Revert of Enable audio streams to send padding.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

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