Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

 /*
  *  Copyright (c) 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.
  */
 
@@ skipping 68 matching lines @@
   RtpRtcpAudioTest() : fake_clock(123456) {
     test_CSRC[0] = 1234;
     test_CSRC[2] = 2345;
     test_ssrc = 3456;
     test_timestamp = 4567;
     test_sequence_number = 2345;
   }
   ~RtpRtcpAudioTest() {}
 
   void SetUp() override {
+    audioFeedback = new NullRtpAudioFeedback();
     data_receiver1 = new VerifyingAudioReceiver();
     data_receiver2 = new VerifyingAudioReceiver();
     rtp_callback = new RTPCallback();
     transport1 = new LoopBackTransport();
     transport2 = new LoopBackTransport();
 
     receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
     receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
 
     rtp_payload_registry1_.reset(new RTPPayloadRegistry(
         RTPPayloadStrategy::CreateStrategy(true)));
     rtp_payload_registry2_.reset(new RTPPayloadRegistry(
         RTPPayloadStrategy::CreateStrategy(true)));
 
     RtpRtcp::Configuration configuration;
     configuration.audio = true;
     configuration.clock = &fake_clock;
     configuration.receive_statistics = receive_statistics1_.get();
     configuration.outgoing_transport = transport1;
 
     module1 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
-        &fake_clock, data_receiver1, NULL, rtp_payload_registry1_.get()));
+        &fake_clock, audioFeedback, data_receiver1, NULL,
+        rtp_payload_registry1_.get()));
 
     configuration.receive_statistics = receive_statistics2_.get();
     configuration.outgoing_transport = transport2;
 
     module2 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
-        &fake_clock, data_receiver2, NULL, rtp_payload_registry2_.get()));
+        &fake_clock, audioFeedback, data_receiver2, NULL,
+        rtp_payload_registry2_.get()));
 
     transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
                               rtp_receiver2_.get(), receive_statistics2_.get());
     transport2->SetSendModule(module1, rtp_payload_registry1_.get(),
                               rtp_receiver1_.get(), receive_statistics1_.get());
   }
 
   void TearDown() override {
     delete module1;
     delete module2;
     delete transport1;
     delete transport2;
+    delete audioFeedback;
     delete data_receiver1;
     delete data_receiver2;
     delete rtp_callback;
   }
 
   RtpRtcp* module1;
   RtpRtcp* module2;
   rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
   rtc::scoped_ptr<ReceiveStatistics> receive_statistics2_;
   rtc::scoped_ptr<RtpReceiver> rtp_receiver1_;
   rtc::scoped_ptr<RtpReceiver> rtp_receiver2_;
   rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
   rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
   VerifyingAudioReceiver* data_receiver1;
   VerifyingAudioReceiver* data_receiver2;
   LoopBackTransport* transport1;
   LoopBackTransport* transport2;
+  NullRtpAudioFeedback* audioFeedback;
   RTPCallback* rtp_callback;
   uint32_t test_ssrc;
   uint32_t test_timestamp;
   uint16_t test_sequence_number;
   uint32_t test_CSRC[webrtc::kRtpCsrcSize];
   SimulatedClock fake_clock;
 };
 
 TEST_F(RtpRtcpAudioTest, Basic) {
   module1->SetSSRC(test_ssrc);
@@ skipping 180 matching lines @@
   for (; timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
 
 }  // namespace
 }  // namespace webrtc