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..bf5b357c8140672e988763ef47bf469ce0bf40d5 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -433,6 +433,10 @@ int AudioProcessingImpl::InitializeLocked() { |
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()); |
@@ -1278,6 +1282,22 @@ int AudioProcessingImpl::StopDebugRecording() { |
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP |
} |
+AudioProcessing::AudioProcessingStatistics AudioProcessingImpl::GetStatistics() |
+ const { |
+ AudioProcessingStatistics stats; |
+ EchoCancellation::Metrics metrics; |
+ public_submodules_->echo_cancellation->GetMetrics(&metrics); |
+ stats.a_nlp.Set(metrics.a_nlp); |
+ stats.divergent_filter_fraction = metrics.divergent_filter_fraction; |
+ stats.echo_return_loss.Set(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); |
+ 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(); |
} |