Make sure RTCP is sent in tests when receiving packets even if REMB is delayed.

webrtc/video/rampup_tests.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 106 matching lines @@
   rtc::CritScope lock(&crit_);
   RTPHeader header;
   EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
   receive_stats_->IncomingPacket(header, length, false);
   payload_registry_->SetIncomingPayloadType(header);
   DCHECK(remote_bitrate_estimator_ != nullptr);
   remote_bitrate_estimator_->IncomingPacket(
       clock_->TimeInMilliseconds(), length - header.headerLength, header, true);
   if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
     remote_bitrate_estimator_->Process();
+    rtp_rtcp_->Process();
   }
   total_sent_ += length;
   padding_sent_ += header.paddingLength;
   ++total_packets_sent_;
   if (header.paddingLength > 0)
     ++padding_packets_sent_;
   // Handle RTX retransmission, but only for non-padding-only packets.
   if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end() &&
       header.headerLength + header.paddingLength != length) {
     rtx_media_sent_ += length - header.headerLength - header.paddingLength;
@@ skipping 417 matching lines @@
 
 TEST_F(RampUpTest, AbsSendTimeSimulcastByRedWithRtx) {
   RunRampUpTest(3, 0, RtpExtension::kAbsSendTime, true, true);
 }
 
 TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
   RunRampUpTest(1, 0.9 * kSingleStreamTargetBps, RtpExtension::kAbsSendTime,
                 false, false);
 }
 }  // namespace webrtc