Revert of Opus implementation of the AudioEncoderFactoryTemplate API

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 15 matching lines @@
 
 namespace webrtc {
 using ::testing::NiceMock;
 using ::testing::Return;
 
 namespace {
 
 const CodecInst kDefaultOpusSettings = {105, "opus", 48000, 960, 1, 32000};
 constexpr int64_t kInitialTimeUs = 12345678;
 
-AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst) {
-  AudioEncoderOpusConfig config;
+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 = codec_inst.rate;
-  config.application = config.num_channels == 1
-                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
-                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+  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;
 }
 
-AudioEncoderOpusConfig CreateConfigWithParameters(
+AudioEncoderOpus::Config CreateConfigWithParameters(
     const SdpAudioFormat::Parameters& params) {
-  const SdpAudioFormat format("opus", 48000, 2, params);
-  return *AudioEncoderOpusImpl::SdpToConfig(format);
+  SdpAudioFormat format("opus", 48000, 2, params);
+  return AudioEncoderOpus::CreateConfig(0, format);
 }
 
 struct AudioEncoderOpusStates {
   std::shared_ptr<MockAudioNetworkAdaptor*> mock_audio_network_adaptor;
   MockSmoothingFilter* mock_bitrate_smoother;
-  std::unique_ptr<AudioEncoderOpusImpl> encoder;
+  std::unique_ptr<AudioEncoderOpus> encoder;
   std::unique_ptr<rtc::ScopedFakeClock> fake_clock;
-  AudioEncoderOpusConfig config;
+  AudioEncoderOpus::Config config;
 };
 
 AudioEncoderOpusStates CreateCodec(size_t num_channels) {
   AudioEncoderOpusStates states;
   states.mock_audio_network_adaptor =
       std::make_shared<MockAudioNetworkAdaptor*>(nullptr);
   states.fake_clock.reset(new rtc::ScopedFakeClock());
   states.fake_clock->SetTimeMicros(kInitialTimeUs);
   std::weak_ptr<MockAudioNetworkAdaptor*> mock_ptr(
       states.mock_audio_network_adaptor);
-  AudioEncoderOpusImpl::AudioNetworkAdaptorCreator creator =
+  AudioEncoderOpus::AudioNetworkAdaptorCreator creator =
       [mock_ptr](const std::string&, RtcEventLog* event_log) {
         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;
       };
 
   CodecInst codec_inst = kDefaultOpusSettings;
   codec_inst.channels = num_channels;
   states.config = CreateConfig(codec_inst);
   std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
       new MockSmoothingFilter());
   states.mock_bitrate_smoother = bitrate_smoother.get();
 
-  states.encoder.reset(new AudioEncoderOpusImpl(
-      states.config, codec_inst.pltype, std::move(creator),
-      std::move(bitrate_smoother)));
+  states.encoder.reset(new AudioEncoderOpus(states.config, std::move(creator),
+                                            std::move(bitrate_smoother)));
   return states;
 }
 
 AudioEncoderRuntimeConfig 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;
   AudioEncoderRuntimeConfig 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 AudioEncoderOpusImpl* encoder,
+void CheckEncoderRuntimeConfig(const AudioEncoderOpus* encoder,
                                const AudioEncoderRuntimeConfig& 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());
 }
 
 // Create 10ms audio data blocks for a total packet size of "packet_size_ms".
 std::unique_ptr<test::AudioLoop> Create10msAudioBlocks(
-    const std::unique_ptr<AudioEncoderOpusImpl>& encoder,
+    const std::unique_ptr<AudioEncoderOpus>& encoder,
     int packet_size_ms) {
   const std::string file_name =
       test::ResourcePath("audio_coding/testfile32kHz", "pcm");
 
   std::unique_ptr<test::AudioLoop> speech_data(new test::AudioLoop());
   int audio_samples_per_ms =
       rtc::CheckedDivExact(encoder->SampleRateHz(), 1000);
   if (!speech_data->Init(
           file_name,
           packet_size_ms * audio_samples_per_ms *
               encoder->num_channels_to_encode(),
           10 * audio_samples_per_ms * encoder->num_channels_to_encode()))
     return nullptr;
   return speech_data;
 }
 
