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()); |