Bugfix that fixes the integration issue that cause WebRTC AEC mobile to fail

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  Copyright (c) 2012 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 261 matching lines @@
                          int output_sample_rate_hz,
                          int reverse_sample_rate_hz,
                          ChannelLayout input_layout,
                          ChannelLayout output_layout,
                          ChannelLayout reverse_layout) = 0;
 
   // Pass down additional options which don't have explicit setters. This
   // ensures the options are applied immediately.
   virtual void SetExtraOptions(const Config& config) = 0;
 
+  // TODO(peah): Remove after voice engine no longer requires it to resample
+  // the reverse stream to the forward rate.
+  virtual int input_sample_rate_hz() const = 0;

[the sun, 2015/12/15 22:23:18]

Note: as "luck" would have it, this method was never removed from
MockAudioProcessing so there's no need to add it.

+
   // TODO(ajm): Only intended for internal use. Make private and friend the
   // necessary classes?
   virtual int proc_sample_rate_hz() const = 0;
   virtual int proc_split_sample_rate_hz() const = 0;
   virtual int num_input_channels() const = 0;
   virtual int num_output_channels() const = 0;
   virtual int num_reverse_channels() const = 0;
 
   // Set to true when the output of AudioProcessing will be muted or in some
   // other way not used. Ideally, the captured audio would still be processed,
@@ skipping 652 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_