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Unified Diff: webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc

Issue 1639253007: Validates sending RTCP before RTP. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: adjusted RtpRtcpImplTest to comply with stricter conditions for Sender Report Created 4 years, 11 months ago
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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

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