Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_state_unittest.cc

Index: webrtc/audio/audio_state_unittest.cc
diff --git a/webrtc/audio/audio_state_unittest.cc b/webrtc/audio/audio_state_unittest.cc
index 38485a88bf14665ce6b15d9a149cefec6513751f..7c34d75d006feb10901a5a5615f288506e0b786b 100644
--- a/webrtc/audio/audio_state_unittest.cc
+++ b/webrtc/audio/audio_state_unittest.cc
@@ -83,39 +83,6 @@ TEST(AudioStateTest, TypingNoiseDetected) {
   EXPECT_FALSE(audio_state->typing_noise_detected());
 }
 
-// Test that RecordedDataIsAvailable calls get to the original transport.
-TEST(AudioStateTest, RecordedAudioArrivesAtOriginalTransport) {
-  using testing::_;
 
-  ConfigHelper helper;
-  auto& voice_engine = helper.voice_engine();
-  auto device =
-      static_cast<MockAudioDeviceModule*>(voice_engine.audio_device_module());
-
-  AudioTransport* audio_transport_proxy = nullptr;
-  ON_CALL(*device, RegisterAudioCallback(_))
-      .WillByDefault(
-          testing::Invoke([&audio_transport_proxy](AudioTransport* transport) {
-            audio_transport_proxy = transport;
-            return 0;
-          }));
-
-  MockAudioTransport original_audio_transport;
-  ON_CALL(voice_engine, audio_transport())
-      .WillByDefault(testing::Return(&original_audio_transport));
-
-  EXPECT_CALL(voice_engine, audio_device_module());
-  std::unique_ptr<internal::AudioState> audio_state(
-      new internal::AudioState(helper.config()));
-
-  // Setup completed. Ensure call of old transport is forwarded to new.
-  uint32_t new_mic_level;
-  EXPECT_CALL(original_audio_transport,
-              RecordedDataIsAvailable(nullptr, 80, 2, 1, 8000, 0, 0, 0, false,
-                                      testing::Ref(new_mic_level)));
-
-  audio_transport_proxy->RecordedDataIsAvailable(nullptr, 80, 2, 1, 8000, 0, 0,
-                                                 0, false, new_mic_level);
-}
 }  // namespace test
 }  // namespace webrtc