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 1427 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