Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h

webrtc/voice_engine/transmit_mixer.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 16 matching lines @@
 namespace webrtc {
 namespace voe {
 
 // TODO(ajm): The thread safety of this is dubious...
 void
 TransmitMixer::OnPeriodicProcess()
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::OnPeriodicProcess()");
 
-#if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     bool send_typing_noise_warning = false;
     bool typing_noise_detected = false;
     {
       rtc::CritScope cs(&_critSect);
       if (_typingNoiseWarningPending) {
         send_typing_noise_warning = true;
         typing_noise_detected = _typingNoiseDetected;
         _typingNoiseWarningPending = false;
       }
     }
@@ skipping 136 matching lines @@
     _processThreadPtr(NULL),
     // Avoid conflict with other channels by adding 1024 - 1026,
     // won't use as much as 1024 channels.
     _filePlayerId(instanceId + 1024),
     _fileRecorderId(instanceId + 1025),
     _fileCallRecorderId(instanceId + 1026),
     _filePlaying(false),
     _fileRecording(false),
     _fileCallRecording(false),
     _audioLevel(),
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     _typingNoiseWarningPending(false),
     _typingNoiseDetected(false),
 #endif
     _saturationWarning(false),
     _instanceId(instanceId),
     _mixFileWithMicrophone(false),
     _captureLevel(0),
     external_postproc_ptr_(NULL),
     external_preproc_ptr_(NULL),
     _mute(false),
@@ skipping 130 matching lines @@
     }
 
     // --- Near-end audio processing.
     ProcessAudio(totalDelayMS, clockDrift, currentMicLevel, keyPressed);
 
     if (swap_stereo_channels_ && stereo_codec_)
       // Only bother swapping if we're using a stereo codec.
       AudioFrameOperations::SwapStereoChannels(&_audioFrame);
 
     // --- Annoying typing detection (utilizes the APM/VAD decision)
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     TypingDetection(keyPressed);
 #endif
 
     // --- Mute signal
     AudioFrameOperations::Mute(&_audioFrame, _mute, _mute);
 
     // --- Mix with file (does not affect the mixing frequency)
     if (_filePlaying)
     {
         MixOrReplaceAudioWithFile(_audioFrame.sample_rate_hz_);
@@ skipping 804 matching lines @@
   }
 
   // Store new capture level. Only updated when analog AGC is enabled.
   _captureLevel = agc->stream_analog_level();
 
   rtc::CritScope cs(&_critSect);
   // Triggers a callback in OnPeriodicProcess().
   _saturationWarning |= agc->stream_is_saturated();
 }
 
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
 void TransmitMixer::TypingDetection(bool keyPressed)
 {
   // We let the VAD determine if we're using this feature or not.
   if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) {
     return;
   }
 
   bool vadActive = _audioFrame.vad_activity_ == AudioFrame::kVadActive;
   if (_typingDetection.Process(keyPressed, vadActive)) {
     rtc::CritScope cs(&_critSect);
@@ skipping 10 matching lines @@
   }
 }
 #endif
 
 int TransmitMixer::GetMixingFrequency()
 {
     assert(_audioFrame.sample_rate_hz_ != 0);
     return _audioFrame.sample_rate_hz_;
 }
 
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
 int TransmitMixer::TimeSinceLastTyping(int &seconds)
 {
     // We check in VoEAudioProcessingImpl that this is only called when
     // typing detection is active.
     seconds = _typingDetection.TimeSinceLastDetectionInSeconds();
     return 0;
 }
 #endif
 
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
 int TransmitMixer::SetTypingDetectionParameters(int timeWindow,
                                                 int costPerTyping,
                                                 int reportingThreshold,
                                                 int penaltyDecay,
                                                 int typeEventDelay)
 {
     _typingDetection.SetParameters(timeWindow,
                                    costPerTyping,
                                    reportingThreshold,
                                    penaltyDecay,
                                    typeEventDelay,
                                    0);
     return 0;
 }
 #endif
 
 void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
   swap_stereo_channels_ = enable;
 }
 
 bool TransmitMixer::IsStereoChannelSwappingEnabled() {
   return swap_stereo_channels_;
 }
 
 }  // namespace voe
 }  // namespace webrtc