New statistics interface for APM

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 96 matching lines @@
   int proc_sample_rate_hz() const override;
   int proc_split_sample_rate_hz() const override;
   size_t num_input_channels() const override;
   size_t num_proc_channels() const override;
   size_t num_output_channels() const override;
   size_t num_reverse_channels() const override;
   int stream_delay_ms() const override;
   bool was_stream_delay_set() const override
       EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
+  AudioProcessingStatistics GetStatistics() const override;
+
   // Methods returning pointers to APM submodules.
   // No locks are aquired in those, as those locks
   // would offer no protection (the submodules are
   // created only once in a single-treaded manner
   // during APM creation).
   EchoCancellation* echo_cancellation() const override;
   EchoControlMobile* echo_control_mobile() const override;
   GainControl* gain_control() const override;
   HighPassFilter* high_pass_filter() const override;
   LevelEstimator* level_estimator() const override;
@@ skipping 233 matching lines @@
     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_