项目概述
本文分享的视频通话应用程序由两个主要部分组成：
- WebRTC 集成：处理点对点视频通信。
- Firebase 集成：管理信令和用户状态更新。
我们将使用 Hilt 进行依赖注入，Kotlin 协程进行异步编程，并采用简洁的架构方法来保持代码的模块化和可维护性。
设置项目
首先，创建一个新的 Kotlin 项目，并为 WebRTC、Firebase 和 Hilt 添加必要的依赖关系，以实现依赖注入。
依赖项
在 build.gradle 文件中添加以下依赖项：
dependencies {
.
.
implementation 'com.google.code.gson:gson:2.10.1'
implementation 'com.mesibo.api:webrtc:1.0.5'
implementation 'com.google.dagger:hilt-android:2.51'
kapt "com.google.dagger:hilt-compiler:2.51"
}关键组件
- MainService：管理视频通话的整体流程，包括初始化服务、处理来电和去电以及管理 UI 元素。
- MainRepository：充当服务和 WebRTC/Firebase 客户端之间的中介，处理数据流和状态管理。
- NSWebRTCClient：用于管理 WebRTC 连接的自定义客户端，包括处理媒体流、ICE 候选和会话描述。
数据流
- UI 交互：
- 用户与 UI（
Activity或Fragment）交互，以开始或接听电话。
- 服务通信：
- UI 向
MainService发送请求，以启动或管理呼叫。
- 业务逻辑：
MainService将请求委托给MainRepository，以处理呼叫设置或更新。
- 数据处理：
MainRepository与WebRTCClient交互管理视频通话，并与 FirebaseClient 交互同步通话相关数据。
- 实时操作：
WebRTCClient负责管理实时视频和音频流。FirebaseClient可确保呼叫相关事件在客户端之间实时同步和更新。
- 更新：
MainRepository接收来自WebRTCClient和FirebaseClient的更新，并通知MainService。MainService会根据最新的呼叫状态和数据更新 UI（Activity或Fragment）。
MainService 主服务
类MainService管理应用程序的核心功能，例如启动和停止服务、设置视图和处理呼叫事件。
@Singleton
class MainService @Inject constructor(private val mainRepository: MainRepository) : MainRepository.MainRepositoryListener {
companion object {
var incomingCallListener: IncomingCallListener? = null
var endCallListener: EndCallListener? = null
var localSurfaceView: SurfaceViewRenderer? = null
var remoteSurfaceView: SurfaceViewRenderer? = null
}
interface EndCallListener {
fun onCallEnded()
}
interface IncomingCallListener {
fun onCallReceived(model: NSDataModel)
}
/**
* Initializes the WebRTC client and Firebase client, sets the repository listener,
* and starts the service for the given username.
*
* @param username The username of the current user.
*/
fun startService(username: String) {
mainRepository.mainRepositoryListener = this
CoroutineScope(Dispatchers.IO).launch {
mainRepository.initWebrtcClient(username)
mainRepository.initFirebase()
}
}
/**
* Sets up the views for displaying video streams, initializes the local and remote
* surface views, and starts the call if not initiated by the caller.
*
* @param videoCall Indicates if the call is a video call.
* @param caller Indicates if the current user is the caller.
* @param target The target username to connect with.
*/
fun setupViews(videoCall: Boolean, caller: Boolean, target: String) {
mainRepository.setTarget(target)
mainRepository.initLocalSurfaceView(localSurfaceView!!, videoCall)
mainRepository.initRemoteSurfaceView(remoteSurfaceView!!)
if (!caller) {
mainRepository.startCall()
}
}
/**
* Receives and handles the latest event data, such as incoming call requests.
*
* @param data The data model containing event information.
*/
override fun onLatestEventReceived(data: NSDataModel) {
if (data.isValid()) {
when (data.type) {
NSDataModelType.StartVideoCall, NSDataModelType.StartAudioCall -> {
incomingCallListener?.onCallReceived(data)
}
else -> Unit
}
}
}
/**
* Ends the call and notifies the listener.