 }  // namespace
 
 TEST(AudioEncoderOpusTest, DefaultApplicationModeMono) {
   auto states = CreateCodec(1);
-  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
-            states.encoder->application());
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
 TEST(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
   auto states = CreateCodec(2);
-  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
-            states.encoder->application());
+  EXPECT_EQ(AudioEncoderOpus::kAudio, states.encoder->application());
 }
 
 TEST(AudioEncoderOpusTest, ChangeApplicationMode) {
   auto states = CreateCodec(2);
   EXPECT_TRUE(
       states.encoder->SetApplication(AudioEncoder::Application::kSpeech));
-  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
-            states.encoder->application());
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
 TEST(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
   auto states = CreateCodec(2);
 
   // Trigger a reset.
   states.encoder->Reset();
   // Verify that the mode is still kAudio.
-  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
-            states.encoder->application());
+  EXPECT_EQ(AudioEncoderOpus::kAudio, states.encoder->application());
 
   // Now change to kVoip.
   EXPECT_TRUE(
       states.encoder->SetApplication(AudioEncoder::Application::kSpeech));
-  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
-            states.encoder->application());
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 
   // Trigger a reset again.
   states.encoder->Reset();
   // Verify that the mode is still kVoip.
-  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
-            states.encoder->application());
+  EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
 TEST(AudioEncoderOpusTest, ToggleDtx) {
   auto states = CreateCodec(2);
   // Enable DTX
   EXPECT_TRUE(states.encoder->SetDtx(true));
   EXPECT_TRUE(states.encoder->GetDtx());
   // Turn off DTX.
   EXPECT_TRUE(states.encoder->SetDtx(false));
   EXPECT_FALSE(states.encoder->GetDtx());
@@ skipping 44 matching lines @@
 }
 
 // Sets the packet loss rate to each number in the vector in turn, and verifies
 // that the loss rate as reported by the encoder is |expected_return| for all
 // of them.
 void TestSetPacketLossRate(AudioEncoderOpusStates* states,
                            const std::vector<float>& losses,
                            float expected_return) {
   // |kSampleIntervalMs| is chosen to ease the calculation since
   // 0.9999 ^ 184198 = 1e-8. Which minimizes the effect of
-  // PacketLossFractionSmoother used in AudioEncoderOpusImpl.
+  // PacketLossFractionSmoother used in AudioEncoderOpus.
   constexpr int64_t kSampleIntervalMs = 184198;
   for (float loss : losses) {
     states->encoder->OnReceivedUplinkPacketLossFraction(loss);
     states->fake_clock->AdvanceTime(
         rtc::TimeDelta::FromMilliseconds(kSampleIntervalMs));
     EXPECT_FLOAT_EQ(expected_return, states->encoder->packet_loss_rate());
   }
 }
 
 }  // namespace
@@ skipping 193 matching lines @@
   // subtracted. The eventual codec rate should be bounded by |kMaxBitrateBps|.
   target_bitrate =
       kOverheadBytesPerPacket * 8 * packet_rate + kMaxBitrateBps + 1;
   states.encoder->OnReceivedUplinkBandwidth(target_bitrate,
                                             rtc::Optional<int64_t>());
   EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
 }
 
 // Verifies that the complexity adaptation in the config works as intended.
 TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
-  AudioEncoderOpusConfig config;
+  AudioEncoderOpus::Config config;
   config.low_rate_complexity = 8;
   config.complexity = 6;
 
   // Bitrate within hysteresis window. Expect empty output.
-  config.bitrate_bps = 12500;
-  EXPECT_EQ(rtc::Optional<int>(),
-            AudioEncoderOpusImpl::GetNewComplexity(config));
+  config.bitrate_bps = rtc::Optional<int>(12500);
+  EXPECT_EQ(rtc::Optional<int>(), config.GetNewComplexity());
 
