audio_processing: Adds two UMA histograms logging delay jumps in AEC

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 116 matching lines @@
   int stream_delay_ms() const override;
   bool was_stream_delay_set() const override;
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
   bool stream_key_pressed() const override;
   int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
   int StartDebugRecording(FILE* handle) override;
   int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
+  void UpdateHistogramsOnCallEnd() override;
   EchoCancellation* echo_cancellation() const override;
   EchoControlMobile* echo_control_mobile() const override;
   GainControl* gain_control() const override;
   HighPassFilter* high_pass_filter() const override;
   LevelEstimator* level_estimator() const override;
   NoiseSuppression* noise_suppression() const override;
   VoiceDetection* voice_detection() const override;
 
  protected:
   // Overridden in a mock.
@@ skipping 56 matching lines @@
   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_;
+  int stream_delay_jumps_;
+  int aec_system_delay_jumps_;
 
   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_