This CL renames variables and method and removes some one-line methods inside the 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 286 matching lines @@
   //     |processing_config.input_stream()|.
   //
   // The float interfaces accept arbitrary rates and support differing input and
   // output layouts, but the output must have either one channel or the same
   // number of channels as the input.
   virtual int Initialize(const ProcessingConfig& processing_config) = 0;
 
   // Initialize with unpacked parameters. See Initialize() above for details.
   //
   // TODO(mgraczyk): Remove once clients are updated to use the new interface.
-  virtual 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) = 0;
+  virtual int Initialize(int capture_input_sample_rate_hz,
+                         int capture_output_sample_rate_hz,
+                         int render_sample_rate_hz,
+                         ChannelLayout capture_input_layout,
+                         ChannelLayout capture_output_layout,
+                         ChannelLayout render_input_layout) = 0;
 
   // TODO(peah): This method is a temporary solution used to take control
   // over the parameters in the audio processing module and is likely to change.
   virtual void ApplyConfig(const Config& config) = 0;
 
   // Pass down additional options which don't have explicit setters. This
   // ensures the options are applied immediately.
   virtual void SetExtraOptions(const webrtc::Config& config) = 0;
 
   // TODO(ajm): Only intended for internal use. Make private and friend the
@@ skipping 64 matching lines @@
   //
   // The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_|
   // members of |frame| must be valid.
   virtual int ProcessReverseStream(AudioFrame* frame) = 0;
 
   // Accepts deinterleaved float audio with the range [-1, 1]. Each element
   // of |data| points to a channel buffer, arranged according to |layout|.
   // TODO(mgraczyk): Remove once clients are updated to use the new interface.
   virtual int AnalyzeReverseStream(const float* const* data,
                                    size_t samples_per_channel,
-                                   int rev_sample_rate_hz,
+                                   int sample_rate_hz,
                                    ChannelLayout layout) = 0;
 
   // Accepts deinterleaved float audio with the range [-1, 1]. Each element of
   // |data| points to a channel buffer, arranged according to |reverse_config|.
   virtual int ProcessReverseStream(const float* const* src,
-                                   const StreamConfig& reverse_input_config,
-                                   const StreamConfig& reverse_output_config,
+                                   const StreamConfig& input_config,
+                                   const StreamConfig& output_config,
                                    float* const* dest) = 0;
 
   // This must be called if and only if echo processing is enabled.
   //
   // Sets the |delay| in ms between ProcessReverseStream() receiving a far-end
   // frame and ProcessStream() receiving a near-end frame containing the
   // corresponding echo. On the client-side this can be expressed as
   //   delay = (t_render - t_analyze) + (t_process - t_capture)
   // where,
   //   - t_analyze is the time a frame is passed to ProcessReverseStream() and
@@ skipping 580 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_