Simplify WebRTC DelayMetrics query in MediaStreamAudioProcessor

As of webrtc roll https://chromium.googlesource.com/chromium/src/+/724ab4480d4af0c4a64752078a7cd5046782789c
a longer aggregation window is used and we can now match it to simplify the query.

BUG=450193
TESTED=locally on Mac

Review URL: https://codereview.chromium.org/982333002

Cr-Commit-Position: refs/heads/master@{#319564}

content/renderer/media/media_stream_audio_processor_options.cc

@@ -221,28 +221,25 @@ bool MediaAudioConstraints::GetDefaultValueForConstraint(
 }
 
 EchoInformation::EchoInformation()
-    : num_chunks_(0),
-      num_queries_(0),
-      echo_fraction_poor_delays_(0.0f) {}
+    : num_chunks_(0) {}
 
 EchoInformation::~EchoInformation() {}
 
 void EchoInformation::UpdateAecDelayStats(
     webrtc::EchoCancellation* echo_cancellation) {
   // In WebRTC, three echo delay metrics are calculated and updated every
-  // second. We use one of them, |fraction_poor_delays|, but aggregate over
-  // five seconds to log in a UMA histogram to monitor Echo Cancellation
-  // quality. Since the stat in WebRTC has a fixed aggregation window of one
-  // second we query the stat every second and average over five such queries.
-  // WebRTC process audio in 10 ms chunks.
-  const int kNumChunksInOneSecond = 100;
+  // five seconds. We use one of them, |fraction_poor_delays| to log in a UMA
+  // histogram an Echo Cancellation quality metric. The stat in WebRTC has a
+  // fixed aggregation window of five seconds, so we use the same query
+  // frequency to avoid logging old values.
+  const int kNumChunksInFiveSeconds = 500;
   if (!echo_cancellation->is_delay_logging_enabled() ||
       !echo_cancellation->is_enabled()) {
     return;
   }
 
   num_chunks_++;
-  if (num_chunks_ < kNumChunksInOneSecond) {
+  if (num_chunks_ < kNumChunksInFiveSeconds) {
     return;
   }
 
@@ -251,30 +248,13 @@ void EchoInformation::UpdateAecDelayStats(
   if (echo_cancellation->GetDelayMetrics(
           &dummy_median, &dummy_std, &fraction_poor_delays) ==
       webrtc::AudioProcessing::kNoError) {
-    echo_fraction_poor_delays_ += fraction_poor_delays;
-    num_queries_++;
     num_chunks_ = 0;
   }
-  LogAecDelayStats();
-}
-
-void EchoInformation::LogAecDelayStats() {
-  // We update the UMA statistics every 5 seconds.
-  const int kNumQueriesIn5Seconds = 5;
-  if (num_queries_ < kNumQueriesIn5Seconds) {
-    return;
-  }
-
-  // Calculate how frequent the AEC delay was out of bounds since last time we
-  // updated UMA histograms by averaging |echo_fraction_poor_delays_| over
-  // |num_queries_|. Then store the result into one of four histogram buckets;
-  // see DelayBasedEchoQuality.
-  float poor_delay_frequency = echo_fraction_poor_delays_ / num_queries_;
+  // Map |fraction_poor_delays| to an Echo Cancellation quality and log in UMA
+  // histogram. See DelayBasedEchoQuality for information on histogram buckets.
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AecDelayBasedQuality",
-                            EchoDelayFrequencyToQuality(poor_delay_frequency),
+                            EchoDelayFrequencyToQuality(fraction_poor_delays),
                             DELAY_BASED_ECHO_QUALITY_MAX);
-  num_queries_ = 0;
-  echo_fraction_poor_delays_ = 0.0f;
 }
 
 void EnableEchoCancellation(AudioProcessing* audio_processing) {

content/renderer/media/media_stream_audio_processor_options.h

@@ -97,14 +97,9 @@ class CONTENT_EXPORT EchoInformation {
   void UpdateAecDelayStats(webrtc::EchoCancellation* echo_cancellation);
 
  private:
-  // Updates UMA histograms with an interval of 5 seconds.
-  void LogAecDelayStats();
-
-  // Counters to be able to aquire a 5 second aggregated metric out of 1 second
-  // aggregated webrtc::EchoCancellation::GetEchoDelayMetrics() queries.
+  // Counter to track 5 seconds of processed 10 ms chunks in order to query a
+  // new metric from webrtc::EchoCancellation::GetEchoDelayMetrics().
   int num_chunks_;
-  int num_queries_;
-  float echo_fraction_poor_delays_;
 
   DISALLOW_COPY_AND_ASSIGN(EchoInformation);
 };