Update smoothed bitrate.

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc

 /*
  *  Copyright (c) 2014 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 80 matching lines @@
       .WillOnce(Return(17U));
   EXPECT_EQ(17U, red_->Num10MsFramesInNextPacket());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
   EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17U));
   EXPECT_EQ(17U, red_->Max10MsFramesInAPacket());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
-  EXPECT_CALL(*mock_encoder_, OnReceivedTargetAudioBitrate(4711));
-  red_->OnReceivedTargetAudioBitrate(4711);
+  EXPECT_CALL(*mock_encoder_,
+              OnReceivedTargetAudioBitrate(4711, rtc::Optional<int64_t>()));
+  red_->OnReceivedTargetAudioBitrate(4711, rtc::Optional<int64_t>());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
   EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
   red_->OnReceivedUplinkPacketLossFraction(0.5);
 }
 
 // Checks that the an Encode() call is immediately propagated to the speech
 // encoder.
 TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) {
@@ skipping 184 matching lines @@
   config.speech_encoder = NULL;
   EXPECT_DEATH(red = new AudioEncoderCopyRed(std::move(config)),
                "Speech encoder not provided.");
   // The delete operation is needed to avoid leak reports from memcheck.
   delete red;
 }
 
 #endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 }  // namespace webrtc