Index: webrtc/call/call_unittest.cc |
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc |
index 8f0a340e72074d65601b562fd337f7164a445088..5267e7a6eea706335625e76e3b0dffd38858d448 100644 |
--- a/webrtc/call/call_unittest.cc |
+++ b/webrtc/call/call_unittest.cc |
@@ -37,8 +37,8 @@ struct CallHelper { |
webrtc::AudioState::Config audio_state_config; |
audio_state_config.voice_engine = &voice_engine_; |
audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create(); |
+ audio_state_config.audio_processing = webrtc::AudioProcessing::Create(); |
EXPECT_CALL(voice_engine_, audio_device_module()); |
- EXPECT_CALL(voice_engine_, audio_processing()); |
EXPECT_CALL(voice_engine_, audio_transport()); |
webrtc::Call::Config config(&event_log_); |
config.audio_state = webrtc::AudioState::Create(audio_state_config); |
@@ -453,11 +453,13 @@ TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) { |
}; |
ScopedVoiceEngine voice_engine; |
- voice_engine.base->Init(&mock_adm); |
AudioState::Config audio_state_config; |
audio_state_config.voice_engine = voice_engine.voe; |
audio_state_config.audio_mixer = mock_mixer; |
+ audio_state_config.audio_processing = AudioProcessing::Create(); |
+ voice_engine.base->Init(&mock_adm, audio_state_config.audio_processing.get()); |
auto audio_state = AudioState::Create(audio_state_config); |
+ |
RtcEventLogNullImpl event_log; |
Call::Config call_config(&event_log); |
call_config.audio_state = audio_state; |