Relanding of "Adding debug dump to audio network adaptor."

webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc

 /*
  *  Copyright (c) 2016 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.
  */
 
 #include <utility>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
 
 namespace webrtc {
 
 using ::testing::_;
 using ::testing::NiceMock;
 using ::testing::Return;
+using ::testing::SetArgPointee;
 
 namespace {
 
 constexpr size_t kNumControllers = 2;
 
+constexpr int64_t kClockInitialTimeMs = 12345678;
+
 MATCHER_P(NetworkMetricsIs, metric, "") {
   return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps &&
          arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps &&
+         arg.rtt_ms == metric.rtt_ms &&
          arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
 }
 
 MATCHER_P(ConstraintsReceiverFrameLengthRangeIs, frame_length_range, "") {
   return arg.receiver_frame_length_range->min_frame_length_ms ==
              frame_length_range.min_frame_length_ms &&
          arg.receiver_frame_length_range->max_frame_length_ms ==
              frame_length_range.max_frame_length_ms;
 }
 
+MATCHER_P(EncoderRuntimeConfigIs, config, "") {
+  return arg.bitrate_bps == config.bitrate_bps &&
+         arg.frame_length_ms == config.frame_length_ms &&
+         arg.uplink_packet_loss_fraction ==
+             config.uplink_packet_loss_fraction &&
+         arg.enable_fec == config.enable_fec &&
+         arg.enable_dtx == config.enable_dtx &&
+         arg.num_channels == config.num_channels;
+}
+
 struct AudioNetworkAdaptorStates {
   std::unique_ptr<AudioNetworkAdaptorImpl> audio_network_adaptor;
   std::vector<std::unique_ptr<MockController>> mock_controllers;
+  std::unique_ptr<SimulatedClock> simulated_clock;
+  MockDebugDumpWriter* mock_debug_dump_writer;
 };
 
 AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() {
   AudioNetworkAdaptorStates states;
   std::vector<Controller*> controllers;
   for (size_t i = 0; i < kNumControllers; ++i) {
     auto controller =
         std::unique_ptr<MockController>(new NiceMock<MockController>());
     EXPECT_CALL(*controller, Die());
     controllers.push_back(controller.get());
     states.mock_controllers.push_back(std::move(controller));
   }
 
   auto controller_manager = std::unique_ptr<MockControllerManager>(
       new NiceMock<MockControllerManager>());
 
   EXPECT_CALL(*controller_manager, Die());
   EXPECT_CALL(*controller_manager, GetControllers())
       .WillRepeatedly(Return(controllers));
   EXPECT_CALL(*controller_manager, GetSortedControllers(_))
       .WillRepeatedly(Return(controllers));
 
+  states.simulated_clock.reset(new SimulatedClock(kClockInitialTimeMs * 1000));
+
+  auto debug_dump_writer =
+      std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
+  EXPECT_CALL(*debug_dump_writer, Die());
+  states.mock_debug_dump_writer = debug_dump_writer.get();
+
+  AudioNetworkAdaptorImpl::Config config;
+  config.clock = states.simulated_clock.get();
   // AudioNetworkAdaptorImpl governs the lifetime of controller manager.
   states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl(
-      AudioNetworkAdaptorImpl::Config(), std::move(controller_manager)));
+      config,
+      std::move(controller_manager), std::move(debug_dump_writer)));
 
   return states;
 }
 
 }  // namespace
 
 TEST(AudioNetworkAdaptorImplTest,
      MakeDecisionIsCalledOnGetEncoderRuntimeConfig) {
   auto states = CreateAudioNetworkAdaptor();
 
@@ skipping 21 matching lines @@
   auto states = CreateAudioNetworkAdaptor();
 
   for (auto& mock_controller : states.mock_controllers) {
     EXPECT_CALL(*mock_controller,
                 SetConstraints(ConstraintsReceiverFrameLengthRangeIs(
                     Controller::Constraints::FrameLengthRange(20, 120))));
   }
   states.audio_network_adaptor->SetReceiverFrameLengthRange(20, 120);
 }
 
+TEST(AudioNetworkAdaptorImplTest,
+     DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
+  auto states = CreateAudioNetworkAdaptor();
+
+  AudioNetworkAdaptor::EncoderRuntimeConfig config;
+  config.bitrate_bps = rtc::Optional<int>(32000);
+  config.enable_fec = rtc::Optional<bool>(true);
+
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_, _))
+      .WillOnce(SetArgPointee<1>(config));
+
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpEncoderRuntimeConfig(EncoderRuntimeConfigIs(config),
+                                       kClockInitialTimeMs));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
+  auto states = CreateAudioNetworkAdaptor();
+
+  constexpr int kBandwidth = 16000;
+  constexpr float kPacketLoss = 0.7f;
+  constexpr int kRtt = 100;
+
+  Controller::NetworkMetrics check;
+  check.uplink_bandwidth_bps = rtc::Optional<int>(kBandwidth);
+  int64_t timestamp_check = kClockInitialTimeMs;
+
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
+
+  states.simulated_clock->AdvanceTimeMilliseconds(100);
+  timestamp_check += 100;
+  check.uplink_packet_loss_fraction = rtc::Optional<float>(kPacketLoss);
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
+
+  states.simulated_clock->AdvanceTimeMilliseconds(200);
+  timestamp_check += 200;
+  check.rtt_ms = rtc::Optional<int>(kRtt);
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetRtt(kRtt);
+}
+
 }  // namespace webrtc