Index: webrtc/api/mediastreaminterface.h |
diff --git a/webrtc/api/mediastreaminterface.h b/webrtc/api/mediastreaminterface.h |
index 693e5d00f07b673a39dcd8408ff21a5a08753cc2..a52f0c76455e0daeeff445b3c4a1822bdf7c8833 100644 |
--- a/webrtc/api/mediastreaminterface.h |
+++ b/webrtc/api/mediastreaminterface.h |
@@ -203,14 +203,16 @@ class AudioSourceInterface : public MediaSourceInterface { |
class AudioProcessorInterface : public rtc::RefCountInterface { |
public: |
struct AudioProcessorStats { |
- AudioProcessorStats() : typing_noise_detected(false), |
- echo_return_loss(0), |
- echo_return_loss_enhancement(0), |
- echo_delay_median_ms(0), |
- echo_delay_std_ms(0), |
- aec_quality_min(0.0), |
- residual_echo_likelihood(0.0f), |
- aec_divergent_filter_fraction(0.0) {} |
+ AudioProcessorStats() |
+ : typing_noise_detected(false), |
+ echo_return_loss(0), |
+ echo_return_loss_enhancement(0), |
+ echo_delay_median_ms(0), |
+ echo_delay_std_ms(0), |
+ aec_quality_min(0.0), |
+ residual_echo_likelihood(0.0f), |
+ residual_echo_likelihood_recent_max(0.0f), |
+ aec_divergent_filter_fraction(0.0) {} |
~AudioProcessorStats() {} |
bool typing_noise_detected; |
@@ -220,6 +222,7 @@ class AudioProcessorInterface : public rtc::RefCountInterface { |
int echo_delay_std_ms; |
float aec_quality_min; |
float residual_echo_likelihood; |
+ float residual_echo_likelihood_recent_max; |
float aec_divergent_filter_fraction; |
}; |