Introduced new scheme for controlling the functionality inside the audio processing module

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 24 matching lines @@
 static const int kOpusBandwidthSwb = 12000;
 static const int kOpusBandwidthFb = 20000;
 
 #define WEBRTC_CHECK_CHANNEL(channel) \
   if (channels_.find(channel) == channels_.end()) return -1;
 
 class FakeAudioProcessing : public webrtc::AudioProcessing {
  public:
   FakeAudioProcessing() : experimental_ns_enabled_(false) {}
 
+  WEBRTC_BOOL_STUB(ApplyConfig, (const AudioProcessing::Config& config));

[the sun, 2016/09/01 13:48:00]

Use same order as in audio_processing.h

[peah-webrtc, 2016/09/02 08:21:59]

Good find!
Done.

   WEBRTC_STUB(Initialize, ())
   WEBRTC_STUB(Initialize, (
       int input_sample_rate_hz,
       int output_sample_rate_hz,
       int reverse_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout input_layout,
       webrtc::AudioProcessing::ChannelLayout output_layout,
       webrtc::AudioProcessing::ChannelLayout reverse_layout));
   WEBRTC_STUB(Initialize, (
       const webrtc::ProcessingConfig& processing_config));
@@ skipping 504 matching lines @@
   webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_