Reland of New statistics interface for APM

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 451 matching lines @@
   virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) = 0;
 
   // 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;
 
+  // TODO(ivoc): Remove when the calling code no longer uses the old Statistics
+  //             API.
+  struct Statistic {
+    int instant = 0;  // Instantaneous value.
+    int average = 0;  // Long-term average.
+    int maximum = 0;  // Long-term maximum.
+    int minimum = 0;  // Long-term minimum.
+  };
+
+  struct Stat {
+    void Set(const Statistic& other) {
+      Set(other.instant, other.average, other.maximum, other.minimum);
+    }
+    void Set(float instant, float average, float maximum, float minimum) {
+      RTC_DCHECK_LE(instant, maximum);
+      RTC_DCHECK_GE(instant, minimum);
+      RTC_DCHECK_LE(average, maximum);
+      RTC_DCHECK_GE(average, minimum);
+      instant_ = instant;
+      average_ = average;
+      maximum_ = maximum;
+      minimum_ = minimum;
+    }
+    float instant() const { return instant_; }
+    float average() const { return average_; }
+    float maximum() const { return maximum_; }
+    float minimum() const { return minimum_; }
+
+   private:
+    float instant_ = 0.0f;  // Instantaneous value.
+    float average_ = 0.0f;  // Long-term average.
+    float maximum_ = 0.0f;  // Long-term maximum.
+    float minimum_ = 0.0f;  // Long-term minimum.
+  };
+
+  struct AudioProcessingStatistics {
+    // AEC Statistics.
+    // RERL = ERL + ERLE
+    Stat residual_echo_return_loss;
+    // ERL = 10log_10(P_far / P_echo)
+    Stat echo_return_loss;
+    // ERLE = 10log_10(P_echo / P_out)
+    Stat echo_return_loss_enhancement;
+    // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
+    Stat a_nlp;
+    // Fraction of time that the AEC linear filter is divergent, in a 1-second
+    // non-overlapped aggregation window.
+    float divergent_filter_fraction = 0.0f;
+
+    // The delay metrics consists of the delay median and standard deviation. It
+    // also consists of the fraction of delay estimates that can make the echo
+    // cancellation perform poorly. The values are aggregated until the first
+    // call to |GetStatistics()| and afterwards aggregated and updated every
+    // second. Note that if there are several clients pulling metrics from
+    // |GetStatistics()| during a session the first call from any of them will
+    // change to one second aggregation window for all.
+    int delay_median = 0;
+    int delay_standard_deviation = 0;
+    float fraction_poor_delays = 0.0f;
+
+    // Residual echo detector likelihood. This value is not yet calculated and
+    // is currently always set to zero.
+    // TODO(ivoc): Implement this stat.
+    float residual_echo_likelihood = 0.0f;
+  };
+
+  // TODO(ivoc): Make this pure virtual when all subclasses have been updated.
+  virtual AudioProcessingStatistics GetStatistics() const;
+
   // 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;
 
-  struct Statistic {
-    int instant;  // Instantaneous value.
-    int average;  // Long-term average.
-    int maximum;  // Long-term maximum.
-    int minimum;  // Long-term minimum.
-  };
-
   enum Error {
     // Fatal errors.
     kNoError = 0,
     kUnspecifiedError = -1,
     kCreationFailedError = -2,
     kUnsupportedComponentError = -3,
     kUnsupportedFunctionError = -4,
     kNullPointerError = -5,
     kBadParameterError = -6,
     kBadSampleRateError = -7,
@@ skipping 201 matching lines @@
     AudioProcessing::Statistic echo_return_loss_enhancement;
 
     // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
     AudioProcessing::Statistic a_nlp;
 
     // Fraction of time that the AEC linear filter is divergent, in a 1-second
     // non-overlapped aggregation window.
     float divergent_filter_fraction;
   };
 
+  // Deprecated. Use GetStatistics on the AudioProcessing interface instead.
   // TODO(ajm): discuss the metrics update period.
   virtual int GetMetrics(Metrics* metrics) = 0;
 
   // Enables computation and logging of delay values. Statistics are obtained
   // through |GetDelayMetrics()|.
   virtual int enable_delay_logging(bool enable) = 0;
   virtual bool is_delay_logging_enabled() const = 0;
 
   // The delay metrics consists of the delay |median| and the delay standard
   // deviation |std|. It also consists of the fraction of delay estimates
   // |fraction_poor_delays| that can make the echo cancellation perform poorly.
   // The values are aggregated until the first call to |GetDelayMetrics()| and
   // afterwards aggregated and updated every second.
   // Note that if there are several clients pulling metrics from
   // |GetDelayMetrics()| during a session the first call from any of them will
   // change to one second aggregation window for all.
-  // TODO(bjornv): Deprecated, remove.
+  // Deprecated. Use GetStatistics on the AudioProcessing interface instead.
   virtual int GetDelayMetrics(int* median, int* std) = 0;
+  // Deprecated. Use GetStatistics on the AudioProcessing interface instead.
   virtual int GetDelayMetrics(int* median, int* std,
                               float* fraction_poor_delays) = 0;
 
   // Returns a pointer to the low level AEC component.  In case of multiple
   // channels, the pointer to the first one is returned.  A NULL pointer is
   // returned when the AEC component is disabled or has not been initialized
   // successfully.
   virtual struct AecCore* aec_core() const = 0;
 
  protected:
@@ skipping 259 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_