UMA histogram WebRTC.AecDelayBasedQuality updated

content/renderer/media/media_stream_audio_processor_options.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
 
 #include <string>
 
 #include "base/files/file.h"
-#include "base/time/time.h"
 #include "content/common/content_export.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 namespace webrtc {
 
 class AudioFrame;
 class AudioProcessing;
+class EchoCancellation;
 class MediaConstraintsInterface;
 class TypingDetection;
 
 }
 
 namespace content {
 
 class RTCMediaConstraints;
 
 using webrtc::AudioProcessing;
@@ skipping 56 matching lines @@
   bool default_audio_processing_constraint_value_;
 };
 
 // A helper class to log echo information in general and Echo Cancellation
 // quality in particular.
 class CONTENT_EXPORT EchoInformation {
  public:
   EchoInformation();
   virtual ~EchoInformation();
 
-  // Updates delay statistics with a new |delay|.
-  void UpdateAecDelayStats(int delay);
+  void UpdateAecDelayStats(webrtc::EchoCancellation* echo_cancellation);
 
  private:
-  // Updates UMA histograms with an interval of |kTimeBetweenLogsInSeconds|.
+  // Updates UMA histograms with an interval of 5 seconds.
   void LogAecDelayStats();
 
-  // Counters for determining how often the estimated delay in the AEC is out of
-  // bounds.
-  int echo_poor_delay_counts_;
-  int echo_total_delay_counts_;
-  base::TimeTicks last_log_time_;
+  // Counters to be able to aquire a 5 second aggregated metric out of 1 second
+  // aggregated webrtc::EchoCancellation::GetEchoDelayMetrics() queries.
+  int num_chunks_;
+  int num_queries_;
+  float echo_fraction_poor_delays_;
 
   DISALLOW_COPY_AND_ASSIGN(EchoInformation);
 };
 
 // Enables the echo cancellation in |audio_processing|.
 void EnableEchoCancellation(AudioProcessing* audio_processing);
 
 // Enables the noise suppression in |audio_processing|.
 void EnableNoiseSuppression(AudioProcessing* audio_processing);
 
 // Enables the high pass filter in |audio_processing|.
 void EnableHighPassFilter(AudioProcessing* audio_processing);
 
 // Enables the typing detection in |audio_processing|.
 void EnableTypingDetection(AudioProcessing* audio_processing,
                            webrtc::TypingDetection* typing_detector);
 
 // Starts the echo cancellation dump in |audio_processing|.
 void StartEchoCancellationDump(AudioProcessing* audio_processing,
                                base::File aec_dump_file);
 
 // Stops the echo cancellation dump in |audio_processing|.
 // This method has no impact if echo cancellation dump has not been started on
 // |audio_processing|.
 void StopEchoCancellationDump(AudioProcessing* audio_processing);
 
 void EnableAutomaticGainControl(AudioProcessing* audio_processing);
 
-void GetAecStats(AudioProcessing* audio_processing,
+void GetAecStats(webrtc::EchoCancellation* echo_cancellation,
                  webrtc::AudioProcessorInterface::AudioProcessorStats* stats);
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_