Adds UMA histogram for system delay jumps

webrtc/modules/audio_processing/audio_processing_impl.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 153 matching lines @@
   int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   bool is_data_processed() const;
   bool output_copy_needed(bool is_data_processed) const;
   bool synthesis_needed(bool is_data_processed) const;
   bool analysis_needed(bool is_data_processed) const;
   void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   EchoCancellationImpl* echo_cancellation_;
   EchoControlMobileImpl* echo_control_mobile_;
   GainControlImpl* gain_control_;
   HighPassFilterImpl* high_pass_filter_;
   LevelEstimatorImpl* level_estimator_;
   NoiseSuppressionImpl* noise_suppression_;
   VoiceDetectionImpl* voice_detection_;
   rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
 
@@ skipping 16 matching lines @@
   // is mutable and is tracked by the capture_audio_.
   AudioRate fwd_proc_format_;
   AudioFormat fwd_out_format_;
   AudioFormat rev_in_format_;
   AudioFormat rev_proc_format_;
   int split_rate_;
 
   int stream_delay_ms_;
   int delay_offset_ms_;
   bool was_stream_delay_set_;
+  int last_stream_delay_ms_;
+  int last_aec_system_delay_ms_;
 
   bool output_will_be_muted_ GUARDED_BY(crit_);
 
   bool key_pressed_;
 
   // Only set through the constructor's Config parameter.
   const bool use_new_agc_;
   rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
   int agc_startup_min_volume_;
 
   bool transient_suppressor_enabled_;
   rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
   const bool beamformer_enabled_;
   rtc::scoped_ptr<Beamformer<float>> beamformer_;
   const std::vector<Point> array_geometry_;
 
   const bool supports_48kHz_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_