Removed unused API functions in AudioProcessing and AudioProcessingModule

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 246 matching lines @@
                          int output_sample_rate_hz,
                          int reverse_sample_rate_hz,
                          ChannelLayout input_layout,
                          ChannelLayout output_layout,
                          ChannelLayout reverse_layout) = 0;
 
   // Pass down additional options which don't have explicit setters. This
   // ensures the options are applied immediately.
   virtual void SetExtraOptions(const Config& config) = 0;
 
-  // DEPRECATED.
-  // TODO(ajm): Remove after Chromium has upgraded to using Initialize().
-  virtual int set_sample_rate_hz(int rate) = 0;
-  // TODO(ajm): Remove after voice engine no longer requires it to resample
-  // the reverse stream to the forward rate.
-  virtual int input_sample_rate_hz() const = 0;
-  // TODO(ajm): Remove after Chromium no longer depends on it.
-  virtual int sample_rate_hz() const = 0;
-
   // TODO(ajm): Only intended for internal use. Make private and friend the
   // necessary classes?
   virtual int proc_sample_rate_hz() const = 0;
   virtual int proc_split_sample_rate_hz() const = 0;
   virtual int num_input_channels() const = 0;
   virtual int num_output_channels() const = 0;
   virtual int num_reverse_channels() const = 0;
 
   // Set to true when the output of AudioProcessing will be muted or in some
   // other way not used. Ideally, the captured audio would still be processed,
   // but some components may change behavior based on this information.
   // Default false.
   virtual void set_output_will_be_muted(bool muted) = 0;
-  virtual bool output_will_be_muted() const = 0;
 
   // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
   // this is the near-end (or captured) audio.
   //
   // If needed for enabled functionality, any function with the set_stream_ tag
   // must be called prior to processing the current frame. Any getter function
   // with the stream_ tag which is needed should be called after processing.
   //
   // The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_|
   // members of |frame| must be valid. If changed from the previous call to this
@@ skipping 80 matching lines @@
   //   - t_capture is the time the first sample of a frame is captured by the
   //     audio hardware and t_pull is the time the same frame is passed to
   //     ProcessStream().
   virtual int set_stream_delay_ms(int delay) = 0;
   virtual int stream_delay_ms() const = 0;
   virtual bool was_stream_delay_set() const = 0;
 
   // Call to signal that a key press occurred (true) or did not occur (false)
   // with this chunk of audio.
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
-  virtual bool stream_key_pressed() const = 0;
 
   // Sets a delay |offset| in ms to add to the values passed in through
   // set_stream_delay_ms(). May be positive or negative.
   //
   // Note that this could cause an otherwise valid value passed to
   // set_stream_delay_ms() to return an error.
   virtual void set_delay_offset_ms(int offset) = 0;
   virtual int delay_offset_ms() const = 0;
 
   // Starts recording debugging information to a file specified by |filename|,
@@ skipping 539 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_