使用Python为Android平台开发WebRTC实时通信应用

WebRTC是一种开源的实时通信技术，可以在浏览器和移动设备上实现音频、视频和数据的实时通信功能。本文介绍如何使用Python为Android平台开发WebRTC实时通信应用，并提供完整的代码示例。

一、环境搭建

在开始开发之前，需要安装以下软件和库：

• Android Studio
• Python 3.x
• Android NDK
• WebRTC Native库

二、Android Studio项目创建

在Android Studio中创建一个新的项目，选择“Empty Activity”。在项目目录下创建一个名为“jni”的文件夹，用于存放C/C++代码。

三、NDK配置

在项目的build.gradle文件中，添加如下代码：

android {
    defaultConfig {
        ndk {
            moduleName "webrtc"
        }
    }
    externalNativeBuild {
        cmake {
            path "CMakeLists.txt"
        }
    }
}

在app/src/main下创建一个新文件夹jniLibs，并将WebRTC Native库拷贝到该文件夹下。在CMakeLists.txt文件中添加以下代码：

cmake_minimum_required(VERSION 3.4.1)

add_library(webrtc SHARED IMPORTED)
set_target_properties(webrtc PROPERTIES IMPORTED_LOCATION ${CMAKE_SOURCE_DIR}/src/main/jniLibs/${ANDROID_ABI}/libjingle_peerconnection_so.so)

add_library( # Sets the name of the library.
             webrtc_native

             # Sets the library as a shared library.
             SHARED

             # Provides a relative path to your source file(s).
             src/main/cpp/webrtc_native.cpp )

target_link_libraries( # Specifies the target library.
                       webrtc_native

                       # Links the target library to the log library
                       # included in the NDK.
                       webrtc
                       log )

四、Python脚本编写

在Python脚本中实现WebRTC的信令交换和媒体协商功能。以下是一个简单的示例：

import websocket

class Signaling:
    def __init__(self):
        self.ws = websocket.WebSocketApp(
            "wss://example.com/ws",
            on_message = self.on_message,
            on_error = self.on_error,
            on_close = self.on_close
        )
        self.ws.on_open = self.on_open
        self.offer = None

    def on_open(self, ws):
        self.ws.send("hello")

    def on_message(self, ws, message):
        if message == "offer":
            self.offer = "sdp"
            self.ws.send("answer")

    def on_error(self, ws, error):
        print(error)

    def on_close(self, ws):
        print("closed")

    def run(self):
        self.ws.run_forever()

if __name__ == "__main__":
    signaling = Signaling()
    signaling.run()

五、应用开发

在Android Studio中创建一个名为“WebRTCActivity”的Activity，并在布局文件中添加一个名为“remote_video_view”的SurfaceView，用于显示远端视频。在Activity中实现WebRTC的媒体协商功能，以及处理远端音视频流。以下是一个简单的示例：

import org.webrtc.*

class WebRTCActivity : AppCompatActivity(), PeerConnection.Observer {
    private lateinit var rootEglBase: EglBase
    private lateinit var peerConnection: PeerConnection
    private lateinit var videoTrack: VideoTrack
    private lateinit var audioTrack: AudioTrack
    private lateinit var videoRenderer: VideoRenderer

    private val signaling = Signaling()

    private val iceServers = listOf(
        PeerConnection.IceServer.builder("stun:stun.l.google.com:19302").createIceServer()
    )

    override fun onCreate(savedInstanceState: Bundle?) {
        super.onCreate(savedInstanceState)
        setContentView(R.layout.activity_webrtc)

        rootEglBase = EglBase.create()
        val surfaceView = findViewById(R.id.remote_video_view)
        surfaceView.init(rootEglBase.eglBaseContext, null)

        val mediaConstraints = MediaConstraints()
        mediaConstraints.mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveVideo", "true"))
        mediaConstraints.mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"))

        val peerConnectionFactory = PeerConnectionFactory.builder()
            .setVideoEncoderFactory(DefaultVideoEncoderFactory(rootEglBase.eglBaseContext, true, false))
            .setVideoDecoderFactory(DefaultVideoDecoderFactory(rootEglBase.eglBaseContext))
            .setAudioDeviceModule(JavaAudioDeviceModule.builder(applicationContext).createAudioDeviceModule())
            .createPeerConnectionFactory()

        peerConnection = peerConnectionFactory.createPeerConnection(iceServers, this)

        val localAudioTrack = peerConnectionFactory.createAudioTrack(
            "100",
            peerConnectionFactory.createAudioSource(MediaConstraints())
        )
        audioTrack = peerConnectionFactory.createAudioTrack(
            "101",
            peerConnectionFactory.createAudioSource(MediaConstraints())
        )
        val localVideoSource = peerConnectionFactory.createVideoSource(false)
        val localVideoTrack = peerConnectionFactory.createVideoTrack(
            "102",
            localVideoSource
        )
        val videoCapturer = createCameraCapturer()
        videoCapturer?.let {
            localVideoSource.adaptOutputFormat(it.videoFormat.width, it.videoFormat.height, 60)
            localVideoTrack.addSink(surfaceView)
            it.startCapture(
                it.videoFormat.width,
                it.videoFormat.height,
                60
            )
        }

        peerConnection.addTrack(localAudioTrack)
        peerConnection.addTrack(localVideoTrack)

        signaling.offer = peerConnection.createOffer(mediaConstraints, object : SdpObserver {
            override fun onCreateSuccess(sdp: SessionDescription?) {
                sdp?.let {
                    peerConnection.setLocalDescription(object : SdpObserver {
                        override fun onCreateSuccess(sdp: SessionDescription?) {
                            signaling.sendOffer(sdp)
                        }

                        override fun onSetSuccess() {}
                        override fun onCreateFailure(p0: String?) {}
                        override fun onSetFailure(p0: String?) {}
                    }, it)
                }
            }

            override fun onSetSuccess() {}
            override fun onCreateFailure(p0: String?) {}
            override fun onSetFailure(p0: String?) {}
        })
    }

    override fun onDestroy() {
        super.onDestroy()
        rootEglBase.release()
        peerConnection.close()
    }

    override fun onIceCandidate(p0: IceCandidate?) {}
    override fun onAddStream(p0: MediaStream?) {
        p0?.let { stream ->
            videoTrack = stream.videoTracks[0]
            val mediaStreamVideoTrackRenderer = MediaStreamVideoTrackRenderer(videoTrack, rootEglBase.surfaceTextureHelper)
            videoRenderer = VideoRenderer(mediaStreamVideoTrackRenderer)
            videoTrack.addRenderer(videoRenderer)
        }
    }
    override fun onDataChannel(p0: DataChannel?) {}
    override fun onIceConnectionReceivingChange(p0: Boolean) {}
    override fun onIceConnectionChange(p0: PeerConnection.IceConnectionState?) {}
    override fun onIceGatheringChange(p0: PeerConnection.IceGatheringState?) {}
    override fun onRemoveStream(p0: MediaStream?) {}
    override fun onSignalingChange(p0: PeerConnection.SignalingState?) {}
    override fun onIceCandidatesRemoved(p0: Array?) {}
    override fun onRenegotiationNeeded() {}
    override fun onAddTrack(p0: RtpReceiver?, p1: Array?) {}

    private fun createCameraCapturer(): CameraVideoCapturer? {
        val camera2Enumerator = Camera2Enumerator(this)
        val deviceNames = camera2Enumerator.deviceNames
        for (deviceName in deviceNames) {
            if (camera2Enumerator.isFrontFacing(deviceName)) {
                val cameraCapturer = camera2Enumerator.createCapturer(deviceName, null)
                if (cameraCapturer != null) {
                    return cameraCapturer
                }
            }
        }
        return null
    }
}

六、总结

本文介绍了如何使用Python为Android平台开发WebRTC实时通信应用。通过NDK配置和Python脚本编写，实现了WebRTC的信令交换和媒体协商功能，并在Android端完成了音视频流处理和显示。开发者可以基于本文提供的代码示例进行二次开发，实现更加丰富的实时通信应用。