Allow an external audio processing module to be used in WebRTC

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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_ENGINE_FAKEWEBRTCVOICEENGINE_H_
 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
 
 #include <map>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/media/engine/webrtcvoe.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 namespace voe {
 class TransmitMixer;
 }  // namespace voe
 }  // namespace webrtc
 
 namespace cricket {
 
 #define WEBRTC_CHECK_CHANNEL(channel) \
   if (channels_.find(channel) == channels_.end()) return -1;
 
 #define WEBRTC_STUB(method, args) \
   int method args override { return 0; }
 
 #define WEBRTC_FUNC(method, args) int method args override
 
 class FakeWebRtcVoiceEngine : public webrtc::VoEBase {
  public:
   struct Channel {
     std::vector<webrtc::CodecInst> recv_codecs;
     size_t neteq_capacity = 0;
     bool neteq_fast_accelerate = false;
   };
 
-  explicit FakeWebRtcVoiceEngine(webrtc::AudioProcessing* apm,
-                                 webrtc::voe::TransmitMixer* transmit_mixer)
-      : apm_(apm), transmit_mixer_(transmit_mixer) {
-  }
+  explicit FakeWebRtcVoiceEngine(webrtc::voe::TransmitMixer* transmit_mixer)
+      : transmit_mixer_(transmit_mixer) {}
   ~FakeWebRtcVoiceEngine() override {
     RTC_CHECK(channels_.empty());
   }
 
   bool IsInited() const { return inited_; }
   int GetLastChannel() const { return last_channel_; }
   int GetNumChannels() const { return static_cast<int>(channels_.size()); }
   void set_fail_create_channel(bool fail_create_channel) {
     fail_create_channel_ = fail_create_channel;
   }
 
   WEBRTC_STUB(Release, ());
 
   // webrtc::VoEBase
   WEBRTC_STUB(RegisterVoiceEngineObserver, (
       webrtc::VoiceEngineObserver& observer));
   WEBRTC_STUB(DeRegisterVoiceEngineObserver, ());
   WEBRTC_FUNC(Init,
               (webrtc::AudioDeviceModule* adm,
                webrtc::AudioProcessing* audioproc,
                const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
                    decoder_factory)) {
     inited_ = true;
     return 0;
   }
   WEBRTC_FUNC(Terminate, ()) {
     inited_ = false;
     return 0;
   }
-  webrtc::AudioProcessing* audio_processing() override {
-    return apm_;
-  }
+
   webrtc::AudioDeviceModule* audio_device_module() override {
     return nullptr;
   }
   webrtc::voe::TransmitMixer* transmit_mixer() override {
     return transmit_mixer_;
   }
   WEBRTC_FUNC(CreateChannel, ()) {
     return CreateChannel(webrtc::VoEBase::ChannelConfig());
   }
   WEBRTC_FUNC(CreateChannel, (const webrtc::VoEBase::ChannelConfig& config)) {
@@ skipping 33 matching lines @@
     auto ch = channels_.find(last_channel_);
     RTC_CHECK(ch != channels_.end());
     return ch->second->neteq_fast_accelerate;
   }
 
  private:
   bool inited_ = false;
   int last_channel_ = -1;
   std::map<int, Channel*> channels_;
   bool fail_create_channel_ = false;
-  webrtc::AudioProcessing* apm_ = nullptr;
   webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_