Android webrtc实战(一)录制本地视频并播放,附带详细的基础知识讲解

目录

一、创建PeerConnectionFactory

初始化

构建对象

二、创建AudioDeviceModule

AudioDeviceModule

JavaAudioDeviceModule

构建对象

setAudioAttributes

setAudioFormat

setAudioSource

创建录制视频相关对象

创建VideoSource

创建VideoCapturer

创建VideoTrack

播放视频

切换前后置摄像头

别忘了申请权限

完整代码


本系列文章带大家熟悉webrtc,最终用webrtc做一个p2p音视频通话的app。本文章作为本系列第一期主要讲解一些基础知识,同时实现用webrtc播放本地录制的视频的功能。文章最后会提供完整的代码。如果有小伙伴还没有Android webrtc源码,可以关注我并私信“Android webrtc源码”,我会给大家提供源码,获取源码后以module的形式导入到自己的项目即可。

一、创建PeerConnectionFactory

PeerConnectionFactory是一个创建、配置和管理其余一切的类,是使用webrtc的起点。

初始化

        PeerConnectionFactory.initialize(InitializationOptions.builder(context)
//            .setFieldTrials("xiongFieldTrials")// 设置实验性功能
//            .setNativeLibraryName("jingle_peerconnection_so")// 底层库的名称,可以不用设置。如果设置名称一定要和底层库的名称一致
            .setEnableInternalTracer(true)// 启用内部追踪器,用来记录一些相关数据
            .createInitializationOptions())

在使用webrtc之前至少要调用一次该方法,主要目的是初始化并加载webrtc。

构建对象

peerConnectionFactory = PeerConnectionFactory.builder()
            .setVideoDecoderFactory(DefaultVideoDecoderFactory(eglContext))// 设置视频解码工厂
            .setVideoEncoderFactory(DefaultVideoEncoderFactory(eglContext, false, true))//设置视频编码工厂
            .setAudioDeviceModule(adm)
            .setOptions(options)
            .createPeerConnectionFactory()

在构建对象时主要设置了编解码工厂、音频管理设备和一些额外选项,其中编解码知识不在本文章讲解范围内,音频管理设备在本文章后面进行讲解。除了这些,这个setOptions是做什么的呢?在setOptions时需要传入一个Options对象,这个对象有三个公开的属性:

    public int networkIgnoreMask;
    public boolean disableEncryption;
    public boolean disableNetworkMonitor;

其中,networkIgnoreMask:用来忽略指定的网络类型,也就是说不会使用这个网络进行webrtc通信,可取以下这些值:

  • ADAPTER_TYPE_UNKNOWN = 0:未知类型的以太网适配器。
  • ADAPTER_TYPE_ETHERNET = 1 << 0:以太网适配器。
  • ADAPTER_TYPE_WIFI = 1 << 1:Wi-Fi 适配器。
  • ADAPTER_TYPE_CELLULAR = 1 << 2:蜂窝移动数据适配器。
  • ADAPTER_TYPE_VPN = 1 << 3:VPN 适配器。
  • ADAPTER_TYPE_LOOPBACK = 1 << 4:回环适配器。
  • ADAPTER_TYPE_ANY = 1 << 5:任何适配器类型都不被忽略。这是默认值。

disableEncryption:true表示不用数据加密

disableNetworkMonitor:true表示禁用网络监视器

二、创建AudioDeviceModule

AudioDeviceModule

AudioDeviceModule是webrtc用来管理和配置音频的,在上边创建peerconnectionFactory时把该对象作为setAudioDeviceModule方法的参数传了进去。AudioDeviceModule是一个接口类,其中有四个方法:

/**
   * Returns a C++ pointer to a webrtc::AudioDeviceModule. Caller does _not_ take ownership and
   * lifetime is handled through the release() call.
   */
  long getNativeAudioDeviceModulePointer();

  /**
   * Release resources for this AudioDeviceModule, including native resources. The object should not
   * be used after this call.
   */
  void release();

  /** Control muting/unmuting the speaker. */
  void setSpeakerMute(boolean mute);

  /** Control muting/unmuting the microphone. */
  void setMicrophoneMute(boolean mute);
  • getNativeAudioDeviceModulePointer:返回顶层c++ webrtc::AudioDeviceModule的指针
  • release: 释放资源,释放后不能再使用这个对象了
  • setSpeakerMute:扬声器开启或者关闭静音
  • setMicrophoneMute:麦克风开启或者关闭静音

JavaAudioDeviceModule

JavaAudioDeviceModule是AudioDeviceModule的实现类,里面封装了一个WebRtcAudioRecord对象作为音频输入、一个WebRtcAudioTrack对象作为音频输出,还封装了一些设置采样率、音频格式、声道数等的配置。其中各种关于音频类的关系以及WebRtcAudioRecord我在另外一篇文章中有比较详细的介绍:webrtc怎么播放本地音频文件。

构建对象

        adm = JavaAudioDeviceModule.builder(context)
            .setAudioAttributes(audioAttributes)// 设置音频属性
            .setAudioFormat(AudioFormat.ENCODING_PCM_16BIT)// 设置音频采样格式
            .setAudioSource(MediaRecorder.AudioSource.VOICE_COMMUNICATION)// 设置音频录制源
            .setSampleRate(44100)// 设置音频采样率
            .setAudioRecordErrorCallback(audioRecordErrorCallback)// audio record错误记录
            .setAudioRecordStateCallback(audioRecordStateCallback)// audio record状态回调
            .setAudioTrackErrorCallback(audioTrackErrorCallback)// audio track错误回调
            .setAudioTrackStateCallback(audioTrackStateCallback)// audio track状态回调
            .setSamplesReadyCallback(samplesReadyCallback)// 每成功发送一次数据就调用该对象中的onWebRtcAudioRecordSamplesReady方法
            .setUseHardwareAcousticEchoCanceler(true)// 使用硬件回声消除
            .setUseHardwareNoiseSuppressor(true)// 使用硬件噪声抑制
            .setUseStereoInput(false)// 使用立体声输入
            .setUseStereoOutput(false)// 使用立体声输出
            .createAudioDeviceModule()

adm就是AudioDeviceModule对象,我们来看一下builder中的各种方法:

setAudioAttributes

此方法需要传递一个AudioAttributes对象,最后到WebrtcAudioTrack的构造方法中。AudioAttributes是用来描述音频流的信息的。可以通过以下方法创建该对象:

        val audioAttributes = AudioAttributes.Builder()
            .setFlags(AudioAttributes.FLAG_AUDIBILITY_ENFORCED)// 设置标志,加强可听性
            .setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)// 设置音频类型,会影响输出
            .setUsage(AudioAttributes.USAGE_VOICE_COMMUNICATION)// 设置使用场景
            .build()

setUsage设置使用场景,可用的取值有:

  • USAGE_MEDIA:用于媒体播放

  • USAGE_VOICE_COMMUNICATION:用于语音通信

  • USAGE_ALARM:用于闹钟提醒

  • USAGE_NOTIFICATION:用于通知提醒

  • USAGE_NOTIFICATION_RINGTONE:用于设置通知铃声

  • USAGE_NOTIFICATION_COMMUNICATION_REQUEST:用于请求通话或使用其它形式的通信

  • USAGE_NOTIFICATION_COMMUNICATION_INSTANT:用于即时通信

  • USAGE_NOTIFICATION_COMMUNICATION_DELAYED:用于延迟发送的即时通信

  • USAGE_ASSISTANCE_ACCESSIBILITY:用于辅助功能,如语音助手和屏幕阅读器。

本篇文章的重点不在这里,想要深入了解的小伙伴可自行去了解,其它的就不展开讲了。

setAudioFormat

设置音频采样格式,ENCODING_PCM_16BIT意味着每个采样点点的位深为16bit,单个采样点位深越大,音频质量越好。

setAudioSource

设置音频源,取值必须来自android.media.MediaRecorder.AudioSource中定义的值,可用取值为:

