Small cleanup of rtp_rtcp testAPI tests.

webrtc/modules/rtp_rtcp/test/testAPI/test_api.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 39 matching lines @@
   }
   PayloadUnion payload_specific;
   if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                   &payload_specific)) {
     return false;
   }
   const uint8_t* payload = data + header.headerLength;
   RTC_CHECK_GE(len, header.headerLength);
   const size_t payload_length = len - header.headerLength;
   receive_statistics_->IncomingPacket(header, len, false);
-  if (!rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
-                                        payload_specific, true)) {
-    return false;
-  }
-  return true;
+  return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
+                                          payload_specific, true);
 }
 
 bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) {
   if (rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
     return false;
   }
   return true;
 }
 
 int32_t TestRtpReceiver::OnReceivedPayloadData(
@@ skipping 23 matching lines @@
 
   void SetUp() override {
     RtpRtcp::Configuration configuration;
     configuration.audio = true;
     configuration.clock = &fake_clock_;
     configuration.outgoing_transport = &null_transport_;
     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
     module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
     module_->SetSSRC(initial_ssrc);
     rtp_payload_registry_.reset(new RTPPayloadRegistry());
-    rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
-        &fake_clock_, NULL, NULL, rtp_payload_registry_.get()));
   }
 
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
-  std::unique_ptr<RtpReceiver> rtp_receiver_;
   std::unique_ptr<RtpRtcp> module_;
   uint32_t test_ssrc_;
   uint32_t test_timestamp_;
   uint16_t test_sequence_number_;
   std::vector<uint32_t> test_csrcs_;
   SimulatedClock fake_clock_;
   test::NullTransport null_transport_;
   RateLimiter retransmission_rate_limiter_;
 };
 
@@ skipping 57 matching lines @@
   rtx_header.payloadType = kRtxPayloadType;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = 0;
   EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = kRtxSsrc;
   rtx_header.payloadType = 0;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
 }
 
 }  // namespace webrtc