Make use of new APM statistics interface.

webrtc/audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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 179 matching lines @@
   }
 
   // Local speech level.
   {
     unsigned int level = 0;
     int error = volume->GetSpeechInputLevelFullRange(level);
     RTC_DCHECK_EQ(0, error);
     stats.audio_level = static_cast<int32_t>(level);
   }
 
-  bool echo_metrics_on = false;
-  int error = processing->GetEcMetricsStatus(echo_metrics_on);
-  RTC_DCHECK_EQ(0, error);
-  if (echo_metrics_on) {
-    // These can also be negative, but in practice -1 is only used to signal
-    // insufficient data, since the resolution is limited to multiples of 4 ms.
-    int median = -1;
-    int std = -1;
-    float dummy = 0.0f;
-    error = processing->GetEcDelayMetrics(median, std, dummy);
-    RTC_DCHECK_EQ(0, error);
-    stats.echo_delay_median_ms = median;
-    stats.echo_delay_std_ms = std;
-
-    // These can take on valid negative values, so use the lowest possible level
-    // as default rather than -1.
-    int erl = -100;
-    int erle = -100;
-    int dummy1 = 0;
-    int dummy2 = 0;
-    error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2);
-    RTC_DCHECK_EQ(0, error);
-    stats.echo_return_loss = erl;
-    stats.echo_return_loss_enhancement = erle;
-  }
-
-  // TODO(ivoc): Hook this up to the residual echo detector.
-  stats.residual_echo_likelihood = 0.0f;
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  RTC_DCHECK(base->audio_processing());
+  auto audio_processing_stats = base->audio_processing()->GetStatistics();
+  stats.echo_delay_median_ms = audio_processing_stats.delay_median;
+  stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation;
+  stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant();
+  stats.echo_return_loss_enhancement =
+      audio_processing_stats.echo_return_loss_enhancement.instant();
+  stats.residual_echo_likelihood =
+      audio_processing_stats.residual_echo_likelihood;
 
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   stats.typing_noise_detected = audio_state->typing_noise_detected();
 
   return stats;
 }
 
 void AudioSendStream::SignalNetworkState(NetworkState state) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 148 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError();
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc