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首页 > 软件编程 > Android > Android WebRTC 对 AudioRecord 的使用

Android WebRTC 对 AudioRecord 的使用技术分享

作者:anyRTC

这篇文章主要介绍了Android WebRTC 对 AudioRecord 的使用技术分享,AudioRecord 是 Android 基于原始PCM音频数据录制的类,接下来和小编进入文章了解更详细的内容吧

前言:

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。

一、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
    // 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。

创建缓冲区:

创建 AudioRecord对象,构造函数有很多参数,分析如下:

二、启动

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。

三、读数据

 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。

四、停止和销毁

  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。

到此这篇关于Android WebRTC 对 AudioRecord 的使用技术分享的文章就介绍到这了,更多相关Android WebRTC 对 AudioRecord 的使用内容请搜索脚本之家以前的文章或继续浏览下面的相关文章希望大家以后多多支持脚本之家!

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