Adding audio network adaptor to AudioEncoderOpus.

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/gtest.h"
 
 namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
 
 namespace {
-const CodecInst kOpusSettings = {105, "opus", 48000, 960, 1, 32000};
+
+const CodecInst kDefaultOpusSettings = {105, "opus", 48000, 960, 1, 32000};
+
+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;
+  return config;
+}
+
+struct AudioEncoderOpusStates {
+  MockAudioNetworkAdaptor* mock_audio_network_adaptor = nullptr;
+  std::unique_ptr<AudioEncoderOpus> encoder;
+
+  explicit AudioEncoderOpusStates(size_t num_channels) {
+    CodecInst codec_inst = kDefaultOpusSettings;
+    codec_inst.channels = num_channels;
+    auto config = CreateConfig(codec_inst);
+    auto creator = [this](const std::string&, const Clock*) {
+      return this->CreateMockAudioNetworkAdaptor();
+    };
+    encoder.reset(new AudioEncoderOpus(config, creator));
+  }
+
+  std::unique_ptr<AudioNetworkAdaptor> CreateMockAudioNetworkAdaptor() {
+    std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
+        new NiceMock<MockAudioNetworkAdaptor>());
+    EXPECT_CALL(*adaptor, Die());
+    mock_audio_network_adaptor = adaptor.get();
+    return adaptor;
+  }

[kwiberg-webrtc, 2016/10/04 17:40:37]

Much better!

However, you could have kept the old structure with a struct and a free Create
function; all you'd have to do is put the lines that create the mock ANA in the
lambda, and set the lamda's capture list to [&states] so that it could write to
states.mock_audio_network_adaptor.

Keeping this structure is fine too.

[minyue-webrtc, 2016/10/04 20:31:03]

I prefer keeping it this way. I now can put
states.encoder->EnableAudioNetworkAdaptor("", nullptr);
in the test code, instead of in the creation of AudioEncoderOpusStates (with the
old define). This illustrates the need of EnableAudioNetworkAdaptor().

It may still be possible in the old structure, but can be trickier.

[kwiberg-webrtc, 2016/10/04 20:47:47]

You'd still get precisely that with my suggestion, which was to replace the
creator declaration in the previous patch set with this:

  AudioEncoderOpus::AudioNetworkAdaptorCreator creator = [&states](
      const std::string&, const Clock*) {
    std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
        new NiceMock<MockAudioNetworkAdaptor>());
    EXPECT_CALL(*adaptor, Die());
    states.mock_audio_network_adaptor = adaptor.get();
    return adaptor;
  };

I find the struct with no methods + free Create function version easier to
follow than the struct with methods version, but pick the one you like best. For
example the one that requires the least amount of rewriting... :-)

+};
+
+AudioNetworkAdaptor::EncoderRuntimeConfig CreateEncoderRuntimeConfig() {
+  constexpr int kBitrate = 40000;
+  constexpr int kFrameLength = 60;
+  constexpr bool kEnableFec = true;
+  constexpr bool kEnableDtx = false;
+  constexpr size_t kNumChannels = 1;
+  constexpr float kPacketLossFraction = 0.1f;
+  AudioNetworkAdaptor::EncoderRuntimeConfig config;
+  config.bitrate_bps = rtc::Optional<int>(kBitrate);
+  config.frame_length_ms = rtc::Optional<int>(kFrameLength);
+  config.enable_fec = rtc::Optional<bool>(kEnableFec);
+  config.enable_dtx = rtc::Optional<bool>(kEnableDtx);
+  config.num_channels = rtc::Optional<size_t>(kNumChannels);
+  config.uplink_packet_loss_fraction =
+      rtc::Optional<float>(kPacketLossFraction);
+  return config;
+}
+
+void CheckEncoderRuntimeConfig(
+    const AudioEncoderOpus* encoder,
+    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+  EXPECT_EQ(*config.bitrate_bps, encoder->GetTargetBitrate());
+  EXPECT_EQ(*config.frame_length_ms, encoder->next_frame_length_ms());
+  EXPECT_EQ(*config.enable_fec, encoder->fec_enabled());
+  EXPECT_EQ(*config.enable_dtx, encoder->GetDtx());
+  EXPECT_EQ(*config.num_channels, encoder->num_channels_to_encode());
+}
+
 }  // namespace
 
