Index: webrtc/audio/audio_receive_stream_unittest.cc |
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc |
index 84efb20b91a92b849d4a716617f42d3395346849..127ea077b0ef6002826104d5478f4b23c3193012 100644 |
--- a/webrtc/audio/audio_receive_stream_unittest.cc |
+++ b/webrtc/audio/audio_receive_stream_unittest.cc |
@@ -17,6 +17,7 @@ |
#include "webrtc/audio/conversion.h" |
#include "webrtc/call/rtp_stream_receiver_controller.h" |
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
+#include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h" |
#include "webrtc/modules/pacing/packet_router.h" |
#include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
@@ -74,13 +75,13 @@ struct ConfigHelper { |
RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); |
EXPECT_CALL(voice_engine_, |
DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); |
- EXPECT_CALL(voice_engine_, audio_processing()); |
EXPECT_CALL(voice_engine_, audio_device_module()); |
EXPECT_CALL(voice_engine_, audio_transport()); |
AudioState::Config config; |
config.voice_engine = &voice_engine_; |
config.audio_mixer = audio_mixer_; |
+ config.audio_processing = new rtc::RefCountedObject<MockAudioProcessing>(); |
audio_state_ = AudioState::Create(config); |
EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) |