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 2a7caf188f69c2df0a2c85b6f331f7a0ad1337b0..bb068e7d22e92dd7b123fbcb878a25e5f228b737 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc |
@@ -219,4 +219,39 @@ 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); |
+ 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 |