Remove the VoiceEngineObserver callback interface.

audio/audio_state_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 "audio/audio_state.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/audio_processing/include/mock_audio_processing.h"
 #include "test/gtest.h"
 #include "test/mock_voice_engine.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 const int kSampleRate = 8000;
 const int kNumberOfChannels = 1;
 const int kBytesPerSample = 2;
 
 struct ConfigHelper {
   ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
-    EXPECT_CALL(mock_voice_engine, RegisterVoiceEngineObserver(testing::_))
-        .WillOnce(testing::Return(0));
-    EXPECT_CALL(mock_voice_engine, DeRegisterVoiceEngineObserver())
-        .WillOnce(testing::Return(0));
     EXPECT_CALL(mock_voice_engine, audio_device_module())
         .Times(testing::AtLeast(1));
     EXPECT_CALL(mock_voice_engine, audio_transport())
         .WillRepeatedly(testing::Return(&audio_transport));
 
     auto device = static_cast<MockAudioDeviceModule*>(
         voice_engine().audio_device_module());
 
     // Populate the audio transport proxy pointer to the most recent
     // transport connected to the Audio Device.
@@ skipping 51 matching lines @@
       new internal::AudioState(helper.config()));
 }
 
 TEST(AudioStateTest, GetVoiceEngine) {
   ConfigHelper helper;
   std::unique_ptr<internal::AudioState> audio_state(
       new internal::AudioState(helper.config()));
   EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine());
 }
 
-TEST(AudioStateTest, TypingNoiseDetected) {
-  ConfigHelper helper;
-  std::unique_ptr<internal::AudioState> audio_state(
-      new internal::AudioState(helper.config()));
-  VoiceEngineObserver* voe_observer =
-      static_cast<VoiceEngineObserver*>(audio_state.get());
-  EXPECT_FALSE(audio_state->typing_noise_detected());
-
-  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
-  EXPECT_FALSE(audio_state->typing_noise_detected());
-
-  voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
-  EXPECT_TRUE(audio_state->typing_noise_detected());
-  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
-  EXPECT_TRUE(audio_state->typing_noise_detected());
-
-  voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
-  EXPECT_FALSE(audio_state->typing_noise_detected());
-  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
-  EXPECT_FALSE(audio_state->typing_noise_detected());
-}
-
 // Test that RecordedDataIsAvailable calls get to the original transport.
 TEST(AudioStateAudioPathTest, RecordedAudioArrivesAtOriginalTransport) {
   ConfigHelper helper;
 
   rtc::scoped_refptr<AudioState> audio_state =
       AudioState::Create(helper.config());
 
   // Setup completed. Ensure call of original transport is forwarded to new.
   uint32_t new_mic_level;
   EXPECT_CALL(
@@ skipping 30 matching lines @@
   int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
   size_t n_samples_out;
   int64_t elapsed_time_ms;
   int64_t ntp_time_ms;
   helper.audio_transport_proxy()->NeedMorePlayData(
       kSampleRate / 100, kBytesPerSample, kNumberOfChannels, kSampleRate,
       audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc