Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
index a6a886be94b023e11361b0cef66dc086dad37dd6..c8bbc52fb0af451140b88cda5a2b0fadcbc30bdd 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
@@ -343,6 +343,7 @@ TEST_F(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) { |
false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_, |
&feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_, |
&send_packet_observer_, &retransmission_rate_limiter_, nullptr)); |
+ rtp_sender_->SetSSRC(kSsrc); |
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
kRtpExtensionTransportSequenceNumber, |
kTransportSequenceNumberExtensionId)); |
@@ -959,7 +960,7 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) { |
new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_, |
nullptr, nullptr, nullptr, nullptr, &callback, nullptr, |
nullptr, nullptr, &retransmission_rate_limiter_, nullptr)); |
- |
+ rtp_sender_->SetSSRC(kSsrc); |
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; |
const uint8_t payload_type = 127; |
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, |
@@ -1021,6 +1022,7 @@ TEST_F(RtpSenderTest, BitrateCallbacks) { |
nullptr, nullptr, nullptr, &callback, nullptr, |
nullptr, nullptr, nullptr, |
&retransmission_rate_limiter_, nullptr)); |
+ rtp_sender_->SetSSRC(kSsrc); |
// Simulate kNumPackets sent with kPacketInterval ms intervals, with the |
// number of packets selected so that we fill (but don't overflow) the one |
@@ -1079,6 +1081,7 @@ class RtpSenderAudioTest : public RtpSenderTest { |
nullptr, nullptr, nullptr, nullptr, nullptr, |
nullptr, nullptr, nullptr, |
&retransmission_rate_limiter_, nullptr)); |
+ rtp_sender_->SetSSRC(kSsrc); |
rtp_sender_->SetSequenceNumber(kSeqNum); |
} |
}; |
@@ -1439,6 +1442,7 @@ TEST_F(RtpSenderTest, OnOverheadChanged) { |
new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, |
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr, |
&retransmission_rate_limiter_, &mock_overhead_observer)); |
+ rtp_sender_->SetSSRC(kSsrc); |
// RTP overhead is 12B. |
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(12)).Times(1); |
@@ -1459,6 +1463,7 @@ TEST_F(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) { |
new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, |
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr, |
&retransmission_rate_limiter_, &mock_overhead_observer)); |
+ rtp_sender_->SetSSRC(kSsrc); |
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); |
SendGenericPayload(); |
@@ -1474,6 +1479,7 @@ TEST_F(RtpSenderTest, AddOverheadToTransportFeedbackObserver) { |
false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_, |
&feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_, |
nullptr, &retransmission_rate_limiter_, &mock_overhead_observer)); |
+ rtp_sender_->SetSSRC(kSsrc); |
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( |
kRtpExtensionTransportSequenceNumber, |
kTransportSequenceNumberExtensionId)); |