基于SRS服务器实现Android-Web端视频通话（3）：Android端向SRS服务器推送WebRTC流

基于SRS服务器实现Android-Web端视频通话（1）：SRS服务器启用HTTPS
基于SRS服务器实现Android-Web端视频通话（2）：Android端从SRS服务器拉取WebRTC流
基于SRS服务器实现Android-Web端视频通话（3）：Android端向SRS服务器推送WebRTC流

实现效果

引库

implementation 'org.webrtc:google-webrtc:1.0.32006'

其他版本，详见

推流流程

createPeerConnectionFactory -> createPeerConnection(addTransceiver) -> createOffer -> setLocalDescription(OFFER) -> get remote sdp(network requset) -> setRemoteDescription(ANSWER)

代码实现

初始化

//加载并初始化 WebRTC，在创建 PeerConnectionFactory 之前必须至少调用一次
PeerConnectionFactory.initialize(
    PeerConnectionFactory.InitializationOptions
        .builder(applicationContext).createInitializationOptions()
)

private val eglBaseContext = EglBase.create().eglBaseContext

createPeerConnectionFactory

private val peerConnectionFactory: PeerConnectionFactory = createPeerConnectionFactory()

private fun createPeerConnectionFactory(): PeerConnectionFactory {
   
	//先做默认配置，后面可能会遇到坑
    val options = PeerConnectionFactory.Options()
    val encoderFactory = DefaultVideoEncoderFactory(eglBaseContext, true, true)
    val decoderFactory = DefaultVideoDecoderFactory(eglBaseContext)
    return PeerConnectionFactory.builder()
        .setOptions(options)
        .setVideoEncoderFactory(encoderFactory)
        .setVideoDecoderFactory(decoderFactory)
        .createPeerConnectionFactory()
}

createPeerConnection(addTransceiver)

private fun initPeerConnection() {
   
    val createAudioSource = peerConnectionFactory.createAudioSource(createAudioConstraints())
    val audioTrack =
        peerConnectionFactory.createAudioTrack("local_audio_track", createAudioSource)

    cameraVideoCapturer = createVideoCapture(this)
    cameraVideoCapturer?.let {
    capture ->
        val videoSource = peerConnectionFactory.createVideoSource(capture.isScreencast)
        videoTrack =
            peerConnectionFactory.createVideoTrack("local_video_track", videoSource).apply {
   
            	//显示到本地画面上
                addSink(mBinding.svr)
            }
        surfaceTextureHelper =
            SurfaceTextureHelper.create("surface_texture_thread", eglBaseContext)
        capture.initialize(surfaceTextureHelper, this, videoSource.capturerObserver)
        //开始捕帧，宽、高、帧率。
        capture.startCapture(640, 480, 20)
    }

    val rtcConfig = PeerConnection.RTCConfiguration(emptyList())
    /*
 	<p>For users who wish to send multiple audio/video streams and need to stay interoperable with legacy WebRTC implementations, specify PLAN_B.
 	<p>For users who wish to send multiple audio/video streams and/or wish to use the new RtpTransceiver API, specify UNIFIED_PLAN.
 	*/
	//使用PeerConnection.SdpSemantics.UNIFIED_PLAN
    rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN

    peerConnection = peerConnectionFactory.createPeerConnection(
        rtcConfig,
        PeerConnectionObserver