Reland of Removes usage of native base::android::GetApplicationContext()

webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 package org.webrtc.voiceengine;
 
 import android.annotation.TargetApi;
 import android.content.Context;
 import android.media.AudioFormat;
 import android.media.AudioRecord;
 import android.media.MediaRecorder.AudioSource;
 import android.os.Process;
 import java.lang.System;
 import java.nio.ByteBuffer;
 import java.util.concurrent.TimeUnit;
+import org.webrtc.ContextUtils;
 import org.webrtc.Logging;
 import org.webrtc.ThreadUtils;
 
 public class WebRtcAudioRecord {
   private static final boolean DEBUG = false;
 
   private static final String TAG = "WebRtcAudioRecord";
 
   // Default audio data format is PCM 16 bit per sample.
   // Guaranteed to be supported by all devices.
   private static final int BITS_PER_SAMPLE = 16;
 
   // Requested size of each recorded buffer provided to the client.
   private static final int CALLBACK_BUFFER_SIZE_MS = 10;
 
   // Average number of callbacks per second.
   private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
 
   // We ask for a native buffer size of BUFFER_SIZE_FACTOR * (minimum required
   // buffer size). The extra space is allocated to guard against glitches under
   // high load.
   private static final int BUFFER_SIZE_FACTOR = 2;
 
   // The AudioRecordJavaThread is allowed to wait for successful call to join()
   // but the wait times out afther this amount of time.
   private static final long AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS = 2000;
 
   private final long nativeAudioRecord;
-  private final Context context;
 
   private WebRtcAudioEffects effects = null;
 
   private ByteBuffer byteBuffer;
 
   private AudioRecord audioRecord = null;
   private AudioRecordThread audioThread = null;
 
   private static volatile boolean microphoneMute = false;
   private byte[] emptyBytes;
@@ skipping 71 matching lines @@
     }
 
     // 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;
     }
   }
 
-  WebRtcAudioRecord(Context context, long nativeAudioRecord) {
+  WebRtcAudioRecord(long nativeAudioRecord) {
     Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
-    this.context = context;
     this.nativeAudioRecord = nativeAudioRecord;
     if (DEBUG) {
       WebRtcAudioUtils.logDeviceInfo(TAG);
     }
     effects = WebRtcAudioEffects.create();
   }
 
   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);
   }
 
   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);
   }
 
   private int initRecording(int sampleRate, int channels) {
     Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")");
-    if (!WebRtcAudioUtils.hasPermission(context, android.Manifest.permission.RECORD_AUDIO)) {
+    if (!WebRtcAudioUtils.hasPermission(
+            ContextUtils.getApplicationContext(), android.Manifest.permission.RECORD_AUDIO)) {
       reportWebRtcAudioRecordInitError("RECORD_AUDIO permission is missing");
       return -1;
     }
     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);
@@ skipping 141 matching lines @@
     }
   }
 
   private void reportWebRtcAudioRecordError(String errorMessage) {
     Logging.e(TAG, "Run-time recording error: " + errorMessage);
     if (errorCallback != null) {
       errorCallback.onWebRtcAudioRecordError(errorMessage);
     }
   }
 }