Drop return value from RtpRtcp::IncomingRtcpPacket.

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

 /*
 *  Copyright (c) 2013 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 134 matching lines @@
     }
     if (!rtp_receiver_->IncomingRtpPacket(header, ptr + header.headerLength,
                                           packet_length - header.headerLength,
                                           payload_specific, true)) {
       return false;
     }
     return true;
   }
 
   bool SendRtcp(const uint8_t* data, size_t len) override {
-    return module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0;
+    module_->IncomingRtcpPacket((const uint8_t*)data, len);
+    return true;
   }
   int count_;
   int packet_loss_;
   int consecutive_drop_start_;
   int consecutive_drop_end_;
   uint32_t rtx_ssrc_;
   int count_rtx_ssrc_;
   RTPPayloadRegistry* rtp_payload_registry_;
   RtpReceiver* rtp_receiver_;
   RtpRtcp* module_;
@@ skipping 168 matching lines @@
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
             *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }
 
 }  // namespace webrtc