| 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 3a8f7f54633aa958426c62eca6ebe76658c6a343..c253775f14e8bcf08d9881baf92439431a3d63a2 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| @@ -1585,6 +1585,19 @@ int AudioProcessingImpl::StopDebugRecording() {
|
| #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
|
| }
|
|
|
| +AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics() {
|
| + residual_echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
|
| + echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
|
| + echo_return_loss_enhancement.Set(-100.0f, -100.0f, -100.0f, -100.0f);
|
| + a_nlp.Set(-100.0f, -100.0f, -100.0f, -100.0f);
|
| +}
|
| +
|
| +AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics(
|
| + const AudioProcessingStatistics& other) = default;
|
| +
|
| +AudioProcessing::AudioProcessingStatistics::~AudioProcessingStatistics() =
|
| + default;
|
| +
|
| // TODO(ivoc): Remove this when GetStatistics() becomes pure virtual.
|
| AudioProcessing::AudioProcessingStatistics AudioProcessing::GetStatistics()
|
| const {
|
| @@ -1606,6 +1619,8 @@ AudioProcessing::AudioProcessingStatistics AudioProcessingImpl::GetStatistics()
|
| }
|
| stats.residual_echo_likelihood =
|
| private_submodules_->residual_echo_detector->echo_likelihood();
|
| + stats.residual_echo_likelihood_recent_max =
|
| + private_submodules_->residual_echo_detector->echo_likelihood_recent_max();
|
| public_submodules_->echo_cancellation->GetDelayMetrics(
|
| &stats.delay_median, &stats.delay_standard_deviation,
|
| &stats.fraction_poor_delays);
|
|
|