%22%20AND%20a:%22stream-webrtc-android-compose%22)
**Stream WebRTC Android** supports some Jetpack Compose components for WebRTC, such as `VideoRenderer` and `FloatingVideoRenderer`. First, add the dependency below to your **module's** `build.gradle` file:
```gradle
dependencies {
implementation "io.getstream:stream-webrtc-android-compose:$version"
}
```
### VideoRenderer
`VideoRenderer` is a composable function that renders a single video track in Jetpack Compose.
```kotlin
VideoRenderer(
videoTrack = remoteVideoTrack,
modifier = Modifier.fillMaxSize()
eglBaseContext = eglBaseContext,
rendererEvents = rendererEvents
)
```
You can observe the rendering state changes by giving `RendererEvents` interface like the below:
```kotlin
val rendererEvents = object : RendererEvents {
override fun onFirstFrameRendered() { .. }
override fun onFrameResolutionChanged(videoWidth: Int, videoHeight: Int, rotation: Int) { .. }
}
```
### FloatingVideoRenderer
`FloatingVideoRenderer` represents a floating item that features a participant video, usually the local participant. You can use this composable function to overlay a single video track on another, and users can move the floating video track with user interactions.
You can use `FloatingVideoRenderer` with `VideoRenderer` like the example below:
```kotlin
var parentSize: IntSize by remember { mutableStateOf(IntSize(0, 0)) }
if (remoteVideoTrack != null) {
VideoRenderer(
videoTrack = remoteVideoTrack,
modifier = Modifier
.fillMaxSize()
.onSizeChanged { parentSize = it },
eglBaseContext = eglBaseContext,
rendererEvents = rendererEvents
)
}
if (localVideoTrack != null) {
FloatingVideoRenderer(
modifier = Modifier
.size(width = 150.dp, height = 210.dp)
.clip(RoundedCornerShape(16.dp))
.align(Alignment.TopEnd),
videoTrack = localVideoTrack,
parentBounds = parentSize,
paddingValues = PaddingValues(0.dp),
eglBaseContext = eglBaseContexteglBaseContext,
rendererEvents = rendererEvents
)
}
```
## WebRTC KTX
[](https://search.maven.org/search?q=g:%22io.getstream%22%20AND%20a:%22stream-webrtc-android-ktx%22)
**Stream WebRTC Android** supports some useful extensions for WebRTC based on [Kotlin's Coroutines](https://kotlinlang.org/docs/coroutines-overview.html). First, add the dependency below to your **module's** `build.gradle` file:
```gradle
dependencies {
implementation "io.getstream:stream-webrtc-android-ktx:$version"
}
```
### addRtcIceCandidate
`addRtcIceCandidate` is a suspend function that allows you to add a given `IceCandidate` to a `PeerConnection`. So you can add an `IceCandidate` to a `PeerConnection` as Coroutines-style, not callback-style.
```kotlin
pendingIceMutex.withLock {
pendingIceCandidates.forEach { iceCandidate ->
connection.addRtcIceCandidate(iceCandidate)
}
pendingIceCandidates.clear()
}
```
### createSessionDescription
You can create a `SessionDescription`, which delegates `SdpObserver` with Coroutines styles:
```kotlin
suspend fun createAnswer(): Result
> **Note**: Chromium WebRTC core libraries can be bulit only in Linux OS. Every step takes its own time based on the machine specs and internet speed, so make sure every step is completed without interruption.
You need to set up AWS instance (pre-requiests):
- Ubuntu 14.04 LTS (trusty with EoL April 2022)
- 8 GB memory ram
- At least 50 GB HDD/SSD storage
To compile the pre-built WebRTC library for Android, you must follow the steps below:
```
1. git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git
2. export PATH="$PATH:${HOME}/depot_tools"
3. mkdir webrtc_android && cd webrtc_android
4. fetch --nohooks webrtc_android
5. gclient sync
6. cd src && ./build/install-build-deps.sh
7. git branch -r
8. git checkout origin/master
# To check you're in origin/master branch and check out to a specific branch if you want.
9. git branch
10. Replace Android sources & NDK/C/C++ files with this repository.
11. tools_webrtc/android/build_aar.py
```
To install all required dependencies for linux, a script is provided for Ubuntu, which is unfortunately only available after your first gclient sync and make sure your current directory is `webrtc_android/src/`:
```
cd src && ./build/install-build-deps.sh
```
You can see the available latest branches looks like the screenshoos below:
Now you can checkout to the latest branch which is `branch-heads/m79` or something, using this command:
```
git checkout branch-heads/m79
```
However, this project reflects the latest updates for WebRTC, so you must check out to the master branch like this:
```
8. git checkout origin/master
```
This will help you to resolve most of compilation issues. To get the details about your current branch you can simply use these commands:
```
9. git branch
```
### Using Manual Compilation:
This process will manually compile the source code for each particular CPU type. Manual Compiling involves these two steps:
1. Generate projects using GN.
2. Compile using Ninja.
This step will compile the library for Debug and Release modes of Development.
Ensure your current working directory is webrtc_android/src/ of your workspace. Then run:
```
11. gn gen out/Debug --args='target_os="android" target_cpu="arm"'
11. gn gen out/Release --args='is_debug=false is_component_build=false rtc_include_tests=false target_os="android" target_cpu="arm"'
```
You can specify a directory of your own choice instead of out/Debug, to enable managing multiple configurations in parallel.
- To build for ARM64: use target_cpu="arm64"
- To build for 32-bit x86: use target_cpu="x86"
- To build for 64-bit x64: use target_cpu="x64"
For compilation you can simply use these following commands for (out/Debug, out/Release):
```
11. ninja -C out/Debug
11. ninja -C out/Release
```
### Using AAR Build Tools:
This is the most simple process, which compiles the source code for all supported CPU types such as:
- arm64-v8a
- armeabi-v7a
- x86
- x86_64
After compiling the package, it includes all these native libraries and `.jar` library into `*.aar` file.
Make sure your current working directory is `webrtc_android/src/` of your workspace. Then run:
```
11. tools_webrtc/android/build_aar.py
```
This process will take some time based on your machine specs and internet speed, so here we go:
Now, if you look in the `webrtc_android/src/` directory, It turns out that you will end up with the compilation and building of `libwebrtc.aar`.
## Find this Android library useful? π
Support it by joining __[stargazers](https://github.com/getStream/stream-webrtc-android/stargazers)__ for this repository. βοΈ