使用Python為Android平台開發WebRTC實時通信應用

WebRTC是一種開源的實時通信技術，可以在瀏覽器和移動設備上實現音頻、視頻和數據的實時通信功能。本文介紹如何使用Python為Android平台開發WebRTC實時通信應用，並提供完整的代碼示例。

一、環境搭建

在開始開發之前，需要安裝以下軟體和庫：

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

二、Android Studio項目創建

在Android Studio中創建一個新的項目，選擇「Empty Activity」。在項目目錄下創建一個名為「jni」的文件夾，用於存放C/C++代碼。

三、NDK配置

在項目的build.gradle文件中，添加如下代碼：

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

在app/src/main下創建一個新文件夾jniLibs，並將WebRTC Native庫拷貝到該文件夾下。在CMakeLists.txt文件中添加以下代碼：

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腳本編寫

在Python腳本中實現WebRTC的信令交換和媒體協商功能。以下是一個簡單的示例：

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、應用開發

在Android Studio中創建一個名為「WebRTCActivity」的Activity，並在布局文件中添加一個名為「remote_video_view」的SurfaceView，用於顯示遠端視頻。在Activity中實現WebRTC的媒體協商功能，以及處理遠端音視頻流。以下是一個簡單的示例：

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、總結

本文介紹了如何使用Python為Android平台開發WebRTC實時通信應用。通過NDK配置和Python腳本編寫，實現了WebRTC的信令交換和媒體協商功能，並在Android端完成了音視頻流處理和顯示。開發者可以基於本文提供的代碼示例進行二次開發，實現更加豐富的實時通信應用。