Introduced the new parameter setting scheme for activating the high-pass filter in APM

webrtc/modules/audio_processing/include/audio_processing.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 181 matching lines @@
 //
 // APM accepts only linear PCM audio data in chunks of 10 ms. The int16
 // interfaces use interleaved data, while the float interfaces use deinterleaved
 // data.
 //
 // Usage example, omitting error checking:
 // AudioProcessing* apm = AudioProcessing::Create(0);
 //
 // AudioProcessing::Config config;
 // config.level_controller.enabled = true;
+// config.high_pass_filter.enabled = true;
 // apm->ApplyConfig(config)
 //
-// apm->high_pass_filter()->Enable(true);
-//
 // apm->echo_cancellation()->enable_drift_compensation(false);
 // apm->echo_cancellation()->Enable(true);
 //
 // apm->noise_reduction()->set_level(kHighSuppression);
 // apm->noise_reduction()->Enable(true);
 //
 // apm->gain_control()->set_analog_level_limits(0, 255);
 // apm->gain_control()->set_mode(kAdaptiveAnalog);
 // apm->gain_control()->Enable(true);
 //
@@ skipping 35 matching lines @@
   // The config is applied by passing the struct to the ApplyConfig method.
   struct Config {
     struct LevelController {
       bool enabled = false;
 
       // Sets the initial peak level to use inside the level controller in order
       // to compute the signal gain. The unit for the peak level is dBFS and
       // the allowed range is [-100, 0].
       float initial_peak_level_dbfs = -6.0206f;
     } level_controller;
+
+    struct HighPassFilter {
+      bool enabled = false;
+    } high_pass_filter;
   };
 
   // TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
   enum ChannelLayout {
     kMono,
     // Left, right.
     kStereo,
     // Mono, keyboard, and mic.
     kMonoAndKeyboard,
     // Left, right, keyboard, and mic.
@@ skipping 197 matching lines @@
   // Use to send UMA histograms at end of a call. Note that all histogram
   // specific member variables are reset.
   virtual void UpdateHistogramsOnCallEnd() = 0;
 
   // These provide access to the component interfaces and should never return
   // NULL. The pointers will be valid for the lifetime of the APM instance.
   // The memory for these objects is entirely managed internally.
   virtual EchoCancellation* echo_cancellation() const = 0;
   virtual EchoControlMobile* echo_control_mobile() const = 0;
   virtual GainControl* gain_control() const = 0;
+  // TODO(peah): Deprecate this API call.
   virtual HighPassFilter* high_pass_filter() const = 0;
   virtual LevelEstimator* level_estimator() const = 0;
   virtual NoiseSuppression* noise_suppression() const = 0;
   virtual VoiceDetection* voice_detection() const = 0;
 
   struct Statistic {
     int instant;  // Instantaneous value.
     int average;  // Long-term average.
     int maximum;  // Long-term maximum.
     int minimum;  // Long-term minimum.
@@ skipping 391 matching lines @@
   // signal reaches digital full-scale) in the current frame and the analog
   // level cannot be reduced.
   //
   // This could be used as an indicator to reduce or disable analog mic gain at
   // the audio HAL.
   virtual bool stream_is_saturated() const = 0;
 
  protected:
   virtual ~GainControl() {}
 };
-
+// TODO(peah): Remove this interface.
 // A filtering component which removes DC offset and low-frequency noise.
 // Recommended to be enabled on the client-side.
 class HighPassFilter {
  public:
   virtual int Enable(bool enable) = 0;
   virtual bool is_enabled() const = 0;
 
  protected:
   virtual ~HighPassFilter() {}
 };
@@ skipping 98 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_