Major updates to the echo removal functionality in AEC3

webrtc/modules/audio_processing/aec3/render_delay_controller.cc

Index: webrtc/modules/audio_processing/aec3/render_delay_controller.cc
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller.cc b/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
index c19945d04908c5960615e116b79473064287b350..3f7b108a751cb06e9a6175355dd1585e4dbd6e6a 100644
--- a/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
@@ -110,7 +110,7 @@ size_t RenderDelayControllerImpl::GetDelay(
     // Compute and set new render delay buffer delay.
     const size_t new_delay =
         ComputeNewBufferDelay(delay_, echo_path_delay_samples_);
-    if (new_delay != delay_ && align_call_counter_ > 250) {
+    if (new_delay != delay_ && align_call_counter_ > kNumBlocksPerSecond) {
       delay_ = new_delay;
     }
 
@@ -119,7 +119,7 @@ size_t RenderDelayControllerImpl::GetDelay(
     const int headroom = echo_path_delay_samples_ - delay_ * kBlockSize;
     RTC_DCHECK_LE(0, headroom);
     headroom_samples_ = rtc::Optional<size_t>(headroom);
-  } else if (++blocks_since_last_delay_estimate_ > 250 * 20) {
+  } else if (++blocks_since_last_delay_estimate_ > 20 * kNumBlocksPerSecond) {
     headroom_samples_ = rtc::Optional<size_t>();
   }