Index: webrtc/modules/audio_processing/audio_processing_impl.cc |
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc |
index c892b7886c71a4ad78603484808cc4e82774e20e..b504e5c583c0383f0516904238022ff3dc9a1391 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -1278,6 +1278,29 @@ int AudioProcessingImpl::StopDebugRecording() { |
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP |
} |
+void AudioProcessingImpl::SetStatisticsEnabled(bool enabled) { |
+ int success = public_submodules_->echo_cancellation->enable_metrics(enabled); |
+ RTC_DCHECK_EQ(0, success); |
+ success = |
+ public_submodules_->echo_cancellation->enable_delay_logging(enabled); |
+ RTC_DCHECK_EQ(0, success); |
+} |
+ |
+void AudioProcessingImpl::GetStatistics( |
+ AudioProcessingStatistics* stats) const { |
+ RTC_DCHECK(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); |
+} |
+ |
EchoCancellation* AudioProcessingImpl::echo_cancellation() const { |
return public_submodules_->echo_cancellation.get(); |
} |