Allow an external audio processing module to be used in WebRTC

webrtc/media/base/fakemediaengine.h

 /*
  *  Copyright (c) 2004 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.
  */
 
 #ifndef WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
 #define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
 
 #include <list>
 #include <map>
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/base/streamparams.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 
 using webrtc::RtpExtension;
 
 namespace cricket {
 
 class FakeMediaEngine;
 class FakeVideoEngine;
 class FakeVoiceEngine;
 
@@ skipping 727 matching lines @@
   RtpCapabilities capabilities_;
 };
 
 class FakeVoiceEngine : public FakeBaseEngine {
  public:
   FakeVoiceEngine(webrtc::AudioDeviceModule* adm,
                   const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
                       audio_encoder_factory,
                   const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
                       audio_decoder_factory,
-                  rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
+                  rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+                  rtc::scoped_refptr<webrtc::AudioProcessing> apm) {
     // Add a fake audio codec. Note that the name must not be "" as there are
     // sanity checks against that.
     codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1));
   }
   void Init() {}
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
     return rtc::scoped_refptr<webrtc::AudioState>();
   }
 
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
@@ skipping 90 matching lines @@
   friend class FakeMediaEngine;
 };
 
 class FakeMediaEngine :
     public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
  public:
   FakeMediaEngine()
       : CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine>(nullptr,
                                                                nullptr,
                                                                nullptr,
+                                                               nullptr,
                                                                nullptr) {}
   virtual ~FakeMediaEngine() {}
 
   void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
     voice_.SetCodecs(codecs);
   }
   void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
     video_.SetCodecs(codecs);
   }
 
@@ skipping 73 matching lines @@
   virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
 
  private:
   std::vector<FakeDataMediaChannel*> channels_;
   std::vector<DataCodec> data_codecs_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_