Removed unused API functions in AudioProcessing and AudioProcessingModule

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  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 263 matching lines @@
   }
   delete crit_;
   crit_ = NULL;
 }
 
 int AudioProcessingImpl::Initialize() {
   CriticalSectionScoped crit_scoped(crit_);
   return InitializeLocked();
 }
 
-int AudioProcessingImpl::set_sample_rate_hz(int rate) {
-  CriticalSectionScoped crit_scoped(crit_);
-
-  ProcessingConfig processing_config = api_format_;
-  processing_config.input_stream().set_sample_rate_hz(rate);
-  processing_config.output_stream().set_sample_rate_hz(rate);
-  return InitializeLocked(processing_config);
-}
-
 int AudioProcessingImpl::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) {
   const ProcessingConfig processing_config = {
       {{input_sample_rate_hz,
         ChannelsFromLayout(input_layout),
         LayoutHasKeyboard(input_layout)},
@@ skipping 165 matching lines @@
   for (auto item : component_list_) {
     item->SetExtraOptions(config);
   }
 
   if (transient_suppressor_enabled_ != config.Get<ExperimentalNs>().enabled) {
     transient_suppressor_enabled_ = config.Get<ExperimentalNs>().enabled;
     InitializeTransient();
   }
 }
 
-int AudioProcessingImpl::input_sample_rate_hz() const {
-  CriticalSectionScoped crit_scoped(crit_);
-  return api_format_.input_stream().sample_rate_hz();
-}
-
-int AudioProcessingImpl::sample_rate_hz() const {
-  CriticalSectionScoped crit_scoped(crit_);
-  return api_format_.input_stream().sample_rate_hz();
-}
 
 int AudioProcessingImpl::proc_sample_rate_hz() const {
   return fwd_proc_format_.sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
   return split_rate_;
 }
 
 int AudioProcessingImpl::num_reverse_channels() const {
   return rev_proc_format_.num_channels();
 }
 
 int AudioProcessingImpl::num_input_channels() const {
   return api_format_.input_stream().num_channels();
 }
 
 int AudioProcessingImpl::num_output_channels() const {
   return api_format_.output_stream().num_channels();
 }
 
 void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
   CriticalSectionScoped lock(crit_);
   output_will_be_muted_ = muted;
   if (agc_manager_.get()) {
     agc_manager_->SetCaptureMuted(output_will_be_muted_);
   }
 }
 
-bool AudioProcessingImpl::output_will_be_muted() const {
-  CriticalSectionScoped lock(crit_);
-  return output_will_be_muted_;
-}
 
 int AudioProcessingImpl::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) {
   CriticalSectionScoped crit_scoped(crit_);
   StreamConfig input_stream = api_format_.input_stream();
@@ skipping 374 matching lines @@
 }
 
 bool AudioProcessingImpl::was_stream_delay_set() const {
   return was_stream_delay_set_;
 }
 
 void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
   key_pressed_ = key_pressed;
 }
 
-bool AudioProcessingImpl::stream_key_pressed() const {
-  return key_pressed_;
-}
-
 void AudioProcessingImpl::set_delay_offset_ms(int offset) {
   CriticalSectionScoped crit_scoped(crit_);
   delay_offset_ms_ = offset;
 }
 
 int AudioProcessingImpl::delay_offset_ms() const {
   return delay_offset_ms_;
 }
 
 int AudioProcessingImpl::StartDebugRecording(
@@ skipping 323 matching lines @@
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc