Reland of New statistics interface for APM

webrtc/modules/audio_processing/echo_cancellation_impl.cc

 /*
  *  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 252 matching lines @@
 int EchoCancellationImpl::enable_metrics(bool enable) {
   {
     rtc::CritScope cs(crit_capture_);
     metrics_enabled_ = enable;
   }
   return Configure();
 }
 
 bool EchoCancellationImpl::are_metrics_enabled() const {
   rtc::CritScope cs(crit_capture_);
-  return metrics_enabled_;
+  return enabled_ && metrics_enabled_;
 }
 
 // TODO(ajm): we currently just use the metrics from the first AEC. Think more
 //            aboue the best way to extend this to multi-channel.
 int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
   rtc::CritScope cs(crit_capture_);
   if (metrics == NULL) {
     return AudioProcessing::kNullPointerError;
   }
 
@@ skipping 42 matching lines @@
 int EchoCancellationImpl::enable_delay_logging(bool enable) {
   {
     rtc::CritScope cs(crit_capture_);
     delay_logging_enabled_ = enable;
   }
   return Configure();
 }
 
 bool EchoCancellationImpl::is_delay_logging_enabled() const {
   rtc::CritScope cs(crit_capture_);
-  return delay_logging_enabled_;
+  return enabled_ && delay_logging_enabled_;
 }
 
 bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return delay_agnostic_enabled_;
 }
 
 bool EchoCancellationImpl::is_aec3_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return aec3_enabled_;
@@ skipping 159 matching lines @@
   return error;
 }
 
 size_t EchoCancellationImpl::NumCancellersRequired(
     size_t num_output_channels,
     size_t num_reverse_channels) {
   return num_output_channels * num_reverse_channels;
 }
 
 }  // namespace webrtc