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

webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index 52ef81a67c050bfdc847a99a40dd0211bc7431e3..c7d90be2df93c6d2a7ce7981be35035347edbad4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -220,4 +220,39 @@ TEST_F(RtpPacketHistoryTest, FullExpansion) {
   }
 }
 
+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->AllocatePayload(kMinPacketRequestBytes - header_size);

[danilchap, 2017/03/30 11:58:08]

SetPayloadSize should be usable if you rebase.
As you mention, that name make more sense.

+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
+
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  packet = CreateRtpPacket(kSeqNum + 1);
+  packet->AllocatePayload(kMinPacketRequestBytes);
+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
+
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  packet = CreateRtpPacket(kSeqNum + 2);
+  packet->AllocatePayload(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