   // Bitrate below hysteresis window. Expect higher complexity.
-  config.bitrate_bps = 10999;
-  EXPECT_EQ(rtc::Optional<int>(8),
-            AudioEncoderOpusImpl::GetNewComplexity(config));
+  config.bitrate_bps = rtc::Optional<int>(10999);
+  EXPECT_EQ(rtc::Optional<int>(8), config.GetNewComplexity());
 
   // Bitrate within hysteresis window. Expect empty output.
-  config.bitrate_bps = 12500;
-  EXPECT_EQ(rtc::Optional<int>(),
-            AudioEncoderOpusImpl::GetNewComplexity(config));
+  config.bitrate_bps = rtc::Optional<int>(12500);
+  EXPECT_EQ(rtc::Optional<int>(), config.GetNewComplexity());
 
   // Bitrate above hysteresis window. Expect lower complexity.
-  config.bitrate_bps = 14001;
-  EXPECT_EQ(rtc::Optional<int>(6),
-            AudioEncoderOpusImpl::GetNewComplexity(config));
+  config.bitrate_bps = rtc::Optional<int>(14001);
+  EXPECT_EQ(rtc::Optional<int>(6), config.GetNewComplexity());
 }
 
 TEST(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
   auto states = CreateCodec(2);
   states.encoder->EnableAudioNetworkAdaptor("", nullptr);
 
   auto config = CreateEncoderRuntimeConfig();
   AudioEncoderRuntimeConfig empty_config;
 
   EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
@@ skipping 61 matching lines @@
 
     // Should encode now.
     states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
                            &encoded);
     EXPECT_GT(encoded.size(), 0u);
     encoded.Clear();
   }
 }
 
 TEST(AudioEncoderOpusTest, TestConfigDefaults) {
-  const auto config_opt = AudioEncoderOpusImpl::SdpToConfig({"opus", 48000, 2});
-  ASSERT_TRUE(config_opt);
-  EXPECT_EQ(48000, config_opt->max_playback_rate_hz);
-  EXPECT_EQ(1u, config_opt->num_channels);
-  EXPECT_FALSE(config_opt->fec_enabled);
-  EXPECT_FALSE(config_opt->dtx_enabled);
-  EXPECT_EQ(20, config_opt->frame_size_ms);
+  const AudioEncoderOpus::Config config =
+      AudioEncoderOpus::CreateConfig(0, {"opus", 48000, 2});
+
+  EXPECT_EQ(48000, config.max_playback_rate_hz);
+  EXPECT_EQ(1u, config.num_channels);
+  EXPECT_FALSE(config.fec_enabled);
+  EXPECT_FALSE(config.dtx_enabled);
+  EXPECT_EQ(20, config.frame_size_ms);
 }
 
 TEST(AudioEncoderOpusTest, TestConfigFromParams) {
-  const auto config1 = CreateConfigWithParameters({{"stereo", "0"}});
-  EXPECT_EQ(1U, config1.num_channels);
+  AudioEncoderOpus::Config config;
 
-  const auto config2 = CreateConfigWithParameters({{"stereo", "1"}});
-  EXPECT_EQ(2U, config2.num_channels);
+  config = CreateConfigWithParameters({{"stereo", "0"}});
+  EXPECT_EQ(1U, config.num_channels);
 
-  const auto config3 = CreateConfigWithParameters({{"useinbandfec", "0"}});
-  EXPECT_FALSE(config3.fec_enabled);
+  config = CreateConfigWithParameters({{"stereo", "1"}});
+  EXPECT_EQ(2U, config.num_channels);
 
-  const auto config4 = CreateConfigWithParameters({{"useinbandfec", "1"}});
-  EXPECT_TRUE(config4.fec_enabled);
+  config = CreateConfigWithParameters({{"useinbandfec", "0"}});
+  EXPECT_FALSE(config.fec_enabled);
 
