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/92007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@301 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_device/main/source/Dummy/audio_device_dummy.h

@@ -155,8 +155,8 @@ private:
     EventWrapper&                                  _recStartEvent;
     EventWrapper&                                  _playStartEvent;
     
-    ThreadWrapper*                                 _ptrThreadPlay;
     ThreadWrapper*                                 _ptrThreadRec;
+    ThreadWrapper*                                 _ptrThreadPlay;
     WebRtc_UWord32                                _recThreadID;
     WebRtc_UWord32                                _playThreadID;
     

src/modules/audio_device/main/source/Dummy/audio_device_utility_dummy.cc

@@ -17,7 +17,8 @@ namespace webrtc
 {
 
 AudioDeviceUtilityDummy::AudioDeviceUtilityDummy(const WebRtc_Word32 id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id),
+    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+    _id(id),
     _lastError(AudioDeviceModule::kAdmErrNone)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,

src/modules/audio_device/main/source/Linux/audio_device_linux_alsa.cc

@@ -70,11 +70,19 @@ AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const WebRtc_Word32 id) :
     _handlePlayout(NULL),
     _recSndcardBuffsize(ALSA_SNDCARD_BUFF_SIZE_REC),
     _playSndcardBuffsize(ALSA_SNDCARD_BUFF_SIZE_PLAY),
+    _samplingFreqRec(REC_SAMPLES_PER_MS),
+    _samplingFreqPlay(PLAY_SAMPLES_PER_MS),
+    _recChannels(1),
+    _playChannels(1),
+    _playbackBufferSize(0),
+    _recordBufferSize(0),
+    _recBuffer(NULL),
+    _playBufType(AudioDeviceModule::kAdaptiveBufferSize),
     _initialized(false),
-    _recIsInitialized(false),
-    _playIsInitialized(false),
     _recording(false),
     _playing(false),
+    _recIsInitialized(false),
+    _playIsInitialized(false),
     _startRec(false),
     _stopRec(false),
     _startPlay(false),
@@ -83,11 +91,11 @@ AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const WebRtc_Word32 id) :
     _buffersizeFromZeroAvail(true),
     _buffersizeFromZeroDelay(true),
     _sndCardPlayDelay(0),
-    _sndCardRecDelay(0),
-    _numReadyRecSamples(0),
     _previousSndCardPlayDelay(0),
     _delayMonitorStatePlay(0),
     _largeDelayCountPlay(0),
+    _sndCardRecDelay(0),
+    _numReadyRecSamples(0),
     _bufferCheckMethodPlay(0),
     _bufferCheckMethodRec(0),
     _bufferCheckErrorsPlay(0),
@@ -98,16 +106,8 @@ AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const WebRtc_Word32 id) :
     _playError(0),
     _recWarning(0),
     _recError(0),
-    _samplingFreqRec(REC_SAMPLES_PER_MS),
-    _samplingFreqPlay(PLAY_SAMPLES_PER_MS),
-    _recChannels(1),
-    _playChannels(1),
-    _playbackBufferSize(0),
-    _recordBufferSize(0),
-    _recBuffer(NULL),
     _playBufDelay(80),
-    _playBufDelayFixed(80),
+    _playBufDelayFixed(80)
-    _playBufType(AudioDeviceModule::kAdaptiveBufferSize)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
                  "%s created", __FUNCTION__);

src/modules/audio_device/main/source/Linux/audio_device_linux_alsa.h

@@ -212,8 +212,8 @@ private:
     EventWrapper& _recStartEvent;
     EventWrapper& _playStartEvent;
 
-    ThreadWrapper* _ptrThreadPlay;
     ThreadWrapper* _ptrThreadRec;
+    ThreadWrapper* _ptrThreadPlay;
     WebRtc_UWord32 _recThreadID;
     WebRtc_UWord32 _playThreadID;
 
@@ -221,10 +221,6 @@ private:
 
     AudioMixerManagerLinuxALSA _mixerManager;
 
