Removing the "initialized after" warnings.

This CL tweat the order of the initialization in the constructor to adapt to the order of declaration of the members.
Review URL: http://webrtc-codereview.appspot.com/99002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@294 4adac7df-926f-26a2-2b94-8c16560cd09d

src/voice_engine/main/source/channel.cc

@@ -1070,22 +1070,15 @@ Channel::Channel(const WebRtc_Word32 channelId,
     _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _transmitCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _channelId(channelId),
     _instanceId(instanceId),
-    _playing(false),
-    _sending(false),
-    _receiving(false),
-    _mixFileWithMicrophone(false),
-    _timeStamp(0), // This is just an offset, RTP module will add it's own random offset
+    _channelId(channelId),
     _rtpRtcpModule(*RtpRtcp::CreateRtpRtcp(VoEModuleId(
-            instanceId, channelId), true)),
+        instanceId, channelId), true)),
     _audioCodingModule(*AudioCodingModule::Create(
-            VoEModuleId(instanceId, channelId))),
+        VoEModuleId(instanceId, channelId))),
 #ifndef WEBRTC_EXTERNAL_TRANSPORT
-    _socketTransportModule(
-        *UdpTransport::Create(
-            VoEModuleId(instanceId, channelId),
-            numSocketThreads)),
+    _socketTransportModule(*UdpTransport::Create(
+        VoEModuleId(instanceId, channelId), numSocketThreads)),
 #endif
 #ifdef WEBRTC_SRTP
     _srtpModule(*SrtpModule::CreateSrtpModule(VoEModuleId(instanceId,
@@ -1093,30 +1086,9 @@ Channel::Channel(const WebRtc_Word32 channelId,
 #endif
     _rtpDumpIn(*RtpDump::CreateRtpDump()),
     _rtpDumpOut(*RtpDump::CreateRtpDump()),
-    _transportPtr(NULL),
-    _encryptionPtr(NULL),
-    _rxAudioProcessingModulePtr(NULL),
-#ifdef WEBRTC_DTMF_DETECTION
-    _telephoneEventDetectionPtr(NULL),
-#endif
-    _rxVadObserverPtr(NULL),
-    _oldVadDecision(-1),
-    _sendFrameType(0),
     _outputAudioLevel(),
-    _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
-    _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
-    _encrypting(false),
-    _decrypting(false),
-    _encryptionRTPBufferPtr(NULL),
-    _decryptionRTPBufferPtr(NULL),
-    _encryptionRTCPBufferPtr(NULL),
-    _decryptionRTCPBufferPtr(NULL),
     _externalTransport(false),
-    _engineStatisticsPtr(NULL),
-    _moduleProcessThreadPtr(NULL),
-    _audioDeviceModulePtr(NULL),
-    _voiceEngineObserverPtr(NULL),
-    _callbackCritSectPtr(NULL),
+    _audioLevel_dBov(100),
     _inputFilePlayerPtr(NULL),
     _outputFilePlayerPtr(NULL),
     _outputFileRecorderPtr(NULL),
@@ -1128,34 +1100,62 @@ Channel::Channel(const WebRtc_Word32 channelId,
     _inputFilePlaying(false),
     _outputFilePlaying(false),
     _outputFileRecording(false),
-    _outputExternalMedia(false),
+    _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
+    _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
     _inputExternalMedia(false),
+    _outputExternalMedia(false),
     _inputExternalMediaCallbackPtr(NULL),
     _outputExternalMediaCallbackPtr(NULL),
+    _encryptionRTPBufferPtr(NULL),
+    _decryptionRTPBufferPtr(NULL),
+    _encryptionRTCPBufferPtr(NULL),
+    _decryptionRTCPBufferPtr(NULL),
+    _timeStamp(0), // This is just an offset, RTP module will add it's own random offset
+    _sendTelephoneEventPayloadType(106),
+    _playoutTimeStampRTP(0),
+    _playoutTimeStampRTCP(0),
+    _numberOfDiscardedPackets(0),
+    _engineStatisticsPtr(NULL),
+    _moduleProcessThreadPtr(NULL),
+    _audioDeviceModulePtr(NULL),
+    _voiceEngineObserverPtr(NULL),
+    _callbackCritSectPtr(NULL),
+    _transportPtr(NULL),
+    _encryptionPtr(NULL),
+    _rxAudioProcessingModulePtr(NULL),
+#ifdef WEBRTC_DTMF_DETECTION
+    _telephoneEventDetectionPtr(NULL),
+#endif
+    _rxVadObserverPtr(NULL),
+    _oldVadDecision(-1),
+    _sendFrameType(0),
     _rtpObserverPtr(NULL),
     _rtcpObserverPtr(NULL),
+    _outputIsOnHold(false),
+    _externalPlayout(false),
+    _inputIsOnHold(false),
+    _playing(false),
+    _sending(false),
+    _receiving(false),
+    _mixFileWithMicrophone(false),
+    _rtpObserver(false),
+    _rtcpObserver(false),
     _mute(false),
     _panLeft(1.0f),
     _panRight(1.0f),
     _outputGain(1.0f),
+    _encrypting(false),
+    _decrypting(false),
     _playOutbandDtmfEvent(false),
     _playInbandDtmfEvent(false),
-    _sendTelephoneEventPayloadType(106),
     _inbandTelephoneEventDetection(false),
     _outOfBandTelephoneEventDetecion(false),
-    _rtpObserver(false),
-    _rtcpObserver(false),
-    _playoutTimeStampRTP(0),
-    _playoutTimeStampRTCP(0),
-    _numberOfDiscardedPackets(0),
     _extraPayloadType(0),
     _insertExtraRTPPacket(false),
     _extraMarkerBit(false),
     _lastLocalTimeStamp(0),
     _lastPayloadType(0),
-    _outputIsOnHold(false),
-    _externalPlayout(false),
-    _inputIsOnHold(false),
+    _includeAudioLevelIndication(false),
     _rtpPacketTimedOut(false),
     _rtpPacketTimeOutIsEnabled(false),
     _rtpTimeOutSeconds(0),
@@ -1171,9 +1171,7 @@ Channel::Channel(const WebRtc_Word32 channelId,
     _RxVadDetection(false),
     _rxApmIsEnabled(false),
     _rxAgcIsEnabled(false),
-    _rxNsIsEnabled(false),
-    _audioLevel_dBov(100),
-    _includeAudioLevelIndication(false)
+    _rxNsIsEnabled(false)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::Channel() - ctor");

