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.
  */
 
 #include <memory>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 using ::testing::NiceMock;
 using ::testing::Return;
 
 namespace {
 
 const CodecInst kDefaultOpusSettings = {105, "opus", 48000, 960, 1, 32000};
 constexpr int64_t kInitialTimeUs = 12345678;
 
 AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderOpus::Config config;
   config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
   config.num_channels = codec_inst.channels;
   config.bitrate_bps = rtc::Optional<int>(codec_inst.rate);
   config.payload_type = codec_inst.pltype;
   config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
                                                 : AudioEncoderOpus::kAudio;
+  config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
   return config;
 }
 
 struct AudioEncoderOpusStates {
   std::shared_ptr<MockAudioNetworkAdaptor*> mock_audio_network_adaptor;
   std::unique_ptr<AudioEncoderOpus> encoder;
   std::unique_ptr<SimulatedClock> simulated_clock;
 };
 
 AudioEncoderOpusStates CreateCodec(size_t num_channels) {
@@ skipping 168 matching lines @@
   TestSetPacketLossRate(states.encoder.get(), I(0.22 + eps, 1.00      ), 0.20);
 
   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, SetReceiverFrameLengthRange) {
+  auto states = CreateCodec(2);
+  // Before calling to |SetReceiverFrameLengthRange|,
+  // |supported_frame_lengths_ms| should contain only the frame length being
+  // used.
+  using ::testing::ElementsAre;
+  EXPECT_THAT(states.encoder->supported_frame_lengths_ms(),
+              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) {
   auto states = CreateCodec(2);
-  printf("passed!\n");
   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));
@@ skipping 47 matching lines @@
 
   // 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