Disable the residual echo detector in audio mixer.

webrtc/modules/audio_mixer/frame_combiner.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 117 matching lines @@
   } else {
     std::transform(add_buffer.begin(), add_buffer.begin() + frame_length,
                    audio_frame_for_mixing->data_,
                    [](int32_t a) { return rtc::saturated_cast<int16_t>(a); });
   }
 }
 
 std::unique_ptr<AudioProcessing> CreateLimiter() {
   Config config;
   config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
+
   std::unique_ptr<AudioProcessing> limiter(AudioProcessing::Create(config));
   RTC_DCHECK(limiter);
 
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.residual_echo_detector.enabled = false;
+  limiter->ApplyConfig(apm_config);
+
   const auto check_no_error = [](int x) {
     RTC_DCHECK_EQ(x, AudioProcessing::kNoError);
   };
   auto* const gain_control = limiter->gain_control();
   check_no_error(gain_control->set_mode(GainControl::kFixedDigital));
 
   // We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
   // divide-by-2 but -7 is used instead to give a bit of headroom since the
   // AGC is not a hard limiter.
   check_no_error(gain_control->set_target_level_dbfs(7));
@@ skipping 50 matching lines @@
     std::vector<rtc::ArrayView<const int16_t>> input_frames;
     for (size_t i = 0; i < mix_list.size(); ++i) {
       input_frames.push_back(rtc::ArrayView<const int16_t>(
           mix_list[i]->data_, samples_per_channel * number_of_channels));
     }
     CombineMultipleFrames(input_frames, use_limiter_this_round, limiter_.get(),
                           audio_frame_for_mixing);
   }
 }
 }  // namespace webrtc