[media_stream_audio_processor] Avoids updating UMA stats before AEC has valid results

content/renderer/media/media_stream_audio_processor_options.cc

 // 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.
 
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/logging.h"
 #include "base/metrics/field_trial.h"
@@ skipping 204 matching lines @@
 
   for (size_t i = 0; i < arraysize(kDefaultAudioConstraints); ++i) {
     if (kDefaultAudioConstraints[i].key == key)
       return kDefaultAudioConstraints[i].value;
   }
 
   return false;
 }
 
 EchoInformation::EchoInformation()
-    : num_chunks_(0) {}
+    : num_chunks_(0), echo_frames_received_(false) {
+}
 
 EchoInformation::~EchoInformation() {}
 
 void EchoInformation::UpdateAecDelayStats(
     webrtc::EchoCancellation* echo_cancellation) {
+  // Only start collecting stats if we know echo cancellation has measured an
+  // echo. Otherwise we clutter the stats with for example cases where only the
+  // microphone is used.
+  if (!echo_frames_received_ & !echo_cancellation->stream_has_echo()) {

[tommi (sloooow) - chröme, 2015/04/23 14:35:35]

no {}

[bjornv, 2015/04/24 08:01:55]

Done.

+    return;
+  }
+  echo_frames_received_ = true;
   // In WebRTC, three echo delay metrics are calculated and updated every
   // 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;
   }
@@ skipping 129 matching lines @@
   int median = 0, std = 0;
   float dummy = 0;
   if (echo_cancellation->GetDelayMetrics(&median, &std, &dummy) ==
       webrtc::AudioProcessing::kNoError) {
     stats->echo_delay_median_ms = median;
     stats->echo_delay_std_ms = std;
   }
 }
 
 }  // namespace content