Revert of Enable -Winconsistent-missing-override flag.

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

 /*
  *  Copyright (c) 2014 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 590 matching lines @@
 class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
  protected:
   static const int kNumPackets = 500;
   static const int kNumPullCalls = 500;
 
   AcmIsacMtTestOldApi()
       : AudioCodingModuleMtTestOldApi(), last_packet_number_(0) {}
 
   ~AcmIsacMtTestOldApi() {}
 
-  void SetUp() override {
+  void SetUp() {
     AudioCodingModuleTestOldApi::SetUp();
     RegisterCodec();  // Must be called before the threads start below.
 
     // Set up input audio source to read from specified file, loop after 5
     // seconds, and deliver blocks of 10 ms.
     const std::string input_file_name =
         webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
     audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
 
     // Generate one packet to have something to insert.
@@ skipping 13 matching lines @@
     static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
     AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
     codec_.pltype = kPayloadType;
 
     // Register iSAC codec in ACM, effectively unregistering the PCM16B codec
     // registered in AudioCodingModuleTestOldApi::SetUp();
     ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
     ASSERT_EQ(0, acm_->RegisterSendCodec(codec_));
   }
 
-  void InsertPacket() override {
+  void InsertPacket() {
     int num_calls = packet_cb_.num_calls();  // Store locally for thread safety.
     if (num_calls > last_packet_number_) {
       // Get the new payload out from the callback handler.
       // Note that since we swap buffers here instead of directly inserting
       // a pointer to the data in |packet_cb_|, we avoid locking the callback
       // for the duration of the IncomingPacket() call.
       packet_cb_.SwapBuffers(&last_payload_vec_);
       ASSERT_GT(last_payload_vec_.size(), 0u);
       rtp_utility_->Forward(&rtp_header_);
       last_packet_number_ = num_calls;
     }
     ASSERT_GT(last_payload_vec_.size(), 0u);
     ASSERT_EQ(
         0,
         acm_->IncomingPacket(
             &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
   }
 
-  void InsertAudio() override {
+  void InsertAudio() {
     // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS
     // this call confuses the number of samples with the number of bytes, and
     // ends up copying only half of what it should.
     memcpy(input_frame_.data_, audio_loop_.GetNextBlock().data(),
            kNumSamples10ms);
     AudioCodingModuleTestOldApi::InsertAudio();
   }
 
   // Override the verification function with no-op, since iSAC produces variable
   // payload sizes.
   void VerifyEncoding() override {}
 
   // This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but
   // here it is using the constants defined in this class (i.e., shorter test
   // run).
-  bool TestDone() override {
+  virtual bool TestDone() {
     if (packet_cb_.num_calls() > kNumPackets) {
       rtc::CritScope lock(&crit_sect_);
       if (pull_audio_count_ > kNumPullCalls) {
         // Both conditions for completion are met. End the test.
         return true;
       }
     }
     return false;
   }
 
@@ skipping 30 matching lines @@
         codec_registered_(false),
         receive_packet_count_(0),
         next_insert_packet_time_ms_(0),
         fake_clock_(new SimulatedClock(0)) {
     AudioEncoderIsac::Config config;
     config.payload_type = kPayloadType;
     isac_encoder_.reset(new AudioEncoderIsac(config));
     clock_ = fake_clock_.get();
   }
 
-  void SetUp() override {
+  void SetUp() {
     AudioCodingModuleTestOldApi::SetUp();
     // Set up input audio source to read from specified file, loop after 5
     // seconds, and deliver blocks of 10 ms.
     const std::string input_file_name =
         webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
     audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
     RegisterCodec();  // Must be called before the threads start below.
     StartThreads();
   }
 
   void RegisterCodec() override {
     static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
     AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
     codec_.pltype = kPayloadType;
 
     // Register iSAC codec in ACM, effectively unregistering the PCM16B codec
     // registered in AudioCodingModuleTestOldApi::SetUp();
     // Only register the decoder for now. The encoder is registered later.
     ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
   }
 
   void StartThreads() {
     receive_thread_.Start();
     receive_thread_.SetPriority(rtc::kRealtimePriority);
     codec_registration_thread_.Start();
     codec_registration_thread_.SetPriority(rtc::kRealtimePriority);
   }
 
-  void TearDown() override {
+  void TearDown() {
     AudioCodingModuleTestOldApi::TearDown();
     receive_thread_.Stop();
     codec_registration_thread_.Stop();
   }
 
   EventTypeWrapper RunTest() {
     return test_complete_->Wait(10 * 60 * 1000);  // 10 minutes' timeout.
   }
 
   static bool CbReceiveThread(void* context) {
@@ skipping 959 matching lines @@
     if (num_packets_++ > kTestNumPackets) {
       EXPECT_TRUE(has_toggled_);
       return NULL;  // Test ended.
     }
 
     // Get the next packet from the source.
     return packet_source_.NextPacket();
   }
 
   // Inherited from test::AudioSink.
-  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+  bool WriteArray(const int16_t* audio, size_t num_samples) {
     // Skip checking the first output frame, since it has a number of zeros
     // due to how NetEq is initialized.
     if (first_output_) {
       first_output_ = false;
       return true;
     }
     for (size_t i = 0; i < num_samples; ++i) {
       EXPECT_EQ(kSampleValue, audio[i]);
     }
     if (num_samples ==
@@ skipping 28 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc