Allow an external audio processing module to be used in WebRTC

webrtc/call/call_perf_tests.cc

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;