-class AudioEncoderOpusTest : public ::testing::Test {
- protected:
-  void CreateCodec(int num_channels) {
-    codec_inst_.channels = num_channels;
-    encoder_.reset(new AudioEncoderOpus(codec_inst_));
-    auto expected_app =
-        num_channels == 1 ? AudioEncoderOpus::kVoip : AudioEncoderOpus::kAudio;
-    EXPECT_EQ(expected_app, encoder_->application());
-  }
-
-  CodecInst codec_inst_ = kOpusSettings;
-  std::unique_ptr<AudioEncoderOpus> encoder_;
-};
-
-TEST_F(AudioEncoderOpusTest, DefaultApplicationModeMono) {
-  CreateCodec(1);
+TEST(AudioEncoderOpusTest, DefaultApplicationModeMono) {
+  AudioEncoderOpusStates states(1);
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
-TEST_F(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
-  CreateCodec(2);
+TEST(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
+  AudioEncoderOpusStates states(2);
+  EXPECT_EQ(AudioEncoderOpus::kAudio, states.encoder->application());
 }
 
-TEST_F(AudioEncoderOpusTest, ChangeApplicationMode) {
-  CreateCodec(2);
-  EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
-  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+TEST(AudioEncoderOpusTest, ChangeApplicationMode) {
+  AudioEncoderOpusStates states(2);
+  EXPECT_TRUE(
+      states.encoder->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
-TEST_F(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
-  CreateCodec(2);
+TEST(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
+  AudioEncoderOpusStates states(2);
 
   // Trigger a reset.
-  encoder_->Reset();
+  states.encoder->Reset();
   // Verify that the mode is still kAudio.
-  EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
+  EXPECT_EQ(AudioEncoderOpus::kAudio, states.encoder->application());
 
   // Now change to kVoip.
-  EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
-  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+  EXPECT_TRUE(
+      states.encoder->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 
   // Trigger a reset again.
-  encoder_->Reset();
+  states.encoder->Reset();
   // Verify that the mode is still kVoip.
-  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
-TEST_F(AudioEncoderOpusTest, ToggleDtx) {
-  CreateCodec(2);
+TEST(AudioEncoderOpusTest, ToggleDtx) {
+  AudioEncoderOpusStates states(2);
   // Enable DTX
-  EXPECT_TRUE(encoder_->SetDtx(true));
+  EXPECT_TRUE(states.encoder->SetDtx(true));
   // Verify that the mode is still kAudio.
-  EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
+  EXPECT_EQ(AudioEncoderOpus::kAudio, states.encoder->application());
   // Turn off DTX.
-  EXPECT_TRUE(encoder_->SetDtx(false));
+  EXPECT_TRUE(states.encoder->SetDtx(false));
 }
 
-TEST_F(AudioEncoderOpusTest, SetBitrate) {
-  CreateCodec(1);
-  // Constants are replicated from audio_encoder_opus.cc.
+TEST(AudioEncoderOpusTest, SetBitrate) {
+  AudioEncoderOpusStates states(1);
+  // Constants are replicated from audio_states.encoderopus.cc.
   const int kMinBitrateBps = 500;
   const int kMaxBitrateBps = 512000;
   // Set a too low bitrate.
-  encoder_->SetTargetBitrate(kMinBitrateBps - 1);
-  EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
+  states.encoder->SetTargetBitrate(kMinBitrateBps - 1);
+  EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
   // Set a too high bitrate.
-  encoder_->SetTargetBitrate(kMaxBitrateBps + 1);
-  EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
+  states.encoder->SetTargetBitrate(kMaxBitrateBps + 1);
+  EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
   // Set the minimum rate.
-  encoder_->SetTargetBitrate(kMinBitrateBps);
-  EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
+  states.encoder->SetTargetBitrate(kMinBitrateBps);
+  EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
   // Set the maximum rate.
-  encoder_->SetTargetBitrate(kMaxBitrateBps);
-  EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
+  states.encoder->SetTargetBitrate(kMaxBitrateBps);
+  EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
   // Set rates from 1000 up to 32000 bps.
   for (int rate = 1000; rate <= 32000; rate += 1000) {
-    encoder_->SetTargetBitrate(rate);
-    EXPECT_EQ(rate, encoder_->GetTargetBitrate());
+    states.encoder->SetTargetBitrate(rate);
+    EXPECT_EQ(rate, states.encoder->GetTargetBitrate());
   }
 }
 
