AEC3 Tuning changes

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 
+#include <math.h>
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
@@ skipping 1117 matching lines @@
     RmsLevel::Levels levels = capture_input_rms_.AverageAndPeak();
     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelAverageRms",
                                 levels.average, 1, RmsLevel::kMinLevelDb, 64);
     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelPeakRms",
                                 levels.peak, 1, RmsLevel::kMinLevelDb, 64);
   }
 
   if (private_submodules_->echo_canceller3) {
     const int new_agc_level = gain_control()->stream_analog_level();
     capture_.echo_path_gain_change =
-        (capture_.previous_agc_level != new_agc_level);
+        abs(capture_.previous_agc_level - new_agc_level) > 5;
     capture_.previous_agc_level = new_agc_level;
     private_submodules_->echo_canceller3->AnalyzeCapture(capture_buffer);
   }
 
   if (constants_.use_experimental_agc &&
       public_submodules_->gain_control->is_enabled()) {
     private_submodules_->agc_manager->AnalyzePreProcess(
         capture_buffer->channels()[0], capture_buffer->num_channels(),
         capture_nonlocked_.capture_processing_format.num_frames());
   }
@@ skipping 841 matching lines @@
       previous_agc_level(0),
       echo_path_gain_change(false) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc