Allow an external audio processing module to be used in WebRTC

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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 38 matching lines @@
 void CallTest::RunBaseTest(BaseTest* test) {
   num_video_streams_ = test->GetNumVideoStreams();
   num_audio_streams_ = test->GetNumAudioStreams();
   num_flexfec_streams_ = test->GetNumFlexfecStreams();
   RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
   Call::Config send_config(test->GetSenderCallConfig());
   if (num_audio_streams_ > 0) {
     CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
     test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
                                     fake_recv_audio_device_.get());
+    apm_send_ = AudioProcessing::Create();
+    apm_recv_ = AudioProcessing::Create();
     CreateVoiceEngines();
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe_send_.voice_engine;
     audio_state_config.audio_mixer = AudioMixerImpl::Create();
+    audio_state_config.audio_processing = apm_send_;
     send_config.audio_state = AudioState::Create(audio_state_config);
   }
   CreateSenderCall(send_config);
   if (test->ShouldCreateReceivers()) {
     Call::Config recv_config(test->GetReceiverCallConfig());
     if (num_audio_streams_ > 0) {
       AudioState::Config audio_state_config;
       audio_state_config.voice_engine = voe_recv_.voice_engine;
       audio_state_config.audio_mixer = AudioMixerImpl::Create();
+      audio_state_config.audio_processing = apm_recv_;
       recv_config.audio_state = AudioState::Create(audio_state_config);
     }
     CreateReceiverCall(recv_config);
   }
   test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
   receive_transport_.reset(test->CreateReceiveTransport());
   send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
 
   if (test->ShouldCreateReceivers()) {
     send_transport_->SetReceiver(receiver_call_->Receiver());
@@ skipping 289 matching lines @@
   for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
     receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream);
 
   video_receive_streams_.clear();
   allocated_decoders_.clear();
 }
 
 void CallTest::CreateVoiceEngines() {
   voe_send_.voice_engine = VoiceEngine::Create();
   voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
-  EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
-                                    decoder_factory_));
+  EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(),
+                                    apm_send_.get(), decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   voe_send_.channel_id = voe_send_.base->CreateChannel(config);
   EXPECT_GE(voe_send_.channel_id, 0);
 
   voe_recv_.voice_engine = VoiceEngine::Create();
   voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
-  EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr,
-                                    decoder_factory_));
+  EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(),
+                                    apm_recv_.get(), decoder_factory_));
   voe_recv_.channel_id = voe_recv_.base->CreateChannel();
   EXPECT_GE(voe_recv_.channel_id, 0);
 }
 
 void CallTest::DestroyVoiceEngines() {
   voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
   voe_recv_.channel_id = -1;
   voe_recv_.base->Release();
   voe_recv_.base = nullptr;
 
@@ skipping 141 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc