Java Logging.e方法代码示例

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean stopPlayout() {
  Logging.d(TAG, "stopPlayout");
  assertTrue(audioThread != null);
  logUnderrunCount();
  audioThread.stopThread();

  final Thread aThread = audioThread;
  audioThread = null;
  if (aThread != null) {
    Logging.d(TAG, "Stopping the AudioTrackThread...");
    aThread.interrupt();
    if (!ThreadUtils.joinUninterruptibly(aThread, AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS)) {
      Logging.e(TAG, "Join of AudioTrackThread timed out.");
    }
    Logging.d(TAG, "AudioTrackThread has now been stopped.");
  }

  releaseAudioResources();
  return true;
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean setStreamVolume(int volume) {
  Logging.d(TAG, "setStreamVolume(" + volume + ")");
  assertTrue(audioManager != null);
  if (isVolumeFixed()) {
    Logging.e(TAG, "The device implements a fixed volume policy.");
    return false;
  }
  audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
  return true;
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean enableBuiltInNS(boolean enable) {
  Logging.d(TAG, "enableBuiltInNS(" + enable + ')');
  if (effects == null) {
    Logging.e(TAG, "Built-in NS is not supported on this platform");
    return false;
  }
  return effects.setNS(enable);
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
public boolean setAEC(boolean enable) {
  Logging.d(TAG, "setAEC(" + enable + ")");
  if (!canUseAcousticEchoCanceler()) {
    Logging.w(TAG, "Platform AEC is not supported");
    shouldEnableAec = false;
    return false;
  }
  if (aec != null && (enable != shouldEnableAec)) {
    Logging.e(TAG, "Platform AEC state can't be modified while recording");
    return false;
  }
  shouldEnableAec = enable;
  return true;
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
public boolean setNS(boolean enable) {
  Logging.d(TAG, "setNS(" + enable + ")");
  if (!canUseNoiseSuppressor()) {
    Logging.w(TAG, "Platform NS is not supported");
    shouldEnableNs = false;
    return false;
  }
  if (ns != null && (enable != shouldEnableNs)) {
    Logging.e(TAG, "Platform NS state can't be modified while recording");
    return false;
  }
  shouldEnableNs = enable;
  return true;
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean enableBuiltInAEC(boolean enable) {
  threadChecker.checkIsOnValidThread();
  Logging.d(TAG, "enableBuiltInAEC(" + enable + ')');
  if (effects == null) {
    Logging.e(TAG, "Built-in AEC is not supported on this platform");
    return false;
  }
  return effects.setAEC(enable);
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean enableBuiltInNS(boolean enable) {
  threadChecker.checkIsOnValidThread();
  Logging.d(TAG, "enableBuiltInNS(" + enable + ')');
  if (effects == null) {
    Logging.e(TAG, "Built-in NS is not supported on this platform");
    return false;
  }
  return effects.setNS(enable);
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean stopRecording() {
  threadChecker.checkIsOnValidThread();
  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");
  }
  audioThread = null;
  if (effects != null) {
    effects.release();
  }
  releaseAudioResources();
  return true;
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private void reportWebRtcAudioRecordStartError(
    AudioRecordStartErrorCode errorCode, String errorMessage) {
  Logging.e(TAG, "Start recording error: " + errorCode + ". " + errorMessage);
  if (errorCallback != null) {
    errorCallback.onWebRtcAudioRecordStartError(errorCode, errorMessage);
  }
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private void threadSleep(long millis) {
  try {
    Thread.sleep(millis);
  } catch (InterruptedException e) {
    Logging.e(TAG, "Thread.sleep failed: " + e.getMessage());
  }
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean startPlayout() {
  Logging.d(TAG, "startPlayout");
  assertTrue(audioTrack != null);
  assertTrue(audioThread == null);
  if (audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
    Logging.e(TAG, "AudioTrack instance is not successfully initialized.");
    return false;
  }
  audioThread = new AudioTrackThread("AudioTrackJavaThread");
  audioThread.start();
  return true;
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
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");
  }
  audioThread = null;
  if (effects != null) {
    effects.release();
  }
  releaseAudioResources();
  return true;
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean isDeviceBlacklistedForOpenSLESUsage() {
  boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden
      ? blacklistDeviceForOpenSLESUsage
      : WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
  if (blacklisted) {
    Logging.e(TAG, Build.MODEL + " is blacklisted for OpenSL ES usage!");
  }
  return blacklisted;
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private boolean enableBuiltInAEC(boolean enable) {
  Logging.d(TAG, "enableBuiltInAEC(" + enable + ')');
  if (effects == null) {
    Logging.e(TAG, "Built-in AEC is not supported on this platform");
    return false;
  }
  return effects.setAEC(enable);
}
 

import org.webrtc.Logging; //导入方法依赖的package包/类
private void reportWebRtcAudioRecordError(String errorMessage) {
  Logging.e(TAG, "Run-time recording error: " + errorMessage);
  if (errorCallback != null) {
    errorCallback.onWebRtcAudioRecordError(errorMessage);
  }
}