-  const auto config5 = CreateConfigWithParameters({{"usedtx", "0"}});
-  EXPECT_FALSE(config5.dtx_enabled);
+  config = CreateConfigWithParameters({{"useinbandfec", "1"}});
+  EXPECT_TRUE(config.fec_enabled);
 
-  const auto config6 = CreateConfigWithParameters({{"usedtx", "1"}});
-  EXPECT_TRUE(config6.dtx_enabled);
+  config = CreateConfigWithParameters({{"usedtx", "0"}});
+  EXPECT_FALSE(config.dtx_enabled);
 
-  const auto config7 = CreateConfigWithParameters({{"cbr", "0"}});
-  EXPECT_FALSE(config7.cbr_enabled);
+  config = CreateConfigWithParameters({{"usedtx", "1"}});
+  EXPECT_TRUE(config.dtx_enabled);
 
-  const auto config8 = CreateConfigWithParameters({{"cbr", "1"}});
-  EXPECT_TRUE(config8.cbr_enabled);
+  config = CreateConfigWithParameters({{"cbr", "0"}});
+  EXPECT_FALSE(config.cbr_enabled);
 
-  const auto config9 =
-      CreateConfigWithParameters({{"maxplaybackrate", "12345"}});
-  EXPECT_EQ(12345, config9.max_playback_rate_hz);
+  config = CreateConfigWithParameters({{"cbr", "1"}});
+  EXPECT_TRUE(config.cbr_enabled);
 
-  const auto config10 =
-      CreateConfigWithParameters({{"maxaveragebitrate", "96000"}});
-  EXPECT_EQ(96000, config10.bitrate_bps);
+  config = CreateConfigWithParameters({{"maxplaybackrate", "12345"}});
+  EXPECT_EQ(12345, config.max_playback_rate_hz);
 
-  const auto config11 = CreateConfigWithParameters({{"maxptime", "40"}});
-  for (int frame_length : config11.supported_frame_lengths_ms) {
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "96000"}});
+  EXPECT_EQ(96000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxptime", "40"}});
+  for (int frame_length : config.supported_frame_lengths_ms) {
     EXPECT_LE(frame_length, 40);
   }
 
-  const auto config12 = CreateConfigWithParameters({{"minptime", "40"}});
-  for (int frame_length : config12.supported_frame_lengths_ms) {
+  config = CreateConfigWithParameters({{"minptime", "40"}});
+  for (int frame_length : config.supported_frame_lengths_ms) {
     EXPECT_GE(frame_length, 40);
   }
 
-  const auto config13 = CreateConfigWithParameters({{"ptime", "40"}});
-  EXPECT_EQ(40, config13.frame_size_ms);
+  config = CreateConfigWithParameters({{"ptime", "40"}});
+  EXPECT_EQ(40, config.frame_size_ms);
 
   constexpr int kMinSupportedFrameLength = 10;
   constexpr int kMaxSupportedFrameLength =
       WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
 
-  const auto config14 = CreateConfigWithParameters({{"ptime", "1"}});
-  EXPECT_EQ(kMinSupportedFrameLength, config14.frame_size_ms);
+  config = CreateConfigWithParameters({{"ptime", "1"}});
+  EXPECT_EQ(kMinSupportedFrameLength, config.frame_size_ms);
 
-  const auto config15 = CreateConfigWithParameters({{"ptime", "2000"}});
-  EXPECT_EQ(kMaxSupportedFrameLength, config15.frame_size_ms);
+  config = CreateConfigWithParameters({{"ptime", "2000"}});
+  EXPECT_EQ(kMaxSupportedFrameLength, config.frame_size_ms);
 }
 
