Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry

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 83 matching lines @@
     test_ssrc = 3456;
     test_timestamp = 4567;
     test_sequence_number = 2345;
   }
   ~RtpRtcpAudioTest() {}
 
   void SetUp() override {
     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)));
+    rtp_payload_registry1_.reset(new RTPPayloadRegistry());
+    rtp_payload_registry2_.reset(new RTPPayloadRegistry());
 
     RtpRtcp::Configuration configuration;
     configuration.audio = true;
     configuration.clock = &fake_clock;
     configuration.receive_statistics = receive_statistics1_.get();
     configuration.outgoing_transport = &transport1;
     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
 
     module1.reset(RtpRtcp::CreateRtpRtcp(configuration));
     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
@@ skipping 165 matching lines @@
                                           nullptr, nullptr, nullptr));
 
     EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
     EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
     EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
     in_timestamp += 10;
   }
 }
 
 }  // namespace webrtc