Removes unused APIs from the ADM (part II)

webrtc/pc/test/fakeaudiocapturemodule.h

 /*
  *  Copyright 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 96 matching lines @@
   int32_t InitSpeaker() override;
   bool SpeakerIsInitialized() const override;
   int32_t InitMicrophone() override;
   bool MicrophoneIsInitialized() const override;
 
   int32_t SpeakerVolumeIsAvailable(bool* available) override;
   int32_t SetSpeakerVolume(uint32_t volume) override;
   int32_t SpeakerVolume(uint32_t* volume) const override;
   int32_t MaxSpeakerVolume(uint32_t* max_volume) const override;
   int32_t MinSpeakerVolume(uint32_t* min_volume) const override;
-  int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override;
 
   int32_t MicrophoneVolumeIsAvailable(bool* available) override;
   int32_t SetMicrophoneVolume(uint32_t volume) override;
   int32_t MicrophoneVolume(uint32_t* volume) const override;
   int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
 
   int32_t MinMicrophoneVolume(uint32_t* min_volume) const override;
-  int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override;
 
   int32_t SpeakerMuteIsAvailable(bool* available) override;
   int32_t SetSpeakerMute(bool enable) override;
   int32_t SpeakerMute(bool* enabled) const override;
 
   int32_t MicrophoneMuteIsAvailable(bool* available) override;
   int32_t SetMicrophoneMute(bool enable) override;
   int32_t MicrophoneMute(bool* enabled) const override;
 
-  int32_t MicrophoneBoostIsAvailable(bool* available) override;
-  int32_t SetMicrophoneBoost(bool enable) override;
-  int32_t MicrophoneBoost(bool* enabled) const override;
-
   int32_t StereoPlayoutIsAvailable(bool* available) const override;
   int32_t SetStereoPlayout(bool enable) override;
   int32_t StereoPlayout(bool* enabled) const override;
   int32_t StereoRecordingIsAvailable(bool* available) const override;
   int32_t SetStereoRecording(bool enable) override;
   int32_t StereoRecording(bool* enabled) const override;
   int32_t SetRecordingChannel(const ChannelType channel) override;
   int32_t RecordingChannel(ChannelType* channel) const override;
 
-  int32_t SetPlayoutBuffer(const BufferType type,
-                           uint16_t size_ms = 0) override;
-  int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override;
   int32_t PlayoutDelay(uint16_t* delay_ms) const override;
   int32_t RecordingDelay(uint16_t* delay_ms) const override;
 
-  int32_t CPULoad(uint16_t* load) const override;
-
-  int32_t StartRawOutputFileRecording(
-      const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
-  int32_t StopRawOutputFileRecording() override;
-  int32_t StartRawInputFileRecording(
-      const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
-  int32_t StopRawInputFileRecording() override;
-
   int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
   int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;
   int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override;
   int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override;
 
-  int32_t ResetAudioDevice() override;
   int32_t SetLoudspeakerStatus(bool enable) override;
   int32_t GetLoudspeakerStatus(bool* enabled) const override;
   bool BuiltInAECIsAvailable() const override { return false; }
   int32_t EnableBuiltInAEC(bool enable) override { return -1; }
   bool BuiltInAGCIsAvailable() const override { return false; }
   int32_t EnableBuiltInAGC(bool enable) override { return -1; }
   bool BuiltInNSIsAvailable() const override { return false; }
   int32_t EnableBuiltInNS(bool enable) override { return -1; }
 #if defined(WEBRTC_IOS)
   int GetPlayoutAudioParameters(
@@ skipping 88 matching lines @@
 
   // Protects variables that are accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_;
   // Protects |audio_callback_| that is accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_callback_;
 };
 
 #endif  // WEBRTC_PC_TEST_FAKEAUDIOCAPTUREMODULE_H_