Replace a few calls to VoEHardware with direct calls on the ADM, in WVoMC.

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.
  */
 
@@ skipping 438 matching lines @@
 
   // webrtc::VoEHardware
   WEBRTC_STUB(GetNumOfRecordingDevices, (int& num));
   WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num));
   WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid));
   WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
   WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
   WEBRTC_STUB(SetPlayoutDevice, (int));
   WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
   WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
-  WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) {
-    recording_sample_rate_ = samples_per_sec;
-    return 0;
-  }
-  WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) {
-    *samples_per_sec = recording_sample_rate_;
-    return 0;
-  }
-  WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) {
-    playout_sample_rate_ = samples_per_sec;
-    return 0;
-  }
-  WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) {
-    *samples_per_sec = playout_sample_rate_;
-    return 0;
-  }
+  WEBRTC_STUB(SetRecordingSampleRate, (unsigned int samples_per_sec));
+  WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
+  WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
+  WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
   virtual bool BuiltInAECIsAvailable() const { return false; }
   WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
   virtual bool BuiltInAGCIsAvailable() const { return false; }
   WEBRTC_STUB(EnableBuiltInNS, (bool enable));
   virtual bool BuiltInNSIsAvailable() const { return false; }
 
   // webrtc::VoENetwork
   WEBRTC_FUNC(RegisterExternalTransport, (int channel,
                                           webrtc::Transport& transport)) {
@@ skipping 237 matching lines @@
   bool agc_enabled_ = false;
   bool highpass_filter_enabled_ = false;
   bool stereo_swapping_enabled_ = false;
   bool typing_detection_enabled_ = false;
   webrtc::EcModes ec_mode_ = webrtc::kEcDefault;
   webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   int playout_fail_channel_ = -1;
-  int recording_sample_rate_ = -1;
-  int playout_sample_rate_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_