Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/pc/test/fakeaudiocapturemodule.cc

 /*
  *  Copyright 2012 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.
  */
 
-#include "webrtc/api/test/fakeaudiocapturemodule.h"
+#include "webrtc/pc/test/fakeaudiocapturemodule.h"
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
 
 // Audio sample value that is high enough that it doesn't occur naturally when
 // frames are being faked. E.g. NetEq will not generate this large sample value
 // unless it has received an audio frame containing a sample of this value.
 // Even simpler buffers would likely just contain audio sample values of 0.
@@ skipping 693 matching lines @@
                                               kNumberBytesPerSample,
                                               kNumberOfChannels,
                                               kSamplesPerSecond, kTotalDelayMs,
                                               kClockDriftMs, current_mic_level,
                                               key_pressed,
                                               current_mic_level) != 0) {
     ASSERT(false);
   }
   SetMicrophoneVolume(current_mic_level);
 }
-