Removed unused API functions in AudioProcessing and AudioProcessingModule

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 50 matching lines @@
   // AudioProcessing methods.
   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;
-  int set_sample_rate_hz(int rate) override;
-  int input_sample_rate_hz() const override;
-  int sample_rate_hz() const override;
   int proc_sample_rate_hz() const override;
   int proc_split_sample_rate_hz() const override;
   int num_input_channels() const override;
   int num_output_channels() const override;
   int num_reverse_channels() const override;
   void set_output_will_be_muted(bool muted) override;
-  bool output_will_be_muted() const override;
   int ProcessStream(AudioFrame* frame) override;
   int ProcessStream(const float* const* src,
                     size_t samples_per_channel,
                     int input_sample_rate_hz,
                     ChannelLayout input_layout,
                     int output_sample_rate_hz,
                     ChannelLayout output_layout,
                     float* const* dest) override;
   int ProcessStream(const float* const* src,
                     const StreamConfig& input_config,
                     const StreamConfig& output_config,
                     float* const* dest) override;
   int AnalyzeReverseStream(AudioFrame* frame) override;
   int ProcessReverseStream(AudioFrame* frame) override;
   int AnalyzeReverseStream(const float* const* data,
                            size_t samples_per_channel,
                            int sample_rate_hz,
                            ChannelLayout layout) override;
   int ProcessReverseStream(const float* const* src,
                            const StreamConfig& reverse_input_config,
                            const StreamConfig& reverse_output_config,
                            float* const* dest) override;
   int set_stream_delay_ms(int delay) override;
   int stream_delay_ms() const override;
   bool was_stream_delay_set() const override;
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
-  bool stream_key_pressed() const override;
   int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
   int StartDebugRecording(FILE* handle) override;
   int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
   void UpdateHistogramsOnCallEnd() override;
   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 85 matching lines @@
   rtc::scoped_ptr<Beamformer<float>> beamformer_;
   const std::vector<Point> array_geometry_;
 
   bool intelligibility_enabled_;
   rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_