| Index: webrtc/modules/audio_processing/aec/aec_core.cc
|
| diff --git a/webrtc/modules/audio_processing/aec/aec_core.cc b/webrtc/modules/audio_processing/aec/aec_core.cc
|
| index 910e21c7c336f40111e69713a07caa2c51fd0851..f249833269e1ecffbab9c57accb2157575b5ca4e 100644
|
| --- a/webrtc/modules/audio_processing/aec/aec_core.cc
|
| +++ b/webrtc/modules/audio_processing/aec/aec_core.cc
|
| @@ -1617,7 +1617,6 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
|
| aec->delay_std = -1;
|
| aec->fraction_poor_delays = -1.0f;
|
|
|
| - aec->signal_delay_correction = 0;
|
| aec->previous_delay = -2; // (-2): Uninitialized.
|
| aec->delay_correction_count = 0;
|
| aec->shift_offset = kInitialShiftOffset;
|
| @@ -1883,7 +1882,6 @@ void WebRtcAec_ProcessFrames(AecCore* aec,
|
| WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
|
| WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
|
| moved_elements);
|
| - aec->signal_delay_correction += moved_elements;
|
| // If we rely on reported system delay values only, a buffer underrun here
|
| // can never occur since we've taken care of that in 1) above. Here, we
|
| // apply signal based delay correction and can therefore end up with
|
|
|