Make a the decisions of ANA optional for the opus encoder.

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.
  */
 
@@ skipping 76 matching lines @@
   config.bitrate_bps = rtc::Optional<int>(kBitrate);
   config.frame_length_ms = rtc::Optional<int>(kFrameLength);
   config.enable_fec = rtc::Optional<bool>(kEnableFec);
   config.enable_dtx = rtc::Optional<bool>(kEnableDtx);
   config.num_channels = rtc::Optional<size_t>(kNumChannels);
   config.uplink_packet_loss_fraction =
       rtc::Optional<float>(kPacketLossFraction);
   return config;
 }
 
 void CheckEncoderRuntimeConfig(

[minyue-webrtc, 2016/12/22 14:11:26]

I suggest we somehow adding a new feature to this method to see that empty
fields follows the old encoder mode.

[minyue-webrtc, 2016/12/22 16:00:43]

Do this

void ApplyEncoderRuntimeConfig(const AudioEncoderOpus* encoder,
                                                         const
AudioNetworkAdaptor::EncoderRuntimeConfig& config) {

  int bitrate = encoder->GetTargetBitrate();
  ...
  // Other method like ... can also trigger a new codec configuration. We use
... here
  encoder->On...
  EXPECT_EQ(config.bitrate_bps.value_or(bitrate), encoder->GetTargetBitrate())
  ...
}

     const AudioEncoderOpus* encoder,
     const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
   EXPECT_EQ(*config.bitrate_bps, encoder->GetTargetBitrate());
   EXPECT_EQ(*config.frame_length_ms, encoder->next_frame_length_ms());
   EXPECT_EQ(*config.enable_fec, encoder->fec_enabled());
   EXPECT_EQ(*config.enable_dtx, encoder->GetDtx());
   EXPECT_EQ(*config.num_channels, encoder->num_channels_to_encode());
 }
 
 }  // namespace
@@ skipping 328 matching lines @@
   EXPECT_EQ(rtc::Optional<int>(8), config.GetNewComplexity());
 
   // Bitrate within hysteresis window. Expect empty output.
   config.bitrate_bps = rtc::Optional<int>(12500);
   EXPECT_EQ(rtc::Optional<int>(), config.GetNewComplexity());
 
   // Bitrate above hysteresis window. Expect lower complexity.
   config.bitrate_bps = rtc::Optional<int>(14001);
   EXPECT_EQ(rtc::Optional<int>(6), config.GetNewComplexity());
 }
+
+TEST(AudioEncoderOpusTest, ApplyAudioNetworkAdaptorCanHandlerNotDefinedEntrys) {

[minyue-webrtc, 2016/12/22 14:11:26]

Handler -> Handle

NotDefined -> Empty

Entrys -> Entries

+  auto states = CreateCodec(2);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  AudioNetworkAdaptor::EncoderRuntimeConfig config;
+  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Donne to force a call of ApplyAudioNetworkAdaptor.

[minyue-webrtc, 2016/12/22 14:11:26]

I suggest remove this comment.

+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(**states.mock_audio_network_adaptor, SetOverhead(kOverhead));
+  states.encoder->OnReceivedOverhead(kOverhead);
+}
+
 }  // namespace webrtc