Make use of new APM statistics interface.

webrtc/modules/audio_processing/audio_processing_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 102 matching lines @@
       return uppermost_native_rate;
     }
     if (rate >= minimum_rate) {
       return rate;
     }
   }
   RTC_NOTREACHED();
   return uppermost_native_rate;
 }
 
+webrtc::AudioProcessing::Statistic FixStatistic(
+    const webrtc::AudioProcessing::Statistic& stat) {
+  webrtc::AudioProcessing::Statistic result = stat;
+  if (result.average == -100 || result.instant == -100 ||
+      result.maximum == -100 || result.minimum == -100) {
+    result.average = result.instant = result.maximum = result.minimum = -100;
+  }
+  return result;
+}
+
 // Maximum length that a frame of samples can have.
 static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
 // Maximum number of frames to buffer in the render queue.
 // TODO(peah): Decrease this once we properly handle hugely unbalanced
 // reverse and forward call numbers.
 static const size_t kMaxNumFramesToBuffer = 100;
 
 }  // namespace
 
 // Throughout webrtc, it's assumed that success is represented by zero.
@@ skipping 1349 matching lines @@
     const {
   return AudioProcessingStatistics();
 }
 
 AudioProcessing::AudioProcessingStatistics AudioProcessingImpl::GetStatistics()
     const {
   AudioProcessingStatistics stats;
   EchoCancellation::Metrics metrics;
   int success = public_submodules_->echo_cancellation->GetMetrics(&metrics);
   if (success == Error::kNoError) {
-    stats.a_nlp.Set(metrics.a_nlp);
+    stats.a_nlp.Set(FixStatistic(metrics.a_nlp));

[peah-webrtc, 2016/11/10 10:36:09]

What you do here is basically to ensure that if any of the metrics.* struct
members is -100, then all is set to -100, right?

That makes sense and seems like the correct way to report the metrics.

It is, however, weird that the APM should ensure that the metrics in the AEC are
correctly reported. 
That constitutes an unneccessary dependency of APM on the implementation of the
AEC.

I think it would make more sense to move this correction into the AEC where the
metrics are computed. In that way APM would not need to be concerned about the
implementation of the AEC metrics.

WDYT?

[ossu, 2016/11/10 11:29:47]

peah: Did you read my analysis of the problem in question? Do you see any
problems with it? I'd prefer it if the problem could be solved at the source,
rather than plastering it over afterwards.

     stats.divergent_filter_fraction = metrics.divergent_filter_fraction;
-    stats.echo_return_loss.Set(metrics.echo_return_loss);
+    stats.echo_return_loss.Set(FixStatistic(metrics.echo_return_loss));
     stats.echo_return_loss_enhancement.Set(
-        metrics.echo_return_loss_enhancement);
-    stats.residual_echo_return_loss.Set(metrics.residual_echo_return_loss);
+        FixStatistic(metrics.echo_return_loss_enhancement));
+    stats.residual_echo_return_loss.Set(
+        FixStatistic(metrics.residual_echo_return_loss));
   }
   public_submodules_->echo_cancellation->GetDelayMetrics(
       &stats.delay_median, &stats.delay_standard_deviation,
       &stats.fraction_poor_delays);
   return stats;
 }
 
 EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
   return public_submodules_->echo_cancellation.get();
 }
@@ skipping 315 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc