Surface the IntelligibilityEnhancer on MediaConstraints

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 258 matching lines @@
                        {kSampleRate16kHz, 1, false},
                        {kSampleRate16kHz, 1, false},
                        {kSampleRate16kHz, 1, false}}}),
           rev_proc_format(kSampleRate16kHz, 1) {}
     ProcessingConfig api_format;
     StreamConfig rev_proc_format;
   } formats_;
 
   // APM constants.
   const struct ApmConstants {
-    ApmConstants(int agc_startup_min_volume,
-                 bool use_experimental_agc,
-                 bool intelligibility_enabled)
+    ApmConstants(int agc_startup_min_volume, bool use_experimental_agc)
         :  // Format of processing streams at input/output call sites.
           agc_startup_min_volume(agc_startup_min_volume),
-          use_experimental_agc(use_experimental_agc),
-          intelligibility_enabled(intelligibility_enabled) {}
+          use_experimental_agc(use_experimental_agc) {}
     int agc_startup_min_volume;
     bool use_experimental_agc;
-    bool intelligibility_enabled;
   } constants_;
 
   struct ApmCaptureState {
     ApmCaptureState(bool transient_suppressor_enabled,
                     const std::vector<Point>& array_geometry,
                     SphericalPointf target_direction)
         : aec_system_delay_jumps(-1),
           delay_offset_ms(0),
           was_stream_delay_set(false),
           last_stream_delay_ms(0),
@@ skipping 19 matching lines @@
     SphericalPointf target_direction;
     std::unique_ptr<AudioBuffer> capture_audio;
     // Only the rate and samples fields of fwd_proc_format_ are used because the
     // forward processing number of channels is mutable and is tracked by the
     // capture_audio_.
     StreamConfig fwd_proc_format;
     int split_rate;
   } capture_ GUARDED_BY(crit_capture_);
 
   struct ApmCaptureNonLockedState {
-    ApmCaptureNonLockedState(bool beamformer_enabled)
+    ApmCaptureNonLockedState(bool beamformer_enabled,
+                             bool intelligibility_enabled)
         : fwd_proc_format(kSampleRate16kHz),
           split_rate(kSampleRate16kHz),
           stream_delay_ms(0),
-          beamformer_enabled(beamformer_enabled) {}
+          beamformer_enabled(beamformer_enabled),
+          intelligibility_enabled(intelligibility_enabled) {}
     // Only the rate and samples fields of fwd_proc_format_ are used because the
     // forward processing number of channels is mutable and is tracked by the
     // capture_audio_.
     StreamConfig fwd_proc_format;
     int split_rate;
     int stream_delay_ms;
     bool beamformer_enabled;
+    bool intelligibility_enabled;
   } capture_nonlocked_;
 
   struct ApmRenderState {
     std::unique_ptr<AudioConverter> render_converter;
     std::unique_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_