  • MediaRecorder.AudioSource.MIC:麦克风录音

  • MediaRecorder.AudioSource.CAMCORDER:摄像头录音

  • MediaRecorder.AudioSource.DEFAULT:默认音频源

  • MediaRecorder.AudioSource.VOICE_RECOGNITION:语音识别

  • MediaRecorder.AudioSource.VOICE_COMMUNICATION:网络电话等实时通信场景

  • MediaRecorder.AudioSource.REMOTE_SUBMIX:捕获远程混音声音的输出流(需要 API level 19 及以上)

  • MediaRecorder.AudioSource.VOICE_DOWNLINK: 用于从电话系统中录制接收到的下行语音的源

  • MediaRecorder.AudioSource.VOICE_UPLINK: 用于从电话系统中录制发送到上行语音的源

创建录制视频相关对象

创建VideoSource

videoSource = peerConnectionFactory.createVideoSource(false)// 参数isScreentcase表示是否是屏幕录制

VideoSource对象是连接VideoCapturer和VideoTrack的桥梁。

创建VideoCapturer

VideoCapturer是所有录制视频的类必需要实现的接口,其中有六个方法:

  /**
   * This function is used to initialize the camera thread, the android application context, and the
   * capture observer. It will be called only once and before any startCapture() request. The
   * camera thread is guaranteed to be valid until dispose() is called. If the VideoCapturer wants
   * to deliver texture frames, it should do this by rendering on the SurfaceTexture in
   * {@code surfaceTextureHelper}, register itself as a listener, and forward the frames to
   * CapturerObserver.onFrameCaptured(). The caller still has ownership of {@code
   * surfaceTextureHelper} and is responsible for making sure surfaceTextureHelper.dispose() is
   * called. This also means that the caller can reuse the SurfaceTextureHelper to initialize a new
   * VideoCapturer once the previous VideoCapturer has been disposed.
   */
  void initialize(SurfaceTextureHelper surfaceTextureHelper, Context applicationContext,
      CapturerObserver capturerObserver);

  /**
   * Start capturing frames in a format that is as close as possible to {@code width x height} and
   * {@code framerate}.
   */
  void startCapture(int width, int height, int framerate);

  /**
   * Stop capturing. This function should block until capture is actually stopped.
   */
  void stopCapture() throws InterruptedException;

  void changeCaptureFormat(int width, int height, int framerate);

  /**
   * Perform any final cleanup here. No more capturing will be done after this call.
   */
  void dispose();

  /**
   * @return true if-and-only-if this is a screen capturer.
   */
  boolean isScreencast();
  • initialize:初始化,在开始录制前必需要调用此方法
  • startCapture:开始录制
  • stopCapture:停止录制
  • changeCaptureFormat:更改参数,包括视频宽高和帧率
  • dispose:释放资源
  • isScreencast:是否是屏幕录制

ScreenCapturerAndroid、FileVideoCapturer和CameraCapturer是VideoCapturer的实现类,此处我们用到的是CameraCapturer,即用摄像头来录制视频。说到CameraCapturer,就有必要了解下CameraEnumerator。CameraEnumerator用来获取设备摄像头,并且创建CameraCapturer。

        var cameraEnumerator:CameraEnumerator// CameraEnumerator是枚举本地设备的类,用来创建cameraCapturer
        if (Camera2Enumerator.isSupported(context)) {
            cameraEnumerator = Camera2Enumerator(context)
        } else {
            cameraEnumerator = Camera1Enumerator()
        }
        for (name in cameraEnumerator.deviceNames) {
            if(cameraEnumerator.isFrontFacing(name)) frontDeviceName = name//前置摄像头
            if (cameraEnumerator.isBackFacing(name)) backDeviceName = name//后置摄像头
        }
        videoCapturer = cameraEnumerator.createCapturer(frontDeviceName!!, cameraEventsHandler)

创建VideoTrack

在创建VideoTrack之前首先要开始录制。

        videoCapturer.initialize(surfaceTextureHelper, context, videoSource.capturerObserver)
        videoCapturer.startCapture(HD_VIDEO_WIDTH, HD_VIDEO_HEIGHT, FRAME_RATE)
        videoTrack = peerConnectionFactory.createVideoTrack("xiong video track", videoSource)

