Eliminate defines in talk/

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 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 583 matching lines @@
   webrtc::EcModes ec_mode = webrtc::kEcConference;
   webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
   webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
   webrtc::NsModes ns_mode = webrtc::kNsHighSuppression;
   if (options.aecm_generate_comfort_noise) {
     LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
                     << *options.aecm_generate_comfort_noise
                     << " (default is false).";
   }
 
-#if defined(IOS)
+#if defined(WEBRTC_IOS)
   // On iOS, VPIO provides built-in EC and AGC.
   options.echo_cancellation = rtc::Optional<bool>(false);
   options.auto_gain_control = rtc::Optional<bool>(false);
   LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead.";
 #elif defined(ANDROID)
   ec_mode = webrtc::kEcAecm;
 #endif
 
-#if defined(IOS) || defined(ANDROID)
+#if defined(WEBRTC_IOS) || defined(ANDROID)
   // Set the AGC mode for iOS as well despite disabling it above, to avoid
   // unsupported configuration errors from webrtc.
   agc_mode = webrtc::kAgcFixedDigital;
   options.typing_detection = rtc::Optional<bool>(false);
   options.experimental_agc = rtc::Optional<bool>(false);
   options.extended_filter_aec = rtc::Optional<bool>(false);
   options.experimental_ns = rtc::Optional<bool>(false);
 #endif
 
   // Delay Agnostic AEC automatically turns on EC if not set except on iOS
   // where the feature is not supported.
   bool use_delay_agnostic_aec = false;
-#if !defined(IOS)
+#if !defined(WEBRTC_IOS)
   if (options.delay_agnostic_aec) {
     use_delay_agnostic_aec = *options.delay_agnostic_aec;
     if (use_delay_agnostic_aec) {
       options.echo_cancellation = rtc::Optional<bool>(true);
       options.extended_filter_aec = rtc::Optional<bool>(true);
       ec_mode = webrtc::kEcConference;
     }
   }
 #endif
 
@@ skipping 217 matching lines @@
             *options.playout_sample_rate)) {
       LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
     }
   }
 
   return true;
 }
 
 void WebRtcVoiceEngine::SetDefaultDevices() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-#if !defined(IOS)
+#if !defined(WEBRTC_IOS)
   int in_id = kDefaultAudioDeviceId;
   int out_id = kDefaultAudioDeviceId;
   LOG(LS_INFO) << "Setting microphone to (id=" << in_id
                << ") and speaker to (id=" << out_id << ")";
 
   bool ret = true;
   if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
     LOG_RTCERR1(SetRecordingDevice, in_id);
     ret = false;
   }
   webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
   if (ap) {
     ap->Initialize();
   }
 
   if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
     LOG_RTCERR1(SetPlayoutDevice, out_id);
     ret = false;
   }
 
   if (ret) {
     LOG(LS_INFO) << "Set microphone to (id=" << in_id
                  << ") and speaker to (id=" << out_id << ")";
   }
-#endif  // !IOS
+#endif  // !WEBRTC_IOS
 }
 
 bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   unsigned int ulevel;
   if (voe_wrapper_->volume()->GetSpeakerVolume(ulevel) == -1) {
     LOG_RTCERR1(GetSpeakerVolume, level);
     return false;
   }
   *level = ulevel;
@@ skipping 1609 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE