Index: webrtc/audio/audio_send_stream_unittest.cc |
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc |
index d5197b313ee6ecf1a95a93bddaaa35848ab4da11..e74c9ed1bf2533dc587009c2b52cfae9abd59903 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -16,6 +16,7 @@ |
#include "webrtc/audio/conversion.h" |
#include "webrtc/base/task_queue.h" |
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
#include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
#include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h" |
@@ -80,6 +81,7 @@ struct ConfigHelper { |
AudioState::Config config; |
config.voice_engine = &voice_engine_; |
+ config.audio_mixer = AudioMixerImpl::Create(); |
audio_state_ = AudioState::Create(config); |
SetupDefaultChannelProxy(); |