src/voice_engine/main/source/channel.h

@@ -584,8 +584,8 @@ private:
     bool _outputFileRecording;
     DtmfInbandQueue _inbandDtmfQueue;
     DtmfInband _inbandDtmfGenerator;
-    bool _outputExternalMedia;
     bool _inputExternalMedia;
+    bool _outputExternalMedia;
     VoEMediaProcess* _inputExternalMediaCallbackPtr;
     VoEMediaProcess* _outputExternalMediaCallbackPtr;
     WebRtc_UWord8* _encryptionRTPBufferPtr;

src/voice_engine/main/source/dtmf_inband.cc

@@ -65,17 +65,17 @@ const WebRtc_Word16 Dtmf_dBm0kHz[37]=
 
 
 DtmfInband::DtmfInband(const WebRtc_Word32 id) :
-    _id(id),
     _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+    _id(id),
     _outputFrequencyHz(8000),
-    _reinit(true),
-    _remainingSamples(0),
     _frameLengthSamples(0),
+    _remainingSamples(0),
     _eventCode(0),
     _attenuationDb(0),
+    _lengthMs(0),
+    _reinit(true),
     _playing(false),
-    _delaySinceLastToneMS(1000),
-    _lengthMs(0)
+    _delaySinceLastToneMS(1000)
 {
     memset(_oldOutputLow, 0, sizeof(_oldOutputLow));
     memset(_oldOutputHigh, 0, sizeof(_oldOutputHigh));

src/voice_engine/main/source/monitor_module.cc

@@ -16,8 +16,8 @@ namespace webrtc  {
 namespace voe  {
 
 MonitorModule::MonitorModule() :
-    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _observerPtr(NULL),
+    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _lastProcessTime(GET_TIME_IN_MS())
 {
 }

src/voice_engine/main/source/output_mixer.cc

@@ -120,18 +120,18 @@ OutputMixer::Create(OutputMixer*& mixer, const WebRtc_UWord32 instanceId)
 OutputMixer::OutputMixer(const WebRtc_UWord32 instanceId) :
     _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _instanceId(instanceId),
-    _outputFileRecorderPtr(NULL),
-    _outputFileRecording(false),
-    _dtmfGenerator(instanceId),
     _mixerModule(*AudioConferenceMixer::
                  CreateAudioConferenceMixer(instanceId)),
-    _externalMediaCallbackPtr(NULL),
     _audioLevel(),
+    _dtmfGenerator(instanceId),
+    _instanceId(instanceId),
+    _externalMediaCallbackPtr(NULL),
     _externalMedia(false),
     _panLeft(1.0f),
     _panRight(1.0f),
-    _mixingFrequencyHz(8000)
+    _mixingFrequencyHz(8000),
+    _outputFileRecorderPtr(NULL),
+    _outputFileRecording(false)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::OutputMixer() - ctor");

src/voice_engine/main/source/shared_data.cc

@@ -25,13 +25,13 @@ static WebRtc_Word32 _gInstanceCounter = 0;
 
 SharedData::SharedData() :
     _instanceId(++_gInstanceCounter),
+    _apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
     _channelManager(_gInstanceCounter),
     _engineStatistics(_gInstanceCounter),
     _usingExternalAudioDevice(false),
     _audioDevicePtr(NULL),
     _audioProcessingModulePtr(NULL),
     _moduleProcessThreadPtr(ProcessThread::CreateProcessThread()),
-    _apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
     _externalRecording(false),
     _externalPlayout(false)
 {

src/voice_engine/main/source/statistics.cc

@@ -23,8 +23,8 @@ namespace voe {
 Statistics::Statistics(const WebRtc_UWord32 instanceId) :
     _critPtr(CriticalSectionWrapper::CreateCriticalSection()),
     _instanceId(instanceId),
-    _isInitialized(false),
-    _lastError(0)
+    _lastError(0),
+    _isInitialized(false)
 {
 }
 	

src/voice_engine/main/source/transmit_mixer.cc

@@ -162,18 +162,11 @@ TransmitMixer::Destroy(TransmitMixer*& mixer)
 }
 
 TransmitMixer::TransmitMixer(const WebRtc_UWord32 instanceId) :
-    _instanceId(instanceId),
     _engineStatisticsPtr(NULL),
     _channelManagerPtr(NULL),
     _audioProcessingModulePtr(NULL),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    _timeActive(0),
-    _penaltyCounter(0),
-    _typingNoiseWarning(0),
-#endif
+    _voiceEngineObserverPtr(NULL),
+    _processThreadPtr(NULL),
     _filePlayerPtr(NULL),
     _fileRecorderPtr(NULL),
     _fileCallRecorderPtr(NULL),
@@ -185,18 +178,24 @@ TransmitMixer::TransmitMixer(const WebRtc_UWord32 instanceId) :
     _filePlaying(false),
     _fileRecording(false),
     _fileCallRecording(false),
+    _audioLevel(),
+    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+    _timeActive(0),
+    _penaltyCounter(0),
+    _typingNoiseWarning(0),
+#endif
+    _saturationWarning(0),
+    _noiseWarning(0),
+    _instanceId(instanceId),
     _mixFileWithMicrophone(false),
     _captureLevel(0),
-    _audioLevel(),
     _externalMedia(false),
     _externalMediaCallbackPtr(NULL),
     _mute(false),
     _remainingMuteMicTimeMs(0),
     _mixingFrequency(0),
-    _voiceEngineObserverPtr(NULL),
-    _processThreadPtr(NULL),
-    _saturationWarning(0),
-    _noiseWarning(0),
     _includeAudioLevelIndication(false),
     _audioLevel_dBov(100)
 {