[#2402] Code indent

sflphone-common/src/account.cpp

@@ -43,14 +43,17 @@ Account::Account (const AccountID& accountID, std::string type) :
         , _ringtonePath ("/usr/share/sflphone/ringtones/konga.ul")
         , _ringtoneEnabled (true)
         , _displayName ("")
-        , _useragent ("SFLphone") {
+        , _useragent ("SFLphone")
+{
     setRegistrationState (Unregistered);
 }
 
-Account::~Account() {
+Account::~Account()
+{
 }
 
-void Account::loadConfig() {
+void Account::loadConfig()
+{
 
     // If IAX is not supported, do not register this account
 #ifndef USE_IAX
@@ -63,7 +66,8 @@ void Account::loadConfig() {
     loadAudioCodecs ();
 }
 
-void Account::setRegistrationState (RegistrationState state) {
+void Account::setRegistrationState (RegistrationState state)
+{
 
     if (state != _registrationState) {
         _debug ("Account: set registration state");
@@ -74,7 +78,8 @@ void Account::setRegistrationState (RegistrationState state) {
     }
 }
 
-void Account::loadAudioCodecs (void) {
+void Account::loadAudioCodecs (void)
+{
 
     // if the user never set the codec list, use the default configuration for this account
     if (_codecStr == "") {
@@ -90,7 +95,8 @@ void Account::loadAudioCodecs (void) {
     }
 }
 
-void Account::setActiveCodecs (const std::vector <std::string> &list) {
+void Account::setActiveCodecs (const std::vector <std::string> &list)
+{
 
     _codecOrder.clear();
     // list contains the ordered payload of active codecs picked by the user for this account

sflphone-common/src/accountcreator.cpp

@@ -36,15 +36,18 @@
 #include "iax/iaxaccount.h"
 #endif
 
-AccountCreator::AccountCreator() {
+AccountCreator::AccountCreator()
+{
 }
 
 
-AccountCreator::~AccountCreator() {
+AccountCreator::~AccountCreator()
+{
 }
 
 Account*
-AccountCreator::createAccount (AccountType type, AccountID accountID) {
+AccountCreator::createAccount (AccountType type, AccountID accountID)
+{
     switch (type) {
 
         case SIP_ACCOUNT: {

sflphone-common/src/audio/alsa/alsalayer.cpp

@@ -65,7 +65,8 @@ AlsaLayer::AlsaLayer (ManagerImpl* manager)
 }
 
 // Destructor
-AlsaLayer::~AlsaLayer (void) {
+AlsaLayer::~AlsaLayer (void)
+{
     _debug ("Audio: Destroy of ALSA layer");
     closeLayer();
 
@@ -76,7 +77,8 @@ AlsaLayer::~AlsaLayer (void) {
 }
 
 bool
-AlsaLayer::closeLayer() {
+AlsaLayer::closeLayer()
+{
     _debugAlsa ("Audio: Close ALSA streams");
 
     try {
@@ -103,7 +105,8 @@ AlsaLayer::closeLayer() {
 }
 
 bool
-AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize, int stream , std::string plugin) {
+AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize, int stream , std::string plugin)
+{
     /* Close the devices before open it */
     if (stream == SFL_PCM_BOTH && is_capture_open() == true && is_playback_open() == true) {
         closeCaptureStream();
@@ -146,7 +149,8 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
 }
 
 void
-AlsaLayer::startStream (void) {
+AlsaLayer::startStream (void)
+{
     _debug ("Audio: Start stream");
 
     if (_audiofilter)
@@ -209,7 +213,8 @@ AlsaLayer::startStream (void) {
 }
 
 void
-AlsaLayer::stopStream (void) {
+AlsaLayer::stopStream (void)
+{
     _debug ("Audio: Stop stream");
 
     try {
@@ -238,7 +243,8 @@ AlsaLayer::stopStream (void) {
 }
 
 
-bool AlsaLayer::isCaptureActive (void) {
+bool AlsaLayer::isCaptureActive (void)
+{
     ost::MutexLock guard (_mutex);
 
     if (_CaptureHandle)
@@ -248,7 +254,8 @@ bool AlsaLayer::isCaptureActive (void) {
 }
 
 
-void AlsaLayer::setEchoCancelState (bool state) {
+void AlsaLayer::setEchoCancelState (bool state)
+{
     // if a stream already running
     if (AudioLayer::_echoCancel)
         _echoCancel->setEchoCancelState (state);
@@ -256,7 +263,8 @@ void AlsaLayer::setEchoCancelState (bool state) {
     AudioLayer::_echocancelstate = state;
 }
 
-void AlsaLayer::setNoiseSuppressState (bool state) {
+void AlsaLayer::setNoiseSuppressState (bool state)
+{
     // if a stream already opened
     if (AudioLayer::_echoCancel)
         _echoCancel->setNoiseSuppressState (state);
@@ -270,7 +278,8 @@ void AlsaLayer::setNoiseSuppressState (bool state) {
 /////////////////   ALSA PRIVATE FUNCTIONS   ////////////////////////////////////////////////
 ////////////////////////////////////////////////////////////////////////////////////////////
 
-void AlsaLayer::stopCaptureStream (void) {
+void AlsaLayer::stopCaptureStream (void)
+{
     int err;
 
     if (_CaptureHandle) {
@@ -284,7 +293,8 @@ void AlsaLayer::stopCaptureStream (void) {
     }
 }
 
-void AlsaLayer::closeCaptureStream (void) {
+void AlsaLayer::closeCaptureStream (void)
+{
     int err;
 
     if (is_capture_prepared() == true && is_capture_running() == true)
@@ -300,7 +310,8 @@ void AlsaLayer::closeCaptureStream (void) {
     }
 }
 
-void AlsaLayer::startCaptureStream (void) {
+void AlsaLayer::startCaptureStream (void)
+{
     int err;
 
     if (_CaptureHandle && !is_capture_running()) {
@@ -313,7 +324,8 @@ void AlsaLayer::startCaptureStream (void) {
     }
 }
 
-void AlsaLayer::prepareCaptureStream (void) {
+void AlsaLayer::prepareCaptureStream (void)
+{
     int err;
 
     if (is_capture_open() && !is_capture_prepared()) {
@@ -326,7 +338,8 @@ void AlsaLayer::prepareCaptureStream (void) {
     }
 }
 
-void AlsaLayer::stopPlaybackStream (void) {
+void AlsaLayer::stopPlaybackStream (void)
+{
     int err;
 
     if (_RingtoneHandle && is_playback_running()) {
@@ -348,7 +361,8 @@ void AlsaLayer::stopPlaybackStream (void) {
 }
 
 
-void AlsaLayer::closePlaybackStream (void) {
+void AlsaLayer::closePlaybackStream (void)
+{
     int err;
 
     if (is_playback_prepared() == true && is_playback_running() == true)
@@ -373,7 +387,8 @@ void AlsaLayer::closePlaybackStream (void) {
 
 }
 
-void AlsaLayer::startPlaybackStream (void) {
+void AlsaLayer::startPlaybackStream (void)
+{
     int err;
 
     if (_PlaybackHandle && !is_playback_running()) {
@@ -386,7 +401,8 @@ void AlsaLayer::startPlaybackStream (void) {
     }
 }
 
-void AlsaLayer::preparePlaybackStream (void) {
+void AlsaLayer::preparePlaybackStream (void)
+{
     int err;
 
     if (is_playback_open() && !is_playback_prepared()) {
@@ -428,7 +444,8 @@ void AlsaLayer::recoverPlaybackStream(int error)
 */
 
 
-bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate) {
+bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
+{
 
     snd_pcm_hw_params_t *hwparams = NULL;
     snd_pcm_sw_params_t *swparams = NULL;
@@ -553,7 +570,8 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate) {
 
 
 bool
-AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r, int flag) {
+AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r, int flag)
+{
 
     int err;
 