-    bool _usingInputDeviceIndex;
-    bool _usingOutputDeviceIndex;
-    AudioDeviceModule::WindowsDeviceType _inputDevice;
-    AudioDeviceModule::WindowsDeviceType _outputDevice;
     WebRtc_UWord16 _inputDeviceIndex;
     WebRtc_UWord16 _outputDeviceIndex;
     bool _inputDeviceIsSpecified;

src/modules/audio_device/main/source/Linux/audio_mixer_manager_linux_alsa.cc

@@ -25,9 +25,12 @@ namespace webrtc
 {
 
 AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA(const WebRtc_Word32 id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id),
+    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _inputMixerHandle(NULL), _outputMixerHandle(NULL),
+    _id(id),
-    _inputMixerElement(NULL), _outputMixerElement(NULL)
+    _outputMixerHandle(NULL),
+    _inputMixerHandle(NULL),
+    _outputMixerElement(NULL),
+    _inputMixerElement(NULL)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
                  "%s constructed", __FUNCTION__);

src/modules/audio_device/main/source/audio_device_buffer.cc

@@ -39,13 +39,13 @@ AudioDeviceBuffer::AudioDeviceBuffer() :
     _recBytesPerSample(0),
     _playBytesPerSample(0),
     _recSamples(0),
-    _playSamples(0),
     _recSize(0),
+    _playSamples(0),
     _playSize(0),
     _recFile(*FileWrapper::Create()),
     _playFile(*FileWrapper::Create()),
-    _newMicLevel(0),
     _currentMicLevel(0),
+    _newMicLevel(0),
     _playDelayMS(0),
     _recDelayMS(0),
     _clockDrift(0),

src/modules/audio_device/main/source/audio_device_impl.cc

@@ -172,14 +172,14 @@ AudioDeviceModuleImpl::AudioDeviceModuleImpl(const WebRtc_Word32 id, const Audio
     _critSectEventCb(*CriticalSectionWrapper::CreateCriticalSection()),
     _critSectAudioCb(*CriticalSectionWrapper::CreateCriticalSection()),
     _ptrCbAudioDeviceObserver(NULL),
+    _ptrAudioDeviceUtility(NULL),
+    _ptrAudioDevice(NULL),
     _id(id),
     _platformAudioLayer(audioLayer),
     _lastProcessTime(AudioDeviceUtility::GetTimeInMS()),
-    _lastError(kAdmErrNone),
     _platformType(kPlatformNotSupported),
-    _ptrAudioDeviceUtility(NULL),
+    _initialized(false),
-    _ptrAudioDevice(NULL),
+    _lastError(kAdmErrNone)
-    _initialized(false)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
 }

src/modules/audio_device/main/test/func_test_manager.cc

@@ -101,11 +101,20 @@ void AudioEventObserver::OnWarningIsReported(const WarningCode warning)
 ;
 
 AudioTransportImpl::AudioTransportImpl(AudioDeviceModule* audioDevice) :
-    _audioDevice(audioDevice), _playFromFile(false), _fullDuplex(false),
+    _audioDevice(audioDevice),
-        _speakerVolume(false), _microphoneVolume(false), _speakerMute(false),
+    _playFromFile(false),
-        _microphoneMute(false), _microphoneBoost(false),
+    _fullDuplex(false),
-        _loopBackMeasurements(false), _microphoneAGC(false), _recCount(0),
+    _speakerVolume(false),
-        _playCount(0), _playFile(*FileWrapper::Create()), _audioList()
+    _speakerMute(false),
+    _microphoneVolume(false),
+    _microphoneMute(false),
+    _microphoneBoost(false),
+    _microphoneAGC(false),
+    _loopBackMeasurements(false),
+    _playFile(*FileWrapper::Create()),
+    _recCount(0),
+    _playCount(0),
+    _audioList()
 {
     _resampler.Reset(48000, 48000, kResamplerSynchronousStereo);
 }
@@ -568,8 +577,10 @@ WebRtc_Word32 AudioTransportImpl::NeedMorePlayData(
 ;
 
 FuncTestManager::FuncTestManager() :
-    _audioDevice(NULL), _processThread(NULL), _audioEventObserver(NULL),
+    _processThread(NULL),
-        _audioTransport(NULL)
+    _audioDevice(NULL),
+    _audioEventObserver(NULL),
+    _audioTransport(NULL)
 {
 }
 ;