Drop return value from RtpRtcp::IncomingRtcpPacket.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_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 70 matching lines @@
   }
   bool SendRtcp(const uint8_t* data, size_t len) override {
     test::RtcpPacketParser parser;
     parser.Parse(data, len);
     last_nack_list_ = parser.nack()->packet_ids();
 
     if (clock_) {
       clock_->AdvanceTimeMilliseconds(delay_ms_);
     }
     EXPECT_TRUE(receiver_);
-    EXPECT_EQ(0, receiver_->IncomingRtcpPacket(data, len));
+    receiver_->IncomingRtcpPacket(data, len);
     return true;
   }
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                 size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override {
     return 0;
   }
   void SetKeepalivePayloadType(uint8_t payload_type) {
     keepalive_payload_type_ = payload_type;
   }
@@ skipping 146 matching lines @@
   void IncomingRtcpNack(const RtpRtcpModule* module, uint16_t sequence_number) {
     bool sender = module->impl_->SSRC() == kSenderSsrc;
     rtcp::Nack nack;
     uint16_t list[1];
     list[0] = sequence_number;
     const uint16_t kListLength = sizeof(list) / sizeof(list[0]);
     nack.SetSenderSsrc(sender ? kReceiverSsrc : kSenderSsrc);
     nack.SetMediaSsrc(sender ? kSenderSsrc : kReceiverSsrc);
     nack.SetPacketIds(list, kListLength);
     rtc::Buffer packet = nack.Build();
-    EXPECT_EQ(0, module->impl_->IncomingRtcpPacket(packet.data(),
-                                                   packet.size()));
+    module->impl_->IncomingRtcpPacket(packet.data(), packet.size());
   }
 };
 
 TEST_F(RtpRtcpImplTest, SetSelectiveRetransmissions_BaseLayer) {
   sender_.impl_->SetSelectiveRetransmissions(kRetransmitBaseLayer);
   EXPECT_EQ(kRetransmitBaseLayer, sender_.impl_->SelectiveRetransmissions());
 
   // Send frames.
   EXPECT_EQ(0, sender_.RtpSent());
   SendFrame(&sender_, kBaseLayerTid);    // kSequenceNumber
@@ skipping 368 matching lines @@
   clock_.AdvanceTimeMilliseconds(kTimeoutMs - 1);
   sender_.impl_->Process();
   EXPECT_EQ(2U, sender_.transport_.NumKeepaliveSent());
 
   // Timeout relative payload, send new keep-alive.
   clock_.AdvanceTimeMilliseconds(1);
   sender_.impl_->Process();
   EXPECT_EQ(3U, sender_.transport_.NumKeepaliveSent());
 }
 }  // namespace webrtc