Allow an external audio processing module to be used in WebRTC

webrtc/audio/audio_state_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.
  */
 
 #include <memory>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 const int kSampleRate = 8000;
 const int kNumberOfChannels = 1;
 const int kBytesPerSample = 2;
 
 struct ConfigHelper {
   ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
     EXPECT_CALL(mock_voice_engine, RegisterVoiceEngineObserver(testing::_))
         .WillOnce(testing::Return(0));
     EXPECT_CALL(mock_voice_engine, DeRegisterVoiceEngineObserver())
         .WillOnce(testing::Return(0));
     EXPECT_CALL(mock_voice_engine, audio_device_module())
         .Times(testing::AtLeast(1));
-    EXPECT_CALL(mock_voice_engine, audio_processing())
-        .Times(testing::AtLeast(1));
     EXPECT_CALL(mock_voice_engine, audio_transport())
         .WillRepeatedly(testing::Return(&audio_transport));
 
     auto device = static_cast<MockAudioDeviceModule*>(
         voice_engine().audio_device_module());
 
     // Populate the audio transport proxy pointer to the most recent
     // transport connected to the Audio Device.
     ON_CALL(*device, RegisterAudioCallback(testing::_))
         .WillByDefault(testing::Invoke([this](AudioTransport* transport) {
           registered_audio_transport = transport;
           return 0;
         }));
 
     audio_state_config.voice_engine = &mock_voice_engine;
     audio_state_config.audio_mixer = audio_mixer;
+    audio_state_config.audio_processing =
+        new rtc::RefCountedObject<MockAudioProcessing>();
   }
   AudioState::Config& config() { return audio_state_config; }
   MockVoiceEngine& voice_engine() { return mock_voice_engine; }
   rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
   MockAudioTransport& original_audio_transport() { return audio_transport; }
   AudioTransport* audio_transport_proxy() { return registered_audio_transport; }
 
  private:
   testing::StrictMock<MockVoiceEngine> mock_voice_engine;
   AudioState::Config audio_state_config;
@@ skipping 103 matching lines @@
   int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
   size_t n_samples_out;
   int64_t elapsed_time_ms;
   int64_t ntp_time_ms;
   helper.audio_transport_proxy()->NeedMorePlayData(
       kSampleRate / 100, kBytesPerSample, kNumberOfChannels, kSampleRate,
       audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc