Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h

webrtc/voice_engine/transmit_mixer.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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
 #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
 
 #include <memory>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_processing/typing_detection.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/monitor_module.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
+#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
+#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1
+#else
+#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0
+#endif
+
 namespace webrtc {
 
 class AudioProcessing;
 class ProcessThread;
 class VoEExternalMedia;
 class VoEMediaProcess;
 
 namespace voe {
 
 class ChannelManager;
@@ skipping 106 matching lines @@
     void PlayNotification(int32_t id,
                           uint32_t durationMs);
 
     void RecordNotification(int32_t id,
                             uint32_t durationMs);
 
     void PlayFileEnded(int32_t id);
 
     void RecordFileEnded(int32_t id);
 
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     // Typing detection
     int TimeSinceLastTyping(int &seconds);
     int SetTypingDetectionParameters(int timeWindow,
                                      int costPerTyping,
                                      int reportingThreshold,
                                      int penaltyDecay,
                                      int typeEventDelay);
 #endif
 
   void EnableStereoChannelSwapping(bool enable);
@@ skipping 11 matching lines @@
                             size_t nChannels,
                             int samplesPerSec);
     int32_t RecordAudioToFile(uint32_t mixingFrequency);
 
     int32_t MixOrReplaceAudioWithFile(
         int mixingFrequency);
 
     void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level,
                       bool key_pressed);
 
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     void TypingDetection(bool keyPressed);
 #endif
 
     // uses
     Statistics* _engineStatisticsPtr;
     ChannelManager* _channelManagerPtr;
     AudioProcessing* audioproc_;
     VoiceEngineObserver* _voiceEngineObserverPtr;
     ProcessThread* _processThreadPtr;
 
     // owns
     MonitorModule _monitorModule;
     AudioFrame _audioFrame;
     PushResampler<int16_t> resampler_;  // ADM sample rate -> mixing rate
     std::unique_ptr<FilePlayer> file_player_;
     std::unique_ptr<FileRecorder> file_recorder_;
     std::unique_ptr<FileRecorder> file_call_recorder_;
     int _filePlayerId;
     int _fileRecorderId;
     int _fileCallRecorderId;
     bool _filePlaying;
     bool _fileRecording;
     bool _fileCallRecording;
     voe::AudioLevel _audioLevel;
     // protect file instances and their variables in MixedParticipants()
     rtc::CriticalSection _critSect;
     rtc::CriticalSection _callbackCritSect;
 
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     webrtc::TypingDetection _typingDetection;
     bool _typingNoiseWarningPending;
     bool _typingNoiseDetected;
 #endif
     bool _saturationWarning;
 
     int _instanceId;
     bool _mixFileWithMicrophone;
     uint32_t _captureLevel;
     VoEMediaProcess* external_postproc_ptr_;
     VoEMediaProcess* external_preproc_ptr_;
     bool _mute;
     bool stereo_codec_;
     bool swap_stereo_channels_;
 };
 
 }  // namespace voe
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H