Remove WVoE::SetAudioDeviceModule() - it is now set in ctor.

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 <list>
 #include <map>
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/engine/fakewebrtccommon.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_device/include/fake_audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace cricket {
 
 static const int kOpusBandwidthNb = 4000;
 static const int kOpusBandwidthMb = 6000;
 static const int kOpusBandwidthWb = 8000;
 static const int kOpusBandwidthSwb = 12000;
 static const int kOpusBandwidthFb = 20000;
 
@@ skipping 73 matching lines @@
   webrtc::VoiceDetection* voice_detection() const override { return NULL; }
 
   bool experimental_ns_enabled() {
     return experimental_ns_enabled_;
   }
 
  private:
   bool experimental_ns_enabled_;
 };
 
+// TODO(solenberg): Swap this for a proper mock of the ADM.
+class FakeAudioDeviceModule : public webrtc::FakeAudioDeviceModule {
+ public:
+  ~FakeAudioDeviceModule() override {
+    RTC_DCHECK_EQ(0, ref_count_);
+  }
+  int32_t AddRef() const override {
+    ref_count_++;
+    return ref_count_;
+  }
+  int32_t Release() const override {
+    RTC_DCHECK_LT(0, ref_count_);
+    ref_count_--;
+    return ref_count_;
+  }
+
+ private:
+  mutable int32_t ref_count_ = 0;
+};
+
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
       public webrtc::VoEHardware,
       public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
     explicit Channel()
         : external_transport(false),
@@ skipping 55 matching lines @@
         ec_mode_(webrtc::kEcDefault),
         aecm_mode_(webrtc::kAecmSpeakerphone),
         ns_mode_(webrtc::kNsDefault),
         agc_mode_(webrtc::kAgcDefault),
         observer_(NULL),
         playout_fail_channel_(-1),
         recording_sample_rate_(-1),
         playout_sample_rate_(-1) {
     memset(&agc_config_, 0, sizeof(agc_config_));
   }
-  ~FakeWebRtcVoiceEngine() {
+  ~FakeWebRtcVoiceEngine() override {
     RTC_CHECK(channels_.empty());
   }
 
   bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
 
   bool IsInited() const { return inited_; }
   int GetLastChannel() const { return last_channel_; }
   int GetNumChannels() const { return static_cast<int>(channels_.size()); }
   uint32_t GetLocalSSRC(int channel) {
     return channels_[channel]->send_ssrc;
@@ skipping 92 matching lines @@
     inited_ = true;
     return 0;
   }
   WEBRTC_FUNC(Terminate, ()) {
     inited_ = false;
     return 0;
   }
   webrtc::AudioProcessing* audio_processing() override {
     return &audio_processing_;
   }
+  webrtc::AudioDeviceModule* audio_device_module() override {
+    return &audio_device_module_;
+  }
   WEBRTC_FUNC(CreateChannel, ()) {
     webrtc::Config empty_config;
     return AddChannel(empty_config);
   }
   WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
     return AddChannel(config);
   }
   WEBRTC_FUNC(DeleteChannel, (int channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
     for (const auto& ch : channels_) {
@@ skipping 449 matching lines @@
   webrtc::EcModes ec_mode_;
   webrtc::AecmModes aecm_mode_;
   webrtc::NsModes ns_mode_;
   webrtc::AgcModes agc_mode_;
   webrtc::AgcConfig agc_config_;
   webrtc::VoiceEngineObserver* observer_;
   int playout_fail_channel_;
   int recording_sample_rate_;
   int playout_sample_rate_;
   FakeAudioProcessing audio_processing_;
+  FakeAudioDeviceModule audio_device_module_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_