Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/pc/dtmfsender_unittest.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/dtmfsender.h"
+#include "webrtc/pc/dtmfsender.h"
 
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
-#include "webrtc/api/audiotrack.h"
 #include "webrtc/base/fakeclock.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
+#include "webrtc/pc/audiotrack.h"
 
 using webrtc::AudioTrackInterface;
 using webrtc::AudioTrack;
 using webrtc::DtmfProviderInterface;
 using webrtc::DtmfSender;
 using webrtc::DtmfSenderObserverInterface;
 
 static const char kTestAudioLabel[] = "test_audio_track";
 // TODO(deadbeef): Even though this test now uses a fake clock, it has a
 // generous 3-second timeout for every test case. The timeout could be tuned
@@ skipping 308 matching lines @@
   std::string tones = "3,4";
   int duration = 100;
   int inter_tone_gap = 50;
 
   EXPECT_FALSE(dtmf_->InsertDtmf(tones, 6001, inter_tone_gap));
   EXPECT_FALSE(dtmf_->InsertDtmf(tones, 69, inter_tone_gap));
   EXPECT_FALSE(dtmf_->InsertDtmf(tones, duration, 49));
 
   EXPECT_TRUE(dtmf_->InsertDtmf(tones, duration, inter_tone_gap));
 }