播放视频

webrtc播放视频的控件是SurfaceViewRenderer,其继承了SurfaceView,真正实现视频渲染的是EglRenderer类。SurfaceViewRenderer必需要先调用init方法初始化。

        surfaceViewRenderer.init(eglContext, null)
        surfaceViewRenderer.setScalingType(RendererCommon.ScalingType.SCALE_ASPECT_FIT)//视频缩放类型
        surfaceViewRenderer.setMirror(isLocal)//是否启用镜像

调用videoTrack的addSink方法即可播放。

        videoTrack.addSink(localRenderer)

切换前后置摄像头

cameraVideoCapturer.switchCamera(cameraSwitchHandler, deviceName)

deviceName为需要切换的摄像头名称,上边在创建VideoCapturer时我们获取到了前后置摄像头的名称,传到这里即可。

别忘了申请权限

使用webrtc需要申请相关的权限,本次我们只需要申请camera和audio_record两个权限。

首先在AndroidManifest.xml文件中添加如下代码:

    
    

然后在运行时动态申请:

        if (ContextCompat.checkSelfPermission(this, Manifest.permission.CAMERA) != PackageManager.PERMISSION_GRANTED ||
        ContextCompat.checkSelfPermission(this, Manifest.permission.RECORD_AUDIO) != PackageManager.PERMISSION_GRANTED) {
            ActivityCompat.requestPermissions(this, new String[] {Manifest.permission.CAMERA, Manifest.permission.RECORD_AUDIO}, 0x01);
        }

完整代码

class MediaClient(context: Context) {
    private lateinit var peerConnectionFactory: PeerConnectionFactory
    private lateinit var videoSource: VideoSource
    private lateinit var videoTrack: VideoTrack
    private lateinit var videoCapturer: VideoCapturer
    private lateinit var audioSource: AudioSource
    private lateinit var audioTrack: AudioTrack
    private lateinit var adm: AudioDeviceModule
    private var eglContext: EglBase.Context = EglBase.create().eglBaseContext
    private var context = context.applicationContext//防止内存泄露
    private var frontDeviceName: String? = null// 前置摄像头name
    private var backDeviceName: String? = null// 后置摄像头name
    private val HD_VIDEO_WIDTH = 320//视频宽
    private val HD_VIDEO_HEIGHT = 240//视频高
    private val FRAME_RATE = 30//视频帧率
    private var isInitialized = false// 是否被初始化,如果没有,使用之前首先要初始化
    private var isFront = true// 是否是前置摄像头,用来切换前后摄像头
    companion object {
        private val TAG = "MediaClientLog"
    }

    fun initMediaClient() {
        isInitialized = true;
        PeerConnectionFactory.initialize(InitializationOptions.builder(context)
//            .setFieldTrials("xiongFieldTrials")// 设置实验性功能
//            .setNativeLibraryName("jingle_peerconnection_so")// 底层库的名称,可以不用设置。如果设置名称一定要和底层库的名称一致
            .setEnableInternalTracer(true)// 启用内部追踪器,用来记录一些相关数据
            .createInitializationOptions())
        createAdm()//创建adm
        val options = PeerConnectionFactory.Options()
        options.networkIgnoreMask = 0x8// 需要忽略的网络类型,忽略了指定的网络后不会尝试连接该类型网络,此处0x8表示
        options.disableEncryption = true// 不用数据加密
        options.disableNetworkMonitor = false// 启用网络监视器
        peerConnectionFactory = PeerConnectionFactory.builder()
            .setVideoDecoderFactory(DefaultVideoDecoderFactory(eglContext))// 设置视频解码工厂
            .setVideoEncoderFactory(DefaultVideoEncoderFactory(eglContext, false, true))//设置视频编码工厂
            .setAudioDeviceModule(adm)
            .setOptions(options)
            .createPeerConnectionFactory()
        initAVResource()//初始化相关资源或对象
    }

    private fun createAdm() {
        val audioAttributes = AudioAttributes.Builder()
            .setFlags(AudioAttributes.FLAG_AUDIBILITY_ENFORCED)// 设置标志,加强可听性
            .setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)// 设置音频类型,会影响输出
            .setUsage(AudioAttributes.USAGE_VOICE_COMMUNICATION)// 设置使用场景
            .build()
        val audioRecordErrorCallback = object : AudioRecordErrorCallback {
            override fun onWebRtcAudioRecordInitError(errorMessage: String?) {
                Log.e(TAG, "onWebRtcAudioRecordInitError: ")
            }

            override fun onWebRtcAudioRecordStartError(
                errorCode: JavaAudioDeviceModule.AudioRecordStartErrorCode?,
                errorMessage: String?
            ) {
                Log.e(TAG, "onWebRtcAudioRecordStartError: ")
            }

            override fun onWebRtcAudioRecordError(errorMessage: String?) {
                Log.e(TAG, "onWebRtcAudioRecordError: ")
            }

        }
        val audioRecordStateCallback = object : AudioRecordStateCallback {
            override fun onWebRtcAudioRecordStart() {
                Log.d(TAG, "onWebRtcAudioRecordStart: ")
            }

            override fun onWebRtcAudioRecordStop() {
                Log.d(TAG, "onWebRtcAudioRecordStop: ")
            }

        }
        val audioTrackErrorCallback = object : AudioTrackErrorCallback {
            override fun onWebRtcAudioTrackInitError(errorMessage: String?) {
                Log.e(TAG, "onWebRtcAudioTrackInitError: ", )
            }

            override fun onWebRtcAudioTrackStartError(
                errorCode: JavaAudioDeviceModule.AudioTrackStartErrorCode?,
                errorMessage: String?
            ) {
                Log.e(TAG, "onWebRtcAudioTrackStartError: ", )
            }

            override fun onWebRtcAudioTrackError(errorMessage: String?) {
                Log.e(TAG, "onWebRtcAudioTrackError: ", )
            }

        }
        val audioTrackStateCallback = object : AudioTrackStateCallback {
            override fun onWebRtcAudioTrackStart() {
                Log.d(TAG, "onWebRtcAudioTrackStart: ")
            }

            override fun onWebRtcAudioTrackStop() {
                Log.d(TAG, "onWebRtcAudioTrackStop: ")
            }

        }
        val samplesReadyCallback = object : SamplesReadyCallback {
            override fun onWebRtcAudioRecordSamplesReady(samples: JavaAudioDeviceModule.AudioSamples?) {
                Log.d(TAG, "onWebRtcAudioRecordSamplesReady: ")
            }
        }
        adm = JavaAudioDeviceModule.builder(context)
            .setAudioAttributes(audioAttributes)// 设置音频属性
            .setAudioFormat(AudioFormat.ENCODING_PCM_16BIT)// 设置音频采样格式
            .setAudioSource(MediaRecorder.AudioSource.VOICE_COMMUNICATION)// 设置音频录制源
            .setSampleRate(44100)// 设置音频采样率
            .setAudioRecordErrorCallback(audioRecordErrorCallback)// audio record错误记录
            .setAudioRecordStateCallback(audioRecordStateCallback)// audio record状态回调
            .setAudioTrackErrorCallback(audioTrackErrorCallback)// audio track错误回调
            .setAudioTrackStateCallback(audioTrackStateCallback)// audio track状态回调
            .setSamplesReadyCallback(samplesReadyCallback)// 每成功发送一次数据就调用该对象中的onWebRtcAudioRecordSamplesReady方法
            .setUseHardwareAcousticEchoCanceler(true)// 使用硬件回声消除
            .setUseHardwareNoiseSuppressor(true)// 使用硬件噪声抑制
            .setUseStereoInput(false)// 使用立体声输入
            .setUseStereoOutput(false)// 使用立体声输出
            .createAudioDeviceModule()
        adm.release()
    }

    private fun initAVResource() {
        val audioConstraints = MediaConstraints()
        audioConstraints.mandatory.add(MediaConstraints.KeyValuePair("googEchoCancellation", "true"))// 启用回声消除
        audioConstraints.mandatory.add(MediaConstraints.KeyValuePair("googAutoGainControl", "true"))// 自动增益
        audioConstraints.mandatory.add(MediaConstraints.KeyValuePair("googNoiseSuppression", "true"))// 降噪
        audioConstraints.mandatory.add(MediaConstraints.KeyValuePair("googHighpassFilter", "true"))// 高通滤波
        audioSource = peerConnectionFactory.createAudioSource(audioConstraints)
        audioTrack = peerConnectionFactory.createAudioTrack("xiong audio track", audioSource)
        var cameraEnumerator:CameraEnumerator// CameraEnumerator是枚举本地设备的类,用来创建cameraCapturer
        if (Camera2Enumerator.isSupported(context)) {
            cameraEnumerator = Camera2Enumerator(context)
        } else {
            cameraEnumerator = Camera1Enumerator()
        }
        for (name in cameraEnumerator.deviceNames) {
            if(cameraEnumerator.isFrontFacing(name)) frontDeviceName = name//前置摄像头
            if (cameraEnumerator.isBackFacing(name)) backDeviceName = name//后置摄像头
        }
        val cameraEventsHandler = object : CameraEventsHandler {
            override fun onCameraError(errorDescription: String?) {
                Log.e(TAG, "onCameraError: ", )
            }

            override fun onCameraDisconnected() {
                Log.e(TAG, "onCameraDisconnected: ", )
            }

            override fun onCameraFreezed(errorDescription: String?) {
                Log.e(TAG, "onCameraFreezed: ", )
            }

            override fun onCameraOpening(cameraName: String?) {
                Log.d(TAG, "onCameraOpening: ")
            }

            override fun onFirstFrameAvailable() {
                Log.d(TAG, "onFirstFrameAvailable: ")
            }

            override fun onCameraClosed() {
                Log.d(TAG, "onCameraClosed: ")
            }

        }//camera事件捕捉
        videoCapturer = cameraEnumerator.createCapturer(frontDeviceName!!, cameraEventsHandler)
        videoSource = peerConnectionFactory.createVideoSource(false)// 参数isScreentcase表示是否是屏幕录制
        val surfaceTextureHelper = SurfaceTextureHelper.create("surface thread", eglContext)//用来创建videoframe
        videoCapturer.initialize(surfaceTextureHelper, context, videoSource.capturerObserver)
        videoCapturer.startCapture(HD_VIDEO_WIDTH, HD_VIDEO_HEIGHT, FRAME_RATE)
        videoTrack = peerConnectionFactory.createVideoTrack("xiong video track", videoSource)
    }

    fun loadVideo(localRenderer: SurfaceViewRenderer) {
        if (!isInitialized) initMediaClient()
        videoTrack.addSink(localRenderer)
    }

    fun initRenderer(surfaceViewRenderer: SurfaceViewRenderer, isLocal: Boolean) {
        surfaceViewRenderer.init(eglContext, null)
        surfaceViewRenderer.setScalingType(RendererCommon.ScalingType.SCALE_ASPECT_FIT)//视频缩放类型
        surfaceViewRenderer.setMirror(isLocal)//是否启用镜像
    }

    fun switchCamera() {
        if (!isInitialized) initMediaClient()
        val cameraVideoCapturer = videoCapturer as CameraVideoCapturer
        val cameraSwitchHandler = object : CameraSwitchHandler {
            override fun onCameraSwitchDone(isFrontCamera: Boolean) {
                Log.d(TAG, "onCameraSwitchDone: ")
            }

            override fun onCameraSwitchError(errorDescription: String?) {
                Log.d(TAG, "onCameraSwitchError: ")
            }

        }
        if (isFront) {
            cameraVideoCapturer.switchCamera(cameraSwitchHandler, backDeviceName!!)
            isFront = false
        }
        else {
            cameraVideoCapturer.switchCamera(cameraSwitchHandler, frontDeviceName)
            isFront = true
        }
    }
}

在自己的activity中先调用initMediaClient方法,然后调用initRenderer方法,最后调用loadVideo方法即可。

最终实现效果如下:

Android webrtc实战(一)录制本地视频并播放,附带详细的基础知识讲解_第1张图片

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