AGC: Route clipping parameter from webrtc::Config to AGC

webrtc/modules/audio_processing/agc/agc_manager_direct.h

 /*
  *  Copyright (c) 2013 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 28 matching lines @@
 //
 // This class is not thread-safe.
 class AgcManagerDirect final {
  public:
   // AgcManagerDirect will configure GainControl internally. The user is
   // responsible for processing the audio using it after the call to Process.
   // The operating range of startup_min_level is [12, 255] and any input value
   // outside that range will be clamped.
   AgcManagerDirect(GainControl* gctrl,
                    VolumeCallbacks* volume_callbacks,
-                   int startup_min_level);
+                   int startup_min_level,
+                   int clipped_level_min);
   // Dependency injection for testing. Don't delete |agc| as the memory is owned
   // by the manager.
   AgcManagerDirect(Agc* agc,
                    GainControl* gctrl,
                    VolumeCallbacks* volume_callbacks,
-                   int startup_min_level);
+                   int startup_min_level,
+                   int clipped_level_min);
   ~AgcManagerDirect();
 
   int Initialize();
   void AnalyzePreProcess(int16_t* audio,
                          int num_channels,
                          size_t samples_per_channel);
   void Process(const int16_t* audio, size_t length, int sample_rate_hz);
 
   // Call when the capture stream has been muted/unmuted. This causes the
   // manager to disregard all incoming audio; chances are good it's background
@@ skipping 25 matching lines @@
   int level_;
   int max_level_;
   int max_compression_gain_;
   int target_compression_;
   int compression_;
   float compression_accumulator_;
   bool capture_muted_;
   bool check_volume_on_next_process_;
   bool startup_;
   int startup_min_level_;
+  const int clipped_level_min_;
 
   std::unique_ptr<DebugFile> file_preproc_;
   std::unique_ptr<DebugFile> file_postproc_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_