Allow an external audio processing module to be used in WebRTC

webrtc/call/call_unittest.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.
  */
 
@@ skipping 19 matching lines @@
 
 namespace {
 
 struct CallHelper {
   explicit CallHelper(
       rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
       : voice_engine_(decoder_factory) {
     webrtc::AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &voice_engine_;
     audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
+    audio_state_config.audio_processing = webrtc::AudioProcessing::Create();
     EXPECT_CALL(voice_engine_, audio_device_module());
-    EXPECT_CALL(voice_engine_, audio_processing());
     EXPECT_CALL(voice_engine_, audio_transport());
     webrtc::Call::Config config(&event_log_);
     config.audio_state = webrtc::AudioState::Create(audio_state_config);
     call_.reset(webrtc::Call::Create(config));
   }
 
   webrtc::Call* operator->() { return call_.get(); }
   webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; }
 
  private:
@@ skipping 394 matching lines @@
     ~ScopedVoiceEngine() {
       base->Release();
       EXPECT_TRUE(VoiceEngine::Delete(voe));
     }
 
     VoiceEngine* voe;
     VoEBase* base;
   };
   ScopedVoiceEngine voice_engine;
 
-  voice_engine.base->Init(&mock_adm);
   AudioState::Config audio_state_config;
   audio_state_config.voice_engine = voice_engine.voe;
   audio_state_config.audio_mixer = mock_mixer;
+  audio_state_config.audio_processing = AudioProcessing::Create();
+  voice_engine.base->Init(&mock_adm, audio_state_config.audio_processing.get());
   auto audio_state = AudioState::Create(audio_state_config);
+
   RtcEventLogNullImpl event_log;
   Call::Config call_config(&event_log);
   call_config.audio_state = audio_state;
   std::unique_ptr<Call> call(Call::Create(call_config));
 
   auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
     AudioSendStream::Config config(nullptr);
     config.rtp.ssrc = ssrc;
     config.voe_channel_id = voice_engine.base->CreateChannel();
     AudioSendStream* stream = call->CreateAudioSendStream(config);
@@ skipping 233 matching lines @@
   mask.min_bitrate_bps = rtc::Optional<int>(2000);
   EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000));
   call->SetBitrateConfigMask(mask);
 
   // Set min to 3000; the clamped value stays the same so nothing happens.
   mask.min_bitrate_bps = rtc::Optional<int>(3000);
   call->SetBitrateConfigMask(mask);
 }
 
 }  // namespace webrtc