Removed unused API functions in AudioProcessing and AudioProcessingModule

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 100 matching lines @@
       webrtc::AudioProcessing::ChannelLayout input_layout,
       webrtc::AudioProcessing::ChannelLayout output_layout,
       webrtc::AudioProcessing::ChannelLayout reverse_layout));
   WEBRTC_STUB(Initialize, (
       const webrtc::ProcessingConfig& processing_config));
 
   WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
     experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
   }
 
-  WEBRTC_STUB(set_sample_rate_hz, (int rate));
-  WEBRTC_STUB_CONST(input_sample_rate_hz, ());
-  WEBRTC_STUB_CONST(sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
   WEBRTC_STUB_CONST(num_input_channels, ());
   WEBRTC_STUB_CONST(num_output_channels, ());
   WEBRTC_STUB_CONST(num_reverse_channels, ());
   WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
-  WEBRTC_BOOL_STUB_CONST(output_will_be_muted, ());
   WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
   WEBRTC_STUB(ProcessStream, (
       const float* const* src,
       size_t samples_per_channel,
       int input_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout input_layout,
       int output_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout output_layout,
       float* const* dest));
   WEBRTC_STUB(ProcessStream,
@@ skipping 10 matching lines @@
       webrtc::AudioProcessing::ChannelLayout layout));
   WEBRTC_STUB(ProcessReverseStream,
               (const float* const* src,
                const webrtc::StreamConfig& reverse_input_config,
                const webrtc::StreamConfig& reverse_output_config,
                float* const* dest));
   WEBRTC_STUB(set_stream_delay_ms, (int delay));
   WEBRTC_STUB_CONST(stream_delay_ms, ());
   WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
   WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
-  WEBRTC_BOOL_STUB_CONST(stream_key_pressed, ());
   WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
   WEBRTC_STUB_CONST(delay_offset_ms, ());
   WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
   WEBRTC_STUB(StartDebugRecording, (FILE* handle));
   WEBRTC_STUB(StopDebugRecording, ());
   WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
   webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
   webrtc::EchoControlMobile* echo_control_mobile() const override {
     return NULL;
   }
@@ skipping 986 matching lines @@
   int playout_sample_rate_;
   DtmfInfo dtmf_info_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_