Using AudioOption to enable audio network adaptor.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
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               SetUplinkBandwidth(kUplinkBandwidth));
   states.encoder->OnReceivedUplinkBandwidth(kUplinkBandwidth);
 
   CheckEncoderRuntimeConfig(states.encoder.get(), config);
 }
 
 TEST(AudioEncoderOpusTest,
      InvokeAudioNetworkAdaptorOnSetUplinkPacketLossFraction) {
   auto states = CreateCodec(2, true);
 
-  auto config = CreateEncoderRuntimeConfig();
+  const auto config = CreateEncoderRuntimeConfig();
   EXPECT_CALL(*states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
       .WillOnce(Return(config));
 
   // Since using mock audio network adaptor, any packet loss fraction is fine.
   constexpr float kUplinkPacketLoss = 0.1f;
   EXPECT_CALL(*states.mock_audio_network_adaptor,
               SetUplinkPacketLossFraction(kUplinkPacketLoss));
   states.encoder->OnReceivedUplinkPacketLossFraction(kUplinkPacketLoss);
 
   CheckEncoderRuntimeConfig(states.encoder.get(), config);
 }
 
 TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetTargetAudioBitrate) {
   auto states = CreateCodec(2, true);
 
-  auto config = CreateEncoderRuntimeConfig();
+  const auto config = CreateEncoderRuntimeConfig();
   EXPECT_CALL(*states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
       .WillOnce(Return(config));
 
   // Since using mock audio network adaptor, any target audio bitrate is fine.
   constexpr int kTargetAudioBitrate = 30000;
   EXPECT_CALL(*states.mock_audio_network_adaptor,
               SetTargetAudioBitrate(kTargetAudioBitrate));
   states.encoder->OnReceivedTargetAudioBitrate(kTargetAudioBitrate);
 
   CheckEncoderRuntimeConfig(states.encoder.get(), config);
 }
 
 TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetRtt) {
   auto states = CreateCodec(2, true);
 
-  auto config = CreateEncoderRuntimeConfig();
+  const auto config = CreateEncoderRuntimeConfig();
   EXPECT_CALL(*states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
       .WillOnce(Return(config));
 
   // Since using mock audio network adaptor, any rtt is fine.
   constexpr int kRtt = 30;
   EXPECT_CALL(*states.mock_audio_network_adaptor, SetRtt(kRtt));
   states.encoder->OnReceivedRtt(kRtt);
 
   CheckEncoderRuntimeConfig(states.encoder.get(), config);
 }
 
 TEST(AudioEncoderOpusTest,
      InvokeAudioNetworkAdaptorOnSetReceiverFrameLengthRange) {
   auto states = CreateCodec(2, true);
 
-  auto config = CreateEncoderRuntimeConfig();
+  const auto config = CreateEncoderRuntimeConfig();
   EXPECT_CALL(*states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
       .WillOnce(Return(config));
 
   constexpr int kMinFrameLength = 10;
   constexpr int kMaxFrameLength = 60;
   EXPECT_CALL(*states.mock_audio_network_adaptor,
               SetReceiverFrameLengthRange(kMinFrameLength, kMaxFrameLength));
   states.encoder->SetReceiverFrameLengthRange(kMinFrameLength, kMaxFrameLength);
 
   CheckEncoderRuntimeConfig(states.encoder.get(), config);
 }
 
 }  // namespace webrtc