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。
