Allow an external audio processing module to be used in WebRTC

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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.
  */
 #include "webrtc/video/video_quality_test.h"
@@ skipping 61 matching lines @@
         base(nullptr),
         send_channel_id(-1),
         receive_channel_id(-1) {}
 
   webrtc::VoiceEngine* voice_engine;
   webrtc::VoEBase* base;
   int send_channel_id;
   int receive_channel_id;
 };
 
-void CreateVoiceEngine(VoiceEngineState* voe,
-                       rtc::scoped_refptr<webrtc::AudioDecoderFactory>
-                           decoder_factory) {
+void CreateVoiceEngine(
+    VoiceEngineState* voe,
+    webrtc::AudioProcessing* apm,
+    rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory) {
   voe->voice_engine = webrtc::VoiceEngine::Create();
   voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine);
-  EXPECT_EQ(0, voe->base->Init(nullptr, nullptr, decoder_factory));
+  EXPECT_EQ(0, voe->base->Init(nullptr, apm, decoder_factory));
   webrtc::VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   voe->send_channel_id = voe->base->CreateChannel(config);
   EXPECT_GE(voe->send_channel_id, 0);
   voe->receive_channel_id = voe->base->CreateChannel();
   EXPECT_GE(voe->receive_channel_id, 0);
 }
 
 void DestroyVoiceEngine(VoiceEngineState* voe) {
   voe->base->DeleteChannel(voe->send_channel_id);
@@ skipping 1744 matching lines @@
 void VideoQualityTest::RunWithRenderers(const Params& params) {
   params_ = params;
   CheckParams();
 
   // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
   // match the full stack tests.
   Call::Config call_config(event_log_.get());
   call_config.bitrate_config = params_.call.call_bitrate_config;
 
   ::VoiceEngineState voe;
+  rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing(
+      webrtc::AudioProcessing::Create());
+
   if (params_.audio.enabled) {
-    CreateVoiceEngine(&voe, decoder_factory_);
+    CreateVoiceEngine(&voe, audio_processing.get(), decoder_factory_);
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe.voice_engine;
     audio_state_config.audio_mixer = AudioMixerImpl::Create();
+    audio_state_config.audio_processing = audio_processing;
     call_config.audio_state = AudioState::Create(audio_state_config);
   }
 
   CreateCalls(call_config, call_config);
 
   // TODO(minyue): consider if this is a good transport even for audio only
   // calls.
   test::LayerFilteringTransport send_transport(
       params.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
       params.video.selected_tl, params_.ss.selected_sl, payload_type_map_);
@@ skipping 145 matching lines @@
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 }  // namespace webrtc