Introduced new scheme for controlling the functionality inside the audio processing module

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 24 matching lines @@
 
 class AgcManagerDirect;
 class AudioConverter;
 
 class NonlinearBeamformer;
 
 class AudioProcessingImpl : public AudioProcessing {
  public:
   // Methods forcing APM to run in a single-threaded manner.
   // Acquires both the render and capture locks.
-  explicit AudioProcessingImpl(const Config& config);
+  explicit AudioProcessingImpl(const webrtc::Config& config);
   // AudioProcessingImpl takes ownership of beamformer.
-  AudioProcessingImpl(const Config& config, NonlinearBeamformer* beamformer);
+  AudioProcessingImpl(const webrtc::Config& config,
+                      NonlinearBeamformer* beamformer);
   ~AudioProcessingImpl() override;
   int Initialize() override;
   int Initialize(int input_sample_rate_hz,
                  int output_sample_rate_hz,
                  int reverse_sample_rate_hz,
                  ChannelLayout input_layout,
                  ChannelLayout output_layout,
                  ChannelLayout reverse_layout) override;
   int Initialize(const ProcessingConfig& processing_config) override;
-  void SetExtraOptions(const Config& config) override;
+  void ApplyConfig(const AudioProcessing::Config& config) override;
+  void SetExtraOptions(const webrtc::Config& config) override;
   void UpdateHistogramsOnCallEnd() override;
   int StartDebugRecording(const char filename[kMaxFilenameSize],
                           int64_t max_log_size_bytes) override;
   int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override;
 
   int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
 
   // Capture-side exclusive methods possibly running APM in a
   // multi-threaded manner. Acquire the capture lock.
@@ skipping 237 matching lines @@
     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,
-                             bool intelligibility_enabled,
-                             bool level_controller_enabled)
+                             bool intelligibility_enabled)
         : fwd_proc_format(kSampleRate16kHz),
           split_rate(kSampleRate16kHz),
           stream_delay_ms(0),
           beamformer_enabled(beamformer_enabled),
-          intelligibility_enabled(intelligibility_enabled),
-          level_controller_enabled(level_controller_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;
-    bool level_controller_enabled;
+    bool level_controller_enabled = false;
   } capture_nonlocked_;
 
   struct ApmRenderState {
     ApmRenderState();
     ~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_