 namespace {
 
 // Returns a vector with the n evenly-spaced numbers a, a + (b - a)/(n - 1),
 // ..., b.
 std::vector<double> IntervalSteps(double a, double b, size_t n) {
   RTC_DCHECK_GT(n, 1u);
   const double step = (b - a) / (n - 1);
@@ skipping 10 matching lines @@
                            const std::vector<double>& losses,
                            double expected_return) {
   for (double loss : losses) {
     encoder->SetProjectedPacketLossRate(loss);
     EXPECT_DOUBLE_EQ(expected_return, encoder->packet_loss_rate());
   }
 }
 
 }  // namespace
 
-TEST_F(AudioEncoderOpusTest, PacketLossRateOptimized) {
-  CreateCodec(1);
+TEST(AudioEncoderOpusTest, PacketLossRateOptimized) {
+  AudioEncoderOpusStates states(1);
   auto I = [](double a, double b) { return IntervalSteps(a, b, 10); };
   const double eps = 1e-15;
 
   // Note that the order of the following calls is critical.
 
   // clang-format off
-  TestSetPacketLossRate(encoder_.get(), I(0.00      , 0.01 - eps), 0.00);
-  TestSetPacketLossRate(encoder_.get(), I(0.01 + eps, 0.06 - eps), 0.01);
-  TestSetPacketLossRate(encoder_.get(), I(0.06 + eps, 0.11 - eps), 0.05);
-  TestSetPacketLossRate(encoder_.get(), I(0.11 + eps, 0.22 - eps), 0.10);
-  TestSetPacketLossRate(encoder_.get(), I(0.22 + eps, 1.00      ), 0.20);
+  TestSetPacketLossRate(states.encoder.get(), I(0.00      , 0.01 - eps), 0.00);
+  TestSetPacketLossRate(states.encoder.get(), I(0.01 + eps, 0.06 - eps), 0.01);
+  TestSetPacketLossRate(states.encoder.get(), I(0.06 + eps, 0.11 - eps), 0.05);
+  TestSetPacketLossRate(states.encoder.get(), I(0.11 + eps, 0.22 - eps), 0.10);
+  TestSetPacketLossRate(states.encoder.get(), I(0.22 + eps, 1.00      ), 0.20);
 
-  TestSetPacketLossRate(encoder_.get(), I(1.00      , 0.18 + eps), 0.20);
-  TestSetPacketLossRate(encoder_.get(), I(0.18 - eps, 0.09 + eps), 0.10);
-  TestSetPacketLossRate(encoder_.get(), I(0.09 - eps, 0.04 + eps), 0.05);
-  TestSetPacketLossRate(encoder_.get(), I(0.04 - eps, 0.01 + eps), 0.01);
-  TestSetPacketLossRate(encoder_.get(), I(0.01 - eps, 0.00      ), 0.00);
+  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) {
+  AudioEncoderOpusStates states(2);
+  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));
+  states.encoder->OnReceivedUplinkBandwidth(kUplinkBandwidth);
+
+  CheckEncoderRuntimeConfig(states.encoder.get(), config);
+}
+
+TEST(AudioEncoderOpusTest,
+     InvokeAudioNetworkAdaptorOnSetUplinkPacketLossFraction) {
+  AudioEncoderOpusStates states(2);
+  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 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) {
+  AudioEncoderOpusStates states(2);
+  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 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) {
+  AudioEncoderOpusStates states(2);
+  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 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) {
+  AudioEncoderOpusStates states(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);
+}
+
 }  // namespace webrtc