@@ -623,7 +641,8 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
 
 //TODO	first frame causes broken pipe (underrun) because not enough data are send --> make the handle wait to be ready
 int
-AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle) {
+AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
+{
     if (_trigger_request == true) {
         _trigger_request = false;
         startPlaybackStream ();
@@ -664,7 +683,8 @@ AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle) {
 }
 
 int
-AlsaLayer::read (void* buffer, int toCopy) {
+AlsaLayer::read (void* buffer, int toCopy)
+{
     //ost::MutexLock lock( _mutex );
 
     int samples;
@@ -709,7 +729,8 @@ AlsaLayer::read (void* buffer, int toCopy) {
 }
 
 void
-AlsaLayer::handle_xrun_capture (void) {
+AlsaLayer::handle_xrun_capture (void)
+{
     _debugAlsa ("Audio: Handle xrun capture");
 
     snd_pcm_status_t* status;
@@ -728,7 +749,8 @@ AlsaLayer::handle_xrun_capture (void) {
 }
 
 void
-AlsaLayer::handle_xrun_playback (snd_pcm_t *handle) {
+AlsaLayer::handle_xrun_playback (snd_pcm_t *handle)
+{
     _debugAlsa ("Audio: Handle xrun playback");
 
     int state;
@@ -752,7 +774,8 @@ AlsaLayer::handle_xrun_playback (snd_pcm_t *handle) {
 }
 
 std::string
-AlsaLayer::buildDeviceTopo (std::string plugin, int card, int subdevice) {
+AlsaLayer::buildDeviceTopo (std::string plugin, int card, int subdevice)
+{
     std::stringstream ss,ss1;
     std::string pcm = plugin;
 
@@ -777,7 +800,8 @@ AlsaLayer::buildDeviceTopo (std::string plugin, int card, int subdevice) {
 }
 
 std::vector<std::string>
-AlsaLayer::getSoundCardsInfo (int stream) {
+AlsaLayer::getSoundCardsInfo (int stream)
+{
     std::vector<std::string> cards_id;
     HwIDPair p;
 
@@ -837,7 +861,8 @@ AlsaLayer::getSoundCardsInfo (int stream) {
 
 
 bool
-AlsaLayer::soundCardIndexExist (int card , int stream) {
+AlsaLayer::soundCardIndexExist (int card , int stream)
+{
     snd_ctl_t* handle;
     snd_pcm_info_t *pcminfo;
     snd_pcm_info_alloca (&pcminfo);
@@ -857,7 +882,8 @@ AlsaLayer::soundCardIndexExist (int card , int stream) {
 }
 
 int
-AlsaLayer::soundCardGetIndex (std::string description) {
+AlsaLayer::soundCardGetIndex (std::string description)
+{
     unsigned int i;
 
     for (i = 0 ; i < IDSoundCards.size() ; i++) {
@@ -871,7 +897,8 @@ AlsaLayer::soundCardGetIndex (std::string description) {
     return 0;
 }
 
-void AlsaLayer::audioCallback (void) {
+void AlsaLayer::audioCallback (void)
+{
 
     int toGet, urgentAvailBytes, normalAvailBytes, maxBytes;
     unsigned short spkrVolume, micVolume;
@@ -1117,7 +1144,8 @@ void AlsaLayer::audioCallback (void) {
     }
 }
 
-void* AlsaLayer::adjustVolume (void* buffer , int len, int stream) {
+void* AlsaLayer::adjustVolume (void* buffer , int len, int stream)
+{
     int vol, i, size;
     SFLDataFormat *src = NULL;
 

sflphone-common/src/audio/audiodevice.cpp

@@ -34,10 +34,12 @@
 const double AudioDevice::DEFAULT_RATE = 8000.0;
 
 AudioDevice::AudioDevice (int id, const std::string& name) :
-        _id (id), _name (name), _rate (DEFAULT_RATE) {
+        _id (id), _name (name), _rate (DEFAULT_RATE)
+{
     _rate = DEFAULT_RATE;
 }
 
-AudioDevice::~AudioDevice() {
+AudioDevice::~AudioDevice()
+{
 }
 

sflphone-common/src/audio/audiolayer.cpp

@@ -30,7 +30,8 @@
 
 #include "audiolayer.h"
 
-void AudioLayer::flushMain (void) