Passed AudioMixer to AudioState::Config.

webrtc/call/call_unittest.cc

Index: webrtc/call/call_unittest.cc
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index b5764421f028195f48b585430d9bc96da0925350..72b9c0d77a5642e8f3fc7d1fdf818da72319afda 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"
@@ -30,6 +31,7 @@ struct CallHelper {
     EXPECT_CALL(voice_engine_, audio_processing());
     EXPECT_CALL(voice_engine_, audio_transport());
     webrtc::Call::Config config(&event_log_);
+    audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();

[the sun, 2016/11/14 20:14:22]

Move up (line 29)

     config.audio_state = webrtc::AudioState::Create(audio_state_config);
     call_.reset(webrtc::Call::Create(config));
   }