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..1b73b6553fd8165fe3a144e205c5b4c3753d84a2 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
@@ -1491,29 +1491,4 @@ |
SendGenericPayload(); |
} |
-TEST_F(RtpSenderTest, SendAudioPadding) { |
- MockTransport transport; |
- const bool kEnableAudio = true; |
- rtp_sender_.reset(new RTPSender( |
- kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr, |
- nullptr, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, |
- nullptr, &retransmission_rate_limiter_, nullptr)); |
- rtp_sender_->SetSendPayloadType(kPayload); |
- rtp_sender_->SetSequenceNumber(kSeqNum); |
- rtp_sender_->SetTimestampOffset(0); |
- rtp_sender_->SetSSRC(kSsrc); |
- |
- const size_t kPaddingSize = 59; |
- EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _)) |
- .WillOnce(testing::Return(true)); |
- EXPECT_EQ(kPaddingSize, rtp_sender_->TimeToSendPadding( |
- kPaddingSize, PacketInfo::kNotAProbe)); |
- |
- // Requested padding size is too small, will send a larger one. |
- const size_t kMinPaddingSize = 50; |
- EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _)) |
- .WillOnce(testing::Return(true)); |
- EXPECT_EQ(kMinPaddingSize, rtp_sender_->TimeToSendPadding( |
- kMinPaddingSize - 5, PacketInfo::kNotAProbe)); |
-} |
} // namespace webrtc |