Adding OnReceivedOverhead to AudioEncoder.

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 223 matching lines @@
               ElementsAre(states.encoder->next_frame_length_ms()));
   states.encoder->SetReceiverFrameLengthRange(0, 12345);
   EXPECT_THAT(states.encoder->supported_frame_lengths_ms(),
               ElementsAre(20, 60));
   states.encoder->SetReceiverFrameLengthRange(21, 60);
   EXPECT_THAT(states.encoder->supported_frame_lengths_ms(), ElementsAre(60));
   states.encoder->SetReceiverFrameLengthRange(20, 59);
   EXPECT_THAT(states.encoder->supported_frame_lengths_ms(), ElementsAre(20));
 }
 
-TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetUplinkBandwidth) {
+TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {

[minyue-webrtc, 2016/11/24 13:31:10]

changed because the name was not very good.

   auto states = CreateCodec(2);
   states.encoder->EnableAudioNetworkAdaptor("", nullptr);
 
   auto config = CreateEncoderRuntimeConfig();
   EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
       .WillOnce(Return(config));
 
   // Since using mock audio network adaptor, any bandwidth value is fine.
   constexpr int kUplinkBandwidth = 50000;
   EXPECT_CALL(**states.mock_audio_network_adaptor,
               SetUplinkBandwidth(kUplinkBandwidth));
   states.encoder->OnReceivedUplinkBandwidth(kUplinkBandwidth);
 
   CheckEncoderRuntimeConfig(states.encoder.get(), config);
 }
 
 TEST(AudioEncoderOpusTest,
-     InvokeAudioNetworkAdaptorOnSetUplinkPacketLossFraction) {
+     InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
   auto states = CreateCodec(2);
   states.encoder->EnableAudioNetworkAdaptor("", nullptr);
 
   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) {
+TEST(AudioEncoderOpusTest,
+     InvokeAudioNetworkAdaptorOnReceivedTargetAudioBitrate) {
   auto states = CreateCodec(2);
   states.encoder->EnableAudioNetworkAdaptor("", nullptr);
 
   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) {
+TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
   auto states = CreateCodec(2);
   states.encoder->EnableAudioNetworkAdaptor("", nullptr);
 
   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, InvokeAudioNetworkAdaptorOnReceivedOverhead) {
+  auto states = CreateCodec(2);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any overhead is fine.
+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(**states.mock_audio_network_adaptor, SetOverhead(kOverhead));
+  states.encoder->OnReceivedOverhead(kOverhead);
+
+  CheckEncoderRuntimeConfig(states.encoder.get(), config);
+}
+
 TEST(AudioEncoderOpusTest,
      PacketLossFractionSmoothedOnSetUplinkPacketLossFraction) {
   auto states = CreateCodec(2);
 
   // The values are carefully chosen so that if no smoothing is made, the test
   // will fail.
   constexpr float kPacketLossFraction_1 = 0.02f;
   constexpr float kPacketLossFraction_2 = 0.198f;
   // |kSecondSampleTimeMs| is chose to ease the calculation since
   // 0.9999 ^ 6931 = 0.5.
   constexpr float kSecondSampleTimeMs = 6931;
 
   // First time, no filtering.
   states.encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
   EXPECT_DOUBLE_EQ(0.01, states.encoder->packet_loss_rate());
 
   states.simulated_clock->AdvanceTimeMilliseconds(kSecondSampleTimeMs);
   states.encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
 
   // Now the output of packet loss fraction smoother should be
   // (0.02 + 0.198) / 2 = 0.109, which reach the threshold for the optimized
   // packet loss rate to increase to 0.05. If no smoothing has been made, the
   // optimized packet loss rate should have been increase to 0.1.
   EXPECT_DOUBLE_EQ(0.05, states.encoder->packet_loss_rate());
 }
 
 }  // namespace webrtc