Revert of Enable audio streams to send padding.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1473 matching lines @@
                         sizeof(kPayloadData) + kGenericHeaderLength +
                             kRtpOverheadBytesPerPacket,
                         PacketInfo::kNotAProbe))
       .Times(1);
   EXPECT_CALL(mock_overhead_observer,
               OnOverheadChanged(kRtpOverheadBytesPerPacket))
       .Times(1);
   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