audio_processing: Adds two UMA histograms logging delay jumps in AEC

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 134 matching lines @@
   WEBRTC_STUB(set_stream_delay_ms, (int delay));
   WEBRTC_STUB_CONST(stream_delay_ms, ());
   WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
   WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
   WEBRTC_BOOL_STUB_CONST(stream_key_pressed, ());
   WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
   WEBRTC_STUB_CONST(delay_offset_ms, ());
   WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
   WEBRTC_STUB(StartDebugRecording, (FILE* handle));
   WEBRTC_STUB(StopDebugRecording, ());
+  WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
   webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
   webrtc::EchoControlMobile* echo_control_mobile() const override {
     return NULL;
   }
   webrtc::GainControl* gain_control() const override { return NULL; }
   webrtc::HighPassFilter* high_pass_filter() const override { return NULL; }
   webrtc::LevelEstimator* level_estimator() const override { return NULL; }
   webrtc::NoiseSuppression* noise_suppression() const override { return NULL; }
   webrtc::VoiceDetection* voice_detection() const override { return NULL; }
 
@@ skipping 1104 matching lines @@
   DtmfInfo dtmf_info_;
   webrtc::VoEMediaProcess* media_processor_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_