Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc |
| index b3ee1a6cb639371a2d76df01011da0dce764bc85..cd3ec10c2683e88ae2ccd2f007c689cb25c7049e 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc |
| +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc |
| @@ -219,6 +219,8 @@ class TestTransport : public Transport, |
| namespace { |
| static const uint32_t kSenderSsrc = 0x11111111; |
| static const uint32_t kRemoteSsrc = 0x22222222; |
| +static const uint32_t kStartRtpTimestamp = 0x34567; |
| +static const uint32_t kRtpTimestamp = 0x45678; |
| } |
| class RtcpSenderTest : public ::testing::Test { |
| @@ -236,6 +238,8 @@ class RtcpSenderTest : public ::testing::Test { |
| nullptr, nullptr, &test_transport_)); |
| rtcp_sender_->SetSSRC(kSenderSsrc); |
| rtcp_sender_->SetRemoteSSRC(kRemoteSsrc); |
| + rtcp_sender_->SetStartTimestamp(kStartRtpTimestamp); |
| + rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds()); |
| } |
| void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) { |
| @@ -295,6 +299,8 @@ TEST_F(RtcpSenderTest, SendSr) { |
| EXPECT_EQ(ntp_frac, parser()->sender_report()->NtpFrac()); |
| EXPECT_EQ(kPacketCount, parser()->sender_report()->PacketCount()); |
| EXPECT_EQ(kOctetCount, parser()->sender_report()->OctetCount()); |
| + EXPECT_EQ(kStartRtpTimestamp + kRtpTimestamp, |
| + parser()->sender_report()->RtpTimestamp()); |
| EXPECT_EQ(0, parser()->report_block()->num_packets()); |
| } |
|
stefan-webrtc
2016/02/01 17:32:07
Should we add a test which verifies we don't send
danilchap
2016/02/01 18:24:18
test in video/end_to_end is one way to see this CL
|