*/
override fun endCall() {
mainRepository.endCall()
endCallListener?.onCallEnded()
}
}主要方法：
startService：初始化 WebRTC 客户端和 Firebase，为应用程序接听来电和拨出电话做好准备。setupViews：配置显示视频流的本地和远程视图。onLatestEventReceived：处理来电等事件，让 UI 做出适当响应。
MainRepository
类MainRepository负责与 WebRTC 客户端和 Firebase 进行交互。它处理呼叫发起、终止和状态更改的逻辑。
@Singleton
class MainRepository @Inject constructor(
private val firebaseClient: FirebaseClient,
private val webRTCClient: NSWebRTCClient,
private val gson: Gson
) : NSWebRTCClient.Listener {
private var target: String? = null
var mainRepositoryListener: MainRepositoryListener? = null
interface MainRepositoryListener {
fun onLatestEventReceived(data: NSDataModel)
fun endCall()
}
/**
* Initializes the WebRTC client with the given username and sets up peer connection observers.
*
* @param username The username of the current user.
*/
fun initWebrtcClient(username: String) {
webRTCClient.listener = this
webRTCClient.initializeWebrtcClient(username, object : NSPeerObserver() {
override fun onAddStream(p0: MediaStream?) {
// When a remote media stream is added, set it to the remote view
p0?.videoTracks?.get(0)?.addSink(remoteView)
}
override fun onIceCandidate(p0: IceCandidate?) {
// Send the ICE candidate to the remote peer
p0?.let {
webRTCClient.sendIceCandidate(target!!, it)
}
}
override fun onConnectionChange(newState: PeerConnection.PeerConnectionState?) {
// Handle connection state changes
if (newState == PeerConnection.PeerConnectionState.CONNECTED) {
firebaseClient.clearLatestEvent()
}
}
})
}
/**
* Initializes the Firebase client to listen for events and handles them based on event type.
*/
fun initFirebase() {
firebaseClient.subscribeForLatestEvent { event ->
mainRepositoryListener?.onLatestEventReceived(event)
when (event.type) {
Offer -> {
webRTCClient.onRemoteSessionReceived(
SessionDescription(SessionDescription.Type.OFFER, event.data.toString())
)
webRTCClient.answer(target!!)
}
Answer -> {
webRTCClient.onRemoteSessionReceived(
SessionDescription(SessionDescription.Type.ANSWER, event.data.toString())
)
}
IceCandidates -> {
// Parse and add ICE candidates received from the remote peer
val candidate = gson.fromJson(event.data.toString(), IceCandidate::class.java)
candidate?.let { webRTCClient.addIceCandidateToPeer(it) }
}
EndCall -> {
mainRepositoryListener?.endCall()
}
else -> Unit
}
}
}
/**
* Initiates a call to the target user.
*/
fun startCall() {
webRTCClient.call(target!!)
}
/**
* Ends the current call and cleans up resources.
*/
fun endCall() {
webRTCClient.closeConnection()
}
}NSWebRTCClient
类NSWebRTCClient处理 WebRTC 操作，包括媒体流和连接管理。
查看服务器的此链接：https：//gist.github.com/sagivo/3a4b2f2c7ac6e1b5267c2f1f59ac6c6b
class NSWebRTCClient(private val context: Context) {
private val peerConnectionFactory: PeerConnectionFactory by lazy { createPeerConnectionFactory() }
private var peerConnection: PeerConnection? = null
private val iceServer = listOf(
PeerConnection.IceServer.builder("YOUR_URL")
.setUsername("YOUR_USERNAME")
.setPassword("YOUR_PASSWORD").createIceServer()
)
private val localVideoSource by lazy { peerConnectionFactory.createVideoSource(false) }
private val localAudioSource by lazy { peerConnectionFactory.createAudioSource(MediaConstraints()) }
private val videoCapturer = getVideoCapturer(context)
private var localStream: MediaStream? = null
private var localAudioTrack: AudioTrack? = null
private var localVideoTrack: VideoTrack? = null
init {
// Initialize the PeerConnectionFactory with the application context
PeerConnectionFactory.initialize(
PeerConnectionFactory.InitializationOptions.builder(context)
.setFieldTrials("WebRTC-H264HighProfile/Enabled/")
.createInitializationOptions()
)
}
/**
* Initializes the WebRTC client with the specified username and sets up the peer connection.
*
* @param username The username of the current user.
* @param observer The observer for peer connection events.
