Added functionality for specifying the initial signal level to use for the gain estimation in the l…

webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc

 /*
  *  Copyright (c) 2016 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/level_controller/peak_level_estimator.h"
 
 #include <algorithm>
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/modules/audio_processing/level_controller/lc_constants.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 
 namespace webrtc {
 
 PeakLevelEstimator::PeakLevelEstimator() {
   Initialize();
 }
 
 PeakLevelEstimator::~PeakLevelEstimator() {}
 
 void PeakLevelEstimator::Initialize() {
-  peak_level_ = kTargetLcPeakLevel;
+  peak_level_ = initial_peak_level_;
   hold_counter_ = 0;
   initialization_phase_ = true;
 }
 
+void PeakLevelEstimator::SetInitialLevel(float level) {

[hlundin-webrtc, 2016/08/18 12:58:13]

The method should be called SetInitialPeakLevel, reflecting the name of the
member variable.

[peah-webrtc, 2016/08/18 17:02:58]

Good find!
Done.

+  initial_peak_level_ = level;
+  Initialize();
+}
+
 float PeakLevelEstimator::Analyze(SignalClassifier::SignalType signal_type,
                                   float frame_peak_level) {
   if (frame_peak_level == 0) {
     RTC_DCHECK_LE(30.f, peak_level_);
     return peak_level_;
   }
 
   if (peak_level_ < frame_peak_level) {
     // Smoothly update the estimate upwards when the frame peak level is
     // higher than the estimate.
@@ skipping 13 matching lines @@
                    peak_level_ * 0.995f);
     }
   }
 
   peak_level_ = std::max(peak_level_, 30.f);
 
   return peak_level_;
 }
 
 }  // namespace webrtc