Only use paced packets for estimating bitrate probes.

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 103 matching lines @@
   rtp_rtcp_->SetREMBData(bitrate, ssrcs);
   rtp_rtcp_->Process();
 }
 
 bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
   rtc::CritScope lock(&crit_);
   RTPHeader header;
   EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
   receive_stats_->IncomingPacket(header, length, false);
   payload_registry_->SetIncomingPayloadType(header);
-  remote_bitrate_estimator_->IncomingPacket(
-      clock_->TimeInMilliseconds(), length - 12, header);
+  remote_bitrate_estimator_->IncomingPacket(clock_->TimeInMilliseconds(),
+                                            length - 12, header, true);
   if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
     remote_bitrate_estimator_->Process();
   }
   total_sent_ += length;
   padding_sent_ += header.paddingLength;
   ++total_packets_sent_;
   if (header.paddingLength > 0)
     ++padding_packets_sent_;
   if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end()) {
     rtx_media_sent_ += length - header.headerLength - header.paddingLength;
@@ skipping 124 matching lines @@
   }
   return test::DirectTransport::SendRtp(data, length);
 }
 
 PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket(
     MediaType media_type, const uint8_t* packet, size_t length) {
   rtc::CritScope lock(&crit_);
   RTPHeader header;
   EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
   receive_stats_->IncomingPacket(header, length, false);
-  remote_bitrate_estimator_->IncomingPacket(
-      clock_->TimeInMilliseconds(), length - 12, header);
+  remote_bitrate_estimator_->IncomingPacket(clock_->TimeInMilliseconds(),
+                                            length - 12, header, true);
   if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
     remote_bitrate_estimator_->Process();
   }
   suspended_in_stats_ = send_stream_->GetStats().suspended;
   return DELIVERY_OK;
 }
 
 bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
   return true;
 }
@@ skipping 269 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