AudioRecord 是 Android 基于原始PCM音频数据录制的类，WebRCT 对其封装的代码位置位于
org/webrtc/audio/WebRtcAudioRecord.JAVA，接下来我们学习一下 AudioRecord 是如何创建启动，读取音频采集数据以及销毁等功能的。
private int initRecording(int sampleRate, int channels) {
Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")");
if (audioRecord != null) {
reportWebRtcAudioRecordInitError("InitRecording called twice without StopRecording.");
return -1;
}
final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
emptyBytes = new byte[byteBuffer.capacity()];
// Rather than passing the ByteBuffer with every callback (requiring
// the potentially expensive GetDirectBufferAddress) we simply have the
// the native class cache the address to the memory once.
nativeCacheDirectBufferAddress(byteBuffer, nativeAudioRecord);
// Get the minimum buffer size required for the successful creation of
// an AudioRecord object, in byte units.
// Note that this size doesn't guarantee a smooth recording under load.
final int channelConfig = channelCountToConfiguration(channels);
int minBufferSize =
AudioRecord.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
if (minBufferSize == AudioRecord.ERROR || minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
reportWebRtcAudioRecordInitError("AudioRecord.getMinBufferSize failed: " + minBufferSize);
return -1;
}
Logging.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize);
// Use a larger buffer size than the minimum required when creating the
// AudioRecord instance to ensure smooth recording under load. It has been
// verified that it does not increase the actual recording latency.
int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
try {
audioRecord = new AudioRecord(audIOSource, sampleRate, channelConfig,
AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes);
} catch (IllegalArgumentException e) {
reportWebRtcAudioRecordInitError("AudioRecord ctor error: " + e.getMessage());
releaseAudioResources();
return -1;
}
if (audioRecord == null || audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
reportWebRtcAudioRecordInitError("Failed to create a new AudioRecord instance");
releaseAudioResources();
return -1;
}
if (effects != null) {
effects.enable(audioRecord.getAudioSessionId());
}
logMainParameters();
logMainParametersExtended();
return framesPerBuffer;
}
在初始化的方法中，主要做了两件事。
private boolean startRecording() {
Logging.d(TAG, "startRecording");
assertTrue(audioRecord != null);
assertTrue(audioThread == null);
try {
audioRecord.startRecording();
} catch (IllegalStateException e) {
reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_EXCEPTION,
"AudioRecord.startRecording failed: " + e.getMessage());
return false;
}
if (audioRecord.getRecordingState() != AudioRecord.RECORDSTATE_RECORDING) {
reportWebRtcAudioRecordStartError(
AudioRecordStartErrorCode.AUDIO_RECORD_START_STATE_MISMATCH,
"AudioRecord.startRecording failed - incorrect state :"
+ audioRecord.getRecordingState());
return false;
}
audioThread = new AudioRecordThread("AudioRecordJavaThread");
audioThread.start();
return true;
}
在该方法中，首先启动了 audioRecord，接着判断了读取线程事都正在录制中。
private class AudioRecordThread extends Thread {
private volatile boolean keepAlive = true;
public AudioRecordThread(String name) {
super(name);
}
// TODO(titovartem) make correct fix during webrtc:9175
@SuppressWarnings("ByteBufferBackingArray")
@Override
public void run() {
Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);
long lastTime = System.nanoTime();
while (keepAlive) {
int bytesRead = audioRecord.read(byteBuffer, byteBuffer.capacity());
if (bytesRead == byteBuffer.capacity()) {
if (microphoneMute) {
byteBuffer.clear();
byteBuffer.put(emptyBytes);
}
// It's possible we've been shut down during the read, and stopRecording() tried and
// failed to join this thread. To be a bit safer, try to avoid calling any native methods
// in case they've been unregistered after stopRecording() returned.
if (keepAlive) {
nativeDataIsRecorded(bytesRead, nativeAudioRecord);
}
if (audioSamplesReadyCallback != null) {
// Copy the entire byte buffer array. Assume that the start of the byteBuffer is
// at index 0.
byte[] data = Arrays.copyOf(byteBuffer.array(), byteBuffer.capacity());
audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
new AudioSamples(audioRecord, data));
}
} else {
String errorMessage = "AudioRecord.read failed: " + bytesRead;
Logging.e(TAG, errorMessage);
if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
keepAlive = false;
reportWebRtcAudioRecordError(errorMessage);
}
}
if (DEBUG) {
long nowTime = System.nanoTime();
long durationInMs = TimeUnit.NANOSECONDS.toMillis((nowTime - lastTime));
lastTime = nowTime;
Logging.d(TAG, "bytesRead[" + durationInMs + "] " + bytesRead);
}
}
try {
if (audioRecord != null) {
audioRecord.stop();
}
} catch (IllegalStateException e) {
Logging.e(TAG, "AudioRecord.stop failed: " + e.getMessage());
}
}
// Stops the inner thread loop and also calls AudioRecord.stop().
// Does not block the calling thread.
public void stopThread() {
Logging.d(TAG, "stopThread");
keepAlive = false;
}
}
从 AudioRecord去数据的逻辑在 AudioRecordThread 线程的 Run函数中。
private boolean stopRecording() {
Logging.d(TAG, "stopRecording");
assertTrue(audioThread != null);
audioThread.stopThread();
if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS)) {
Logging.e(TAG, "Join of AudioRecordJavaThread timed out");
WebRtcAudioUtils.logAudioState(TAG);
}
audioThread = null;
if (effects != null) {
effects.release();
}
releaseAudioResources();
return true;
}
可以看到，这里首先把AudioRecordThread读数据循环的keepAlive条件置为false，接着调用
ThreadUtils.joinUninterruptibly等待AudioRecordThread线程退出。
这里有一点值得一提，keepAlive变量加了volatile关键字进行修饰，这是因为修改和读取这个变量的操作可能发生在不同的线程，使用volatile关键字进行修饰，可以保证修改之后能被立即读取到。
AudioRecordThread线程退出循环后，会调用audioRecord.stop()停止采集；线程退出之后，会调用audioRecord.release()释放AudioRecord对象。
以上，就是 Android WebRTC 音频采集 Java 层的大致流程。
参考《WebRTC 开发实战》
https://chromium.googlesource.com/external/webrtc/+/HEAD/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java
