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 {
+  std::shared_ptr<MockAudioNetworkAdaptor*> mock_audio_network_adaptor;
+  std::unique_ptr<AudioEncoderOpus> encoder;
+};
+
+AudioEncoderOpusStates CreateCodec(size_t num_channels) {
+  AudioEncoderOpusStates states;
+  states.mock_audio_network_adaptor = std::shared_ptr<MockAudioNetworkAdaptor*>(
+      new MockAudioNetworkAdaptor*(nullptr));

[minyue-webrtc, 2016/10/05 14:18:24]

This is different from your suggestion. We have to allocate a
MockAudioNetworkAdaptor*

[kwiberg-webrtc, 2016/10/06 08:51:10]

Yes, of course. I tried to to that in my suggestion, but obviously I was
confused...

You should be able to do this with make_shared if you want, like so:

  states.mock_audio_network_adaptor =
      std::make_shared<MockAudioNetworkAdaptor*>(nullptr);

[minyue-webrtc, 2016/10/06 09:57:06]

yes, that reads better.

+
+  std::weak_ptr<MockAudioNetworkAdaptor*> mock_ptr(
+      states.mock_audio_network_adaptor);
+  AudioEncoderOpus::AudioNetworkAdaptorCreator creator = [mock_ptr](

[minyue-webrtc, 2016/10/05 14:18:24]

I see the point now. We don't use reference, we use a copy.

[kwiberg-webrtc, 2016/10/06 08:51:10]

Yes---we copy the weak_ptr. And the thing that it points to is allocated on the
heap and by construction safely accessible since we have a weak_ptr to it.

+      const std::string&, const Clock*) {
+    std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
+        new NiceMock<MockAudioNetworkAdaptor>());
+    EXPECT_CALL(*adaptor, Die());
+    *mock_ptr.lock() = adaptor.get();

[kwiberg-webrtc, 2016/10/06 08:51:10]

If all shared_ptrs (just one in this test) that pointed to the thing have
previously expired, .lock() will give you a null shared_ptr. This is never
supposed to happen in this test, but in order to fail more intelligibly and be
easier to read, the code should probably do something more verbose instead, like
this:

  if (auto sp = mock_ptr.lock()) {
    *sp = adaptor.get();
  } else {
    FAIL() << "ANA creator was called later than expected";
  }

[minyue-webrtc, 2016/10/06 09:57:06]

FAIL() does not work since it is only allowed in void function(). I'd use
RTC_NOTREACHED(). It also reads better since it is more of an error in writing a
test than error in using AudioEncoderOpus.

+    return adaptor;
+  };
+
+  CodecInst codec_inst = kDefaultOpusSettings;
+  codec_inst.channels = num_channels;
+  auto config = CreateConfig(codec_inst);
+  states.encoder.reset(new AudioEncoderOpus(config, std::move(creator)));
+  return states;
+}
+
+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) {
+  auto states = CreateCodec(1);
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
-TEST_F(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
-  CreateCodec(2);
+TEST(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
+  auto states = CreateCodec(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) {
+  auto states = CreateCodec(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) {
+  auto states = CreateCodec(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) {
+  auto states = CreateCodec(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) {
+  auto states = CreateCodec(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) {
+  auto states = CreateCodec(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) {
+  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));
+  states.encoder->OnReceivedUplinkBandwidth(kUplinkBandwidth);
+
+  CheckEncoderRuntimeConfig(states.encoder.get(), config);
+}
+
+TEST(AudioEncoderOpusTest,
+     InvokeAudioNetworkAdaptorOnSetUplinkPacketLossFraction) {
+  auto states = CreateCodec(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) {
+  auto states = CreateCodec(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) {
+  auto states = CreateCodec(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) {
+  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);
+}
+
 }  // namespace webrtc