| Index: webrtc/call/call_unittest.cc
 | 
| diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
 | 
| index d981b0b013b6a59e14f897586b86eba45cdc02bc..2d75be5eda5a39058056e68c2a6bcc9e8bb6e082 100644
 | 
| --- a/webrtc/call/call_unittest.cc
 | 
| +++ b/webrtc/call/call_unittest.cc
 | 
| @@ -14,6 +14,7 @@
 | 
|  #include "webrtc/api/call/audio_state.h"
 | 
|  #include "webrtc/call.h"
 | 
|  #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 | 
| +#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 | 
|  #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 | 
|  #include "webrtc/test/gtest.h"
 | 
|  #include "webrtc/test/mock_voice_engine.h"
 | 
| @@ -26,6 +27,7 @@ struct CallHelper {
 | 
|        : voice_engine_(decoder_factory) {
 | 
|      webrtc::AudioState::Config audio_state_config;
 | 
|      audio_state_config.voice_engine = &voice_engine_;
 | 
| +    audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
 | 
|      EXPECT_CALL(voice_engine_, audio_device_module());
 | 
|      EXPECT_CALL(voice_engine_, audio_processing());
 | 
|      EXPECT_CALL(voice_engine_, audio_transport());
 | 
| 
 |