Correcting the amount of padding when send side bwe includes RTP overhead.

webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc

 /*
  *  Copyright (c) 2012 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 202 matching lines @@
   }
 
   fake_clock_.AdvanceTimeMilliseconds(100);
 
   // Retransmit all packets currently in buffer.
   for (size_t i = 1; i < RtpPacketHistory::kMaxCapacity + 1; ++i) {
     EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum + i, 100, false));
   }
 }
 
+TEST_F(RtpPacketHistoryTest, GetBestFittingPacket) {
+  constexpr size_t kMinPacketRequestBytes = 50;
+
+  hist_.SetStorePacketsStatus(true, 3);
+
+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum);
+  const size_t header_size = packet->headers_size();
+  const int64_t start_time_ms = fake_clock_.TimeInMilliseconds();
+
+  packet->SetPayloadSize(kMinPacketRequestBytes - header_size);
+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
+
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  packet = CreateRtpPacket(kSeqNum + 1);
+  packet->SetPayloadSize(kMinPacketRequestBytes);
+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
+
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  packet = CreateRtpPacket(kSeqNum + 2);
+  packet->SetPayloadSize(kMinPacketRequestBytes + header_size);
+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
+
+  constexpr bool kIncludeHeader = true;
+  std::unique_ptr<RtpPacketToSend> fit =
+      hist_.GetBestFittingPacket(kMinPacketRequestBytes, kIncludeHeader);
+  ASSERT_TRUE(fit);
+  EXPECT_EQ(start_time_ms, fit->capture_time_ms());
+
+  fit = hist_.GetBestFittingPacket(kMinPacketRequestBytes, !kIncludeHeader);
+  ASSERT_TRUE(fit);
+  EXPECT_EQ(start_time_ms + 1, fit->capture_time_ms());
+}
+
 }  // namespace webrtc