Pass event log to ANA.

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 41 matching lines @@
 };
 
 AudioEncoderOpusStates CreateCodec(size_t num_channels) {
   AudioEncoderOpusStates states;
   states.mock_audio_network_adaptor =
       std::make_shared<MockAudioNetworkAdaptor*>(nullptr);
 
   std::weak_ptr<MockAudioNetworkAdaptor*> mock_ptr(
       states.mock_audio_network_adaptor);
   AudioEncoderOpus::AudioNetworkAdaptorCreator creator = [mock_ptr](
-      const std::string&, const Clock*) {
+      const std::string&, RtcEventLog* event_log, const Clock*) {
     std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
         new NiceMock<MockAudioNetworkAdaptor>());
     EXPECT_CALL(*adaptor, Die());
     if (auto sp = mock_ptr.lock()) {
       *sp = adaptor.get();
     } else {
       RTC_NOTREACHED();
     }
     return adaptor;
   };
@@ skipping 186 matching lines @@
               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,
      InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
   auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr, 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, InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
   auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr, 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;
   constexpr int64_t kProbingIntervalMs = 3000;
   EXPECT_CALL(**states.mock_audio_network_adaptor,
               SetTargetAudioBitrate(kTargetAudioBitrate));
   EXPECT_CALL(*states.mock_bitrate_smoother,
               SetTimeConstantMs(kProbingIntervalMs * 4));
   EXPECT_CALL(*states.mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
   states.encoder->OnReceivedUplinkBandwidth(
       kTargetAudioBitrate, rtc::Optional<int64_t>(kProbingIntervalMs));
 
   CheckEncoderRuntimeConfig(states.encoder.get(), config);
 }
 
 TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
   auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr, 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);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr, 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);
 
@@ skipping 108 matching lines @@
   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, EmptyConfigDoesNotAffectEncoderSettings) {
   auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr, nullptr);
 
   auto config = CreateEncoderRuntimeConfig();
   AudioNetworkAdaptor::EncoderRuntimeConfig empty_config;
 
   EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
       .WillOnce(Return(config))
       .WillOnce(Return(empty_config));
 
   constexpr size_t kOverhead = 64;
   EXPECT_CALL(**states.mock_audio_network_adaptor, SetOverhead(kOverhead))
       .Times(2);
   states.encoder->OnReceivedOverhead(kOverhead);
   states.encoder->OnReceivedOverhead(kOverhead);
 
   CheckEncoderRuntimeConfig(states.encoder.get(), config);
 }
 
 TEST(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
   rtc::ScopedFakeClock fake_clock;
   auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", nullptr, nullptr);
   std::array<int16_t, 480 * 2> audio;
   audio.fill(0);
   rtc::Buffer encoded;
   EXPECT_CALL(*states.mock_bitrate_smoother, GetAverage())
       .WillOnce(Return(rtc::Optional<float>(50000)));
   EXPECT_CALL(**states.mock_audio_network_adaptor, SetUplinkBandwidth(50000));
   states.encoder->Encode(
       0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
 
   // Repeat update uplink bandwidth tests.
   for (int i = 0; i < 5; i++) {
     // Don't update till it is time to update again.
     fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(
         states.config.uplink_bandwidth_update_interval_ms - 1));
     states.encoder->Encode(
         0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
 
     // Update when it is time to update.
     EXPECT_CALL(*states.mock_bitrate_smoother, GetAverage())
         .WillOnce(Return(rtc::Optional<float>(40000)));
     EXPECT_CALL(**states.mock_audio_network_adaptor, SetUplinkBandwidth(40000));
     fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
     states.encoder->Encode(
         0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
   }
 }
 
 }  // namespace webrtc