Simplifying audio network adaptor by moving receiver frame length range to ctor.

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 33 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&, int min_receiver_frame_length_ms,
+      int max_receiver_frame_length_ms, 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 152 matching lines @@
   TestSetPacketLossRate(states.encoder.get(), I(1.00      , 0.18 + eps), 0.20);
   TestSetPacketLossRate(states.encoder.get(), I(0.18 - eps, 0.09 + eps), 0.10);
   TestSetPacketLossRate(states.encoder.get(), I(0.09 - eps, 0.04 + eps), 0.05);
   TestSetPacketLossRate(states.encoder.get(), I(0.04 - eps, 0.01 + eps), 0.01);
   TestSetPacketLossRate(states.encoder.get(), I(0.01 - eps, 0.00      ), 0.00);
   // clang-format on
 }
 
 TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetUplinkBandwidth) {
   auto states = CreateCodec(2);
-  printf("passed!\n");
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", 0, 0, 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) {
   auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", 0, 0, 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) {
   auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", 0, 0, 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) {
   auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+  states.encoder->EnableAudioNetworkAdaptor("", 0, 0, 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,
-     InvokeAudioNetworkAdaptorOnSetReceiverFrameLengthRange) {
-  auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
-
-  auto config = CreateEncoderRuntimeConfig();
-  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
-      .WillOnce(Return(config));
-
-  constexpr int kMinFrameLength = 10;
-  constexpr int kMaxFrameLength = 60;
-  EXPECT_CALL(**states.mock_audio_network_adaptor,
-              SetReceiverFrameLengthRange(kMinFrameLength, kMaxFrameLength));
-  states.encoder->SetReceiverFrameLengthRange(kMinFrameLength, kMaxFrameLength);
-
-  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