Robustified the detection of linear filter quality in AEC3

webrtc/modules/audio_processing/aec3/aec_state.cc

 /*
  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 157 matching lines @@
   blocks_since_last_saturation_ =
       saturated_echo ? 0 : blocks_since_last_saturation_ + 1;
   echo_saturation_ = blocks_since_last_saturation_ < kSaturationLeakageBlocks;
 
   // Flag whether the linear filter estimate is usable.
   constexpr size_t kEchoPathChangeConvergenceBlocks = 2 * kNumBlocksPerSecond;
   usable_linear_estimate_ =
       (!echo_saturation_) &&
       (!render_received_ ||
        blocks_with_filter_adaptation_ > kEchoPathChangeConvergenceBlocks) &&
-      filter_delay_ && echo_path_change_counter_ <= 0;
+      filter_delay_ && echo_path_change_counter_ <= 0 && external_delay_;
 
   // After an amount of active render samples for which an echo should have been
   // detected in the capture signal if the ERL was not infinite, flag that a
   // headset is used.
   constexpr size_t kHeadSetDetectionBlocks = 5 * kNumBlocksPerSecond;
   headset_detected_ = !external_delay_ && !filter_delay_ &&
                       (!render_received_ || blocks_with_filter_adaptation_ >=
                                                 kHeadSetDetectionBlocks);
 
   // Update the room reverb estimate.
@@ skipping 111 matching lines @@
     max_nearend_ = e_abs;
     max_nearend_counter_ = 0;
   } else {
     if (++max_nearend_counter_ > 5 * kNumBlocksPerSecond) {
       max_nearend_ *= 0.995f;
     }
   }
 }
 
 }  // namespace webrtc