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

webrtc/modules/audio_processing/audio_processing_impl.h

 /*
  *  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.
  */
 
@@ skipping 311 matching lines @@
                        {kSampleRate16kHz, 1, false},
                        {kSampleRate16kHz, 1, false},
                        {kSampleRate16kHz, 1, false}}}),
           render_processing_format(kSampleRate16kHz, 1) {}
     ProcessingConfig api_format;
     StreamConfig render_processing_format;
   } formats_;
 
   // APM constants.
   const struct ApmConstants {
-    ApmConstants(int agc_startup_min_volume, bool use_experimental_agc)
+    ApmConstants(int agc_startup_min_volume,
+                 int agc_clipped_level_min,
+                 bool use_experimental_agc)
         :  // Format of processing streams at input/output call sites.
           agc_startup_min_volume(agc_startup_min_volume),
+          agc_clipped_level_min(agc_clipped_level_min),
           use_experimental_agc(use_experimental_agc) {}
     int agc_startup_min_volume;
+    int agc_clipped_level_min;
     bool use_experimental_agc;
   } constants_;
 
   struct ApmCaptureState {
     ApmCaptureState(bool transient_suppressor_enabled,
                     const std::vector<Point>& array_geometry,
                     SphericalPointf target_direction);
     ~ApmCaptureState();
     int aec_system_delay_jumps;
     int delay_offset_ms;
@@ skipping 72 matching lines @@
   std::unique_ptr<
       SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
       agc_render_signal_queue_;
   std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
       red_render_signal_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_