| Index: webrtc/test/call_test.cc
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| diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
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| index ddff55256532a427d5b79cc2b9f7895dc36d1080..28971af986875483801a40c66aca05323a413375 100644
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| --- a/webrtc/test/call_test.cc
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| +++ b/webrtc/test/call_test.cc
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| @@ -15,6 +15,8 @@
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|  #include "webrtc/base/checks.h"
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|  #include "webrtc/config.h"
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|  #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
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| +#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
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| +#include "webrtc/test/call_test.h"
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|  #include "webrtc/test/testsupport/fileutils.h"
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|  #include "webrtc/voice_engine/include/voe_base.h"
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|  
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| @@ -52,6 +54,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
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|      CreateVoiceEngines();
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|      AudioState::Config audio_state_config;
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|      audio_state_config.voice_engine = voe_send_.voice_engine;
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| +    audio_state_config.audio_mixer = AudioMixerImpl::Create();
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|      send_config.audio_state = AudioState::Create(audio_state_config);
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|    }
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|    CreateSenderCall(send_config);
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| @@ -60,6 +63,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
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|      if (num_audio_streams_ > 0) {
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|        AudioState::Config audio_state_config;
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|        audio_state_config.voice_engine = voe_recv_.voice_engine;
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| +      audio_state_config.audio_mixer = AudioMixerImpl::Create();
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|        recv_config.audio_state = AudioState::Create(audio_state_config);
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|      }
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|      CreateReceiverCall(recv_config);
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| 
 |