audio_processing: Adds two UMA histograms logging delay jumps in AEC

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 360 matching lines @@
   // of |handle| and closes it at StopDebugRecording().
   // TODO(xians): Make this interface pure virtual.
   virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) {
       return -1;
   }
 
   // Stops recording debugging information, and closes the file. Recording
   // cannot be resumed in the same file (without overwriting it).
   virtual int StopDebugRecording() = 0;
 
+  // Use to send UMA histograms at end of a call. Note that all histogram
+  // specific member variables are reset.
+  virtual void UpdateHistogramsOnCallEnd() = 0;
+
   // These provide access to the component interfaces and should never return
   // NULL. The pointers will be valid for the lifetime of the APM instance.
   // The memory for these objects is entirely managed internally.
   virtual EchoCancellation* echo_cancellation() const = 0;
   virtual EchoControlMobile* echo_control_mobile() const = 0;
   virtual GainControl* gain_control() const = 0;
   virtual HighPassFilter* high_pass_filter() const = 0;
   virtual LevelEstimator* level_estimator() const = 0;
   virtual NoiseSuppression* noise_suppression() const = 0;
   virtual VoiceDetection* voice_detection() const = 0;
@@ skipping 392 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_