*/
fun initializeWebrtcClient(username: String, observer: PeerConnection.Observer) {
peerConnection = createPeerConnection(observer)
startLocalStreaming()
}
/**
* Creates a peer connection with the specified observer.
*
* @param observer The observer for peer connection events.
* @return The created peer connection.
*/
private fun createPeerConnection(observer: PeerConnection.Observer): PeerConnection? {
return peerConnectionFactory.createPeerConnection(iceServer, observer)
}
/**
* Starts a call by creating an SDP offer and setting it as the local description.
*
* @param target The target user to call.
*/
fun call(target: String) {
peerConnection?.createOffer(object : MySdpObserver() {
override fun onCreateSuccess(desc: SessionDescription?) {
// Set the local description with the created SDP offer
peerConnection?.setLocalDescription(object : MySdpObserver() {
override fun onSetSuccess() {
// Handle success
}
}, desc)
}
}, MediaConstraints().apply {
// Set media constraints for video and audio
mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveVideo", "true"))
mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"))
})
}
/**
* Creates an SDP answer in response to an offer from the target user.
*
* @param target The target user.
*/
fun answer(target: String) {
peerConnection?.createAnswer(object : MySdpObserver() {
override fun onCreateSuccess(desc: SessionDescription?) {
// Set the local description with the created SDP answer
peerConnection?.setLocalDescription(object : MySdpObserver() {
override fun onSetSuccess() {
// Handle success
}
}, desc)
}
}, MediaConstraints().apply {
// Set media constraints for video and audio
mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveVideo", "true"))
mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"))
})
}
/**
* Receives and sets the remote session description.
*
* @param sessionDescription The remote session description.
*/
fun onRemoteSessionReceived(sessionDescription: SessionDescription) {
peerConnection?.setRemoteDescription(MySdpObserver(), sessionDescription)
}
/**
* Adds an ICE candidate to the peer connection.
*
* @param iceCandidate The ICE candidate to be added.
*/
fun addIceCandidateToPeer(iceCandidate: IceCandidate) {
peerConnection?.addIceCandidate(iceCandidate)
}
/**
* Sends an ICE candidate to the target user.
*
* @param target The target user.
* @param iceCandidate The ICE candidate to be sent.
*/
fun sendIceCandidate(target: String, iceCandidate: IceCandidate) {
addIceCandidateToPeer(iceCandidate)
}
/**
* Closes the peer connection and releases resources.
*/
fun closeConnection() {
localStream?.dispose()
peerConnection?.close()
}
/**
* Starts local media streaming by creating a local media stream and adding video and audio tracks.
*/
private fun startLocalStreaming() {
localStream = peerConnectionFactory.createLocalMediaStream("localStream")
localAudioTrack = peerConnectionFactory.createAudioTrack("localAudio", localAudioSource)
localStream?.addTrack(localAudioTrack)
val surfaceTextureHelper = SurfaceTextureHelper.create(Thread.currentThread().name, EglBase.create().eglBaseContext)
videoCapturer.initialize(surfaceTextureHelper, context, localVideoSource.capturerObserver)
videoCapturer.startCapture(720, 480, 30)
localVideoTrack = peerConnectionFactory.createVideoTrack("localVideo", localVideoSource)
localStream?.addTrack(localVideoTrack)
}
/**
* Retrieves a video capturer suitable for the platform.
*
* @param context The application context.
* @return The video capturer.
*/
private fun getVideoCapturer(context: Context): VideoCapturer {
// Implementation to retrieve a video capturer (e.g., using Camera1 or Camera2 API)
// For simplicity, this is left as a placeholder
throw NotImplementedError("Video capturer retrieval is not implemented.")
}
}总结
本文介绍了使用 Kotlin、WebRTC 和 Firebase 构建视频通话应用程序的过程，重点是模块化和可维护的代码设计。通过实现 MainService、MainRepository和NSWebRTCClient类，您可以高效地管理视频通话功能和交互。所提供的代码示例和注释旨在阐明应用程序各部分的目的和功能，便于更好地理解和更轻松地实施强大的视频通话解决方案。
作者：Enes Algan
本文来自作者投稿，版权归原作者所有。如需转载，请注明出处：https://www.nxrte.com/jishu/webrtc/51400.html
