| 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 189d7095e0c89225eb13333fe3b569bce62baaeb..12b4c949046f08ddcc3f974d26024bea7e516b54 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| @@ -443,10 +443,6 @@
|
| num_proc_channels());
|
| AllocateRenderQueue();
|
|
|
| - 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());
|
| @@ -1419,22 +1415,6 @@
|
| #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.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 {
|
|
|