Index: webrtc/call/call_perf_tests.cc |
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc |
index 3b1de73e65147cefbcdad5e75b6e5f86991687f4..0c0a0cfd2966e396327c79b38957a936e3efca2c 100644 |
--- a/webrtc/call/call_perf_tests.cc |
+++ b/webrtc/call/call_perf_tests.cc |
@@ -145,12 +145,15 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, |
const uint32_t kAudioRecvSsrc = 5678; |
metrics::Reset(); |
+ rtc::scoped_refptr<AudioProcessing> audio_processing = |
+ AudioProcessing::Create(); |
VoiceEngine* voice_engine = VoiceEngine::Create(); |
VoEBase* voe_base = VoEBase::GetInterface(voice_engine); |
FakeAudioDevice fake_audio_device( |
FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000), |
FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed); |
- EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_)); |
+ EXPECT_EQ(0, voe_base->Init(&fake_audio_device, audio_processing.get(), |
+ decoder_factory_)); |
VoEBase::ChannelConfig config; |
config.enable_voice_pacing = true; |
int send_channel_id = voe_base->CreateChannel(config); |
@@ -159,7 +162,9 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, |
AudioState::Config send_audio_state_config; |
send_audio_state_config.voice_engine = voice_engine; |
send_audio_state_config.audio_mixer = AudioMixerImpl::Create(); |
+ send_audio_state_config.audio_processing = audio_processing; |
Call::Config sender_config(event_log_.get()); |
+ |
sender_config.audio_state = AudioState::Create(send_audio_state_config); |
Call::Config receiver_config(event_log_.get()); |
receiver_config.audio_state = sender_config.audio_state; |