Hooking up target audio bitrate 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.
  */
 
@@ skipping 88 matching lines @@
 }
 
 }  // namespace
 
 TEST(AudioNetworkAdaptorImplTest,
      MakeDecisionIsCalledOnGetEncoderRuntimeConfig) {
   auto states = CreateAudioNetworkAdaptor();
 
   constexpr int kBandwidth = 16000;
   constexpr float kPacketLoss = 0.7f;
+  constexpr int kRtt = 100;
+  constexpr int kTargetAudioBitrate = 15000;
 
   Controller::NetworkMetrics check;
   check.uplink_bandwidth_bps = rtc::Optional<int>(kBandwidth);
-
   for (auto& mock_controller : states.mock_controllers) {
     EXPECT_CALL(*mock_controller, MakeDecision(NetworkMetricsIs(check), _));
   }
   states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
   states.audio_network_adaptor->GetEncoderRuntimeConfig();
 
   check.uplink_packet_loss_fraction = rtc::Optional<float>(kPacketLoss);
   for (auto& mock_controller : states.mock_controllers) {
     EXPECT_CALL(*mock_controller, MakeDecision(NetworkMetricsIs(check), _));
   }
   states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
   states.audio_network_adaptor->GetEncoderRuntimeConfig();
+
+  check.rtt_ms = rtc::Optional<int>(kRtt);
+  for (auto& mock_controller : states.mock_controllers) {
+    EXPECT_CALL(*mock_controller, MakeDecision(NetworkMetricsIs(check), _));
+  }
+  states.audio_network_adaptor->SetRtt(kRtt);
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+
+  check.target_audio_bitrate_bps = rtc::Optional<int>(kTargetAudioBitrate);
+  for (auto& mock_controller : states.mock_controllers) {
+    EXPECT_CALL(*mock_controller, MakeDecision(NetworkMetricsIs(check), _));
+  }
+  states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
 }
 
 TEST(AudioNetworkAdaptorImplTest, SetConstraintsIsCalledOnSetFrameLengthRange) {
   auto states = CreateAudioNetworkAdaptor();
 
   for (auto& mock_controller : states.mock_controllers) {
     EXPECT_CALL(*mock_controller,
                 SetConstraints(ConstraintsReceiverFrameLengthRangeIs(
                     Controller::Constraints::FrameLengthRange(20, 120))));
   }
@@ skipping 17 matching lines @@
   states.audio_network_adaptor->GetEncoderRuntimeConfig();
 }
 
 TEST(AudioNetworkAdaptorImplTest,
      DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
   auto states = CreateAudioNetworkAdaptor();
 
   constexpr int kBandwidth = 16000;
   constexpr float kPacketLoss = 0.7f;
   constexpr int kRtt = 100;
+  constexpr int kTargetAudioBitrate = 15000;
 
   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);
+
+  states.simulated_clock->AdvanceTimeMilliseconds(150);
+  timestamp_check += 150;
+  check.target_audio_bitrate_bps = rtc::Optional<int>(kTargetAudioBitrate);
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
 }
 
 }  // namespace webrtc