Allow an external audio processing module to be used in WebRTC

webrtc/call/call_perf_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 127 matching lines @@
 void CallPerfTest::TestAudioVideoSync(FecMode fec,
                                       CreateOrder create_first,
                                       float video_ntp_speed,
                                       float video_rtp_speed,
                                       float audio_rtp_speed) {
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   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);
   int recv_channel_id = voe_base->CreateChannel();
 
   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;
   CreateCalls(sender_config, receiver_config);
 
 
   VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
 
   FakeNetworkPipe::Config audio_net_config;
   audio_net_config.queue_delay_ms = 500;
@@ skipping 579 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc