New statistics interface for APM

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 415 matching lines @@
                       formats_.api_format.input_stream().num_channels(),
                       capture_nonlocked_.capture_processing_format.num_frames(),
                       capture_audiobuffer_num_channels,
                       formats_.api_format.output_stream().num_frames()));
 
   public_submodules_->gain_control->Initialize(num_proc_channels(),
                                                proc_sample_rate_hz());
   public_submodules_->echo_cancellation->Initialize(
       proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
       num_proc_channels());
+  int success = public_submodules_->echo_cancellation->enable_metrics(true);
+  RTC_DCHECK_EQ(0, success);
+  success = public_submodules_->echo_cancellation->enable_delay_logging(true);
+  RTC_DCHECK_EQ(0, success);
   public_submodules_->echo_control_mobile->Initialize(
       proc_split_sample_rate_hz(), num_reverse_channels(),
       num_output_channels());
   if (constants_.use_experimental_agc) {
     if (!private_submodules_->agc_manager.get()) {
       private_submodules_->agc_manager.reset(new AgcManagerDirect(
           public_submodules_->gain_control.get(),
           public_submodules_->gain_control_for_experimental_agc.get(),
           constants_.agc_startup_min_volume));
     }
@@ skipping 825 matching lines @@
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // We just return if recording hasn't started.
   debug_dump_.debug_file->CloseFile();
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
+AudioProcessing::AudioProcessingStatistics AudioProcessingImpl::GetStatistics()
+    const {
+  AudioProcessingStatistics stats;
+  EchoCancellation::Metrics metrics;
+  public_submodules_->echo_cancellation->GetMetrics(&metrics);
+  stats.a_nlp = metrics.a_nlp;
+  stats.divergent_filter_fraction = metrics.divergent_filter_fraction;
+  stats.echo_return_loss = metrics.echo_return_loss;
+  stats.echo_return_loss_enhancement = metrics.echo_return_loss_enhancement;
+  stats.residual_echo_return_loss = 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();
 }
 
 EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
   return public_submodules_->echo_control_mobile.get();
 }
 
 GainControl* AudioProcessingImpl::gain_control() const {
   if (constants_.use_experimental_agc) {
@@ skipping 303 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc