Android端WebRTC本地音视频采集流程源码分析

WebRTC源码版本为：org.webrtc:google-webrtc:1.0.32006
本文仅分析Java层源码，在分析之前，先说明一下一些重要类的基本概念。

• MediaSource：WebRTC媒体资源数据来源，它有两个子类：AudioSource（音频资源）、VideoSource（视频资源）；
• MediaStreamTrack：媒体资源轨，一个MediaStreamTrack对应一个MediaSource，创建媒体轨需要MediaSource，同样，它也有两个子类：AudioTrack（音频轨）对应AudioSource、VideoTrack（视频轨）对应VideoSource；
• MediaStream：媒体流，一个媒体流可以添加多条AudioTrack和VideoTrack，一般来说我们对应分别只添加一条。

在使用WebRTC进行音视频通话时，需要构建一个PeerConnectionFactory，这是一个连接工厂，创建本地LocalMediaStream流及客户端PeerConnection等，都需要使用。构建大致代码如下：

//在创建PeerConnectionFactory之前，必须至少调用一次。不得在 PeerConnectionFactory 处于活动状态时调用。
PeerConnectionFactory.initialize(
    PeerConnectionFactory.InitializationOptions
        .builder(applicationContext).createInitializationOptions()
)
val eglBaseContext = EglBase.create().eglBaseContext
//视频编码工厂
val encoderFactory= DefaultVideoEncoderFactory(eglBaseContext, true, true)
//视频解码工厂
val decoderFactory = DefaultVideoDecoderFactory(eglBaseContext)
val audioDeviceModule = JavaAudioDeviceModule.builder(this)
            .setSamplesReadyCallback {
    audioSamples ->
                 //麦克风输入数据，即通话时LocalMediaStream的Audio的数据，pcm格式，通常用于录音。

            }
            .createAudioDeviceModule()
val peerConnectionFactory = PeerConnectionFactory.builder()
            .setVideoEncoderFactory(encoderFactory)
            .setVideoDecoderFactory(decoderFactory)
            .setAudioDeviceModule(audioDeviceModule)
            .createPeerConnectionFactory()

若你有需求WebRTC视频编码要启用H.264，请参考Android端WebRTC启用H264编码。

MediaSource

WebRTC媒体资源数据来源基类。

AudioSource（音频源）

通过 peerConnectionFactory.createAudioSource(MediaConstraints) 创建，参数为媒体约束，大致如下：

    //音频源
    val audioSource = peerConnectionFactory.createAudioSource(createAudioConstraints())
    private fun createAudioConstraints(): MediaConstraints {
   
        val audioConstraints = MediaConstraints()
        //回声消除
        audioConstraints.mandatory.add(
            MediaConstraints.KeyValuePair(
                "googEchoCancellation",
                "true"
            )
        )
        //自动增益
        audioConstraints.mandatory.add(MediaConstraints.KeyValuePair("googAutoGainControl", "true"))
        //高音过滤
        audioConstraints.mandatory.add(MediaConstraints.KeyValuePair("googHighpassFilter", "true"))
        //噪音处理
        audioConstraints.mandatory.add(
            MediaConstraints.KeyValuePair(
                "googNoiseSuppression",
                "true"
            )
        )
        return audioConstraints
    }

事实上音频的输入、输出具体处理在JavaAudioDeviceModule：

package org.webrtc.audio;
/**
 * AudioDeviceModule implemented using android.media.AudioRecord as input and
 * android.media.AudioTrack as output.
 */
public class JavaAudioDeviceModule implements AudioDeviceModule {
   
    ...
    /**
     * 本地麦克风采集的输入数据，android.media.AudioRecord
     */
    private final WebRtcAudioRecord audioInput;
    /**
     * 用于播放通话时对方的音频数据，android.media.AudioTrack
     */
    private final WebRtcAudioTrack audioOutput;
    ...
}

记录本地麦克风采集的音频数据WebRtcAudioRecord：

package org.webrtc.audio;

import android.media.AudioRecord;

class WebRtcAudioRecord {
   
    ...
    private AudioRecord audioRecord;
    /**
     * 读取线程
     */
    private AudioRecordThread audioThread;

  /**
   * Audio thread which keeps calling ByteBuffer.read() waiting for audio
   * to be recorded. Feeds recorded data to the native counterpart as a
   * periodic sequence of callbacks using DataIsRecorded().
   * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
   */
    private class AudioRecordThread extends Thread {
   
        private volatile boolean keepAlive = true;

        public AudioRecordThread(String name) {
   
            super(name);
        }

        public void run() {
   
            Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
             //确保在采集数据状态
            assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);
            //向外回调开始状态
            doAudioRecordStateCallback(AUDIO_RECORD_START);
            //存活状态
            while(keepAlive) {
   
                int bytesRead = audioRecord.read(byteBuffer, byteBuffer.capacity());
                if (bytesRead == byteBuffer.capacity()) {
   
                    if (microphoneMute) {
   
                        //如果麦克风静音，则清空数据
                        byteBuffer.clear();
                        byteBuffer.put(WebRtcAudioRecord.this.emptyBytes);
                    }

                    if (keepAlive) {
   
                        //调用native方法发送数据
                        nativeDataIsRecorded(nativeAudioRecord, bytesRead);
                    }
                    //向外回调音频数据，pcm格式
                    if (audioSamplesReadyCallback != null) {
   
                        byte[] data = Arrays.copyOfRange(.byteBuffer.array(), byteBuffer.arrayOffset(), byteBuffer.capacity() + byteBuffer.arrayOffset());
                        audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(new AudioSamples(audioRecord.getAudioFormat(), audioRecord.getChannelCount(), audioRecord.getSampleRate(), data));
                    }
                } else {
   
                    String errorMessage = "AudioRecord.read failed: " + bytesRead;
                    Logging.e("WebRtcAudioRecordExternal", errorMessage);
                    if (<