 TEST(AudioEncoderOpusTest, TestConfigFromInvalidParams) {
   const webrtc::SdpAudioFormat format("opus", 48000, 2);
-  const auto default_config = *AudioEncoderOpusImpl::SdpToConfig(format);
+  const AudioEncoderOpus::Config default_config =
+      AudioEncoderOpus::CreateConfig(0, format);
 #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
   const std::vector<int> default_supported_frame_lengths_ms({20, 60, 120});
 #else
   const std::vector<int> default_supported_frame_lengths_ms({20, 60});
 #endif
 
-  AudioEncoderOpusConfig config;
+  AudioEncoderOpus::Config config;
   config = CreateConfigWithParameters({{"stereo", "invalid"}});
   EXPECT_EQ(default_config.num_channels, config.num_channels);
 
   config = CreateConfigWithParameters({{"useinbandfec", "invalid"}});
   EXPECT_EQ(default_config.fec_enabled, config.fec_enabled);
 
   config = CreateConfigWithParameters({{"usedtx", "invalid"}});
   EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
 
   config = CreateConfigWithParameters({{"cbr", "invalid"}});
@@ skipping 31 matching lines @@
   config = CreateConfigWithParameters({{"ptime", "invalid"}});
   EXPECT_EQ(default_supported_frame_lengths_ms,
             config.supported_frame_lengths_ms);
 }
 
 // Test that bitrate will be overridden by the "maxaveragebitrate" parameter.
 // Also test that the "maxaveragebitrate" can't be set to values outside the
 // range of 6000 and 510000
 TEST(AudioEncoderOpusTest, SetSendCodecOpusMaxAverageBitrate) {
   // Ignore if less than 6000.
-  const auto config1 = AudioEncoderOpusImpl::SdpToConfig(
-      {"opus", 48000, 2, {{"maxaveragebitrate", "5999"}}});
-  EXPECT_EQ(6000, config1->bitrate_bps);
+  const AudioEncoderOpus::Config config1 = AudioEncoderOpus::CreateConfig(
+      0, {"opus", 48000, 2, {{"maxaveragebitrate", "5999"}}});
+  EXPECT_EQ(6000, config1.bitrate_bps);
 
   // Ignore if larger than 510000.
-  const auto config2 = AudioEncoderOpusImpl::SdpToConfig(
-      {"opus", 48000, 2, {{"maxaveragebitrate", "510001"}}});
-  EXPECT_EQ(510000, config2->bitrate_bps);
+  const AudioEncoderOpus::Config config2 = AudioEncoderOpus::CreateConfig(
+      0, {"opus", 48000, 2, {{"maxaveragebitrate", "510001"}}});
+  EXPECT_EQ(510000, config2.bitrate_bps);
 
-  const auto config3 = AudioEncoderOpusImpl::SdpToConfig(
-      {"opus", 48000, 2, {{"maxaveragebitrate", "200000"}}});
-  EXPECT_EQ(200000, config3->bitrate_bps);
+  const AudioEncoderOpus::Config config3 = AudioEncoderOpus::CreateConfig(
+      0, {"opus", 48000, 2, {{"maxaveragebitrate", "200000"}}});
+  EXPECT_EQ(200000, config3.bitrate_bps);
 }
 
 // Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
 TEST(AudioEncoderOpusTest, SetMaxPlaybackRateNb) {
   auto config = CreateConfigWithParameters({{"maxplaybackrate", "8000"}});
   EXPECT_EQ(8000, config.max_playback_rate_hz);
   EXPECT_EQ(12000, config.bitrate_bps);
 
   config = CreateConfigWithParameters({{"maxplaybackrate", "8000"},
                                        {"stereo", "1"}});
@@ skipping 43 matching lines @@
   EXPECT_EQ(24001, config.max_playback_rate_hz);
   EXPECT_EQ(32000, config.bitrate_bps);
 
   config = CreateConfigWithParameters({{"maxplaybackrate", "24001"},
                                        {"stereo", "1"}});
   EXPECT_EQ(24001, config.max_playback_rate_hz);
   EXPECT_EQ(64000, config.bitrate_bps);
 }
 
 }  // namespace webrtc