Android技术分享| Android WebRTC 对 AudioRecord 的使用

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。

创建和初始化

  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