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

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 56 matching lines @@
       webrtc::AudioProcessing::ChannelLayout input_layout,
       webrtc::AudioProcessing::ChannelLayout output_layout,
       webrtc::AudioProcessing::ChannelLayout reverse_layout));
   WEBRTC_STUB(Initialize, (
       const webrtc::ProcessingConfig& processing_config));
 
   WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
     experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
   }
 
+  WEBRTC_STUB_CONST(input_sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
   WEBRTC_STUB_CONST(num_input_channels, ());
   WEBRTC_STUB_CONST(num_output_channels, ());
   WEBRTC_STUB_CONST(num_reverse_channels, ());
   WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
   WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
   WEBRTC_STUB(ProcessStream, (
       const float* const* src,
       size_t samples_per_channel,
@@ skipping 726 matching lines @@
   int playout_fail_channel_;
   int send_fail_channel_;
   int recording_sample_rate_;
   int playout_sample_rate_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_