Fix issue where the first audio packets significantly impacts initial BWE negatively.

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 102 matching lines @@
   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);
   DCHECK(remote_bitrate_estimator_ != nullptr);
-  remote_bitrate_estimator_->IncomingPacket(clock_->TimeInMilliseconds(),
-                                            length - 12, header, true);
+  remote_bitrate_estimator_->IncomingPacket(
+      clock_->TimeInMilliseconds(), length - header.headerLength, 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_;
   // Handle RTX retransmission, but only for non-padding-only packets.
   if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end() &&
@@ skipping 127 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, true);
+  remote_bitrate_estimator_->IncomingPacket(
+      clock_->TimeInMilliseconds(), length - header.headerLength, 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 171 matching lines @@
 }
 
 void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams,
                                      bool rtx,
                                      bool red) {
   test::DirectTransport receiver_transport;
   LowRateStreamObserver stream_observer(
       &receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx);
 
   Call::Config call_config(&stream_observer);
+  call_config.bitrate_config.start_bitrate_bps = 60000;
   CreateSenderCall(call_config);
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
   CreateSendConfig(number_of_streams);
 
   send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
   send_config_.rtp.extensions.push_back(RtpExtension(
       RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
   send_config_.suspend_below_min_bitrate = true;
+
   if (rtx) {
     send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
     send_config_.rtp.rtx.ssrcs = GenerateSsrcs(number_of_streams, 200);
   }
   if (red) {
     send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
     send_config_.rtp.fec.red_payload_type = kRedPayloadType;
   }
 
   CreateStreams();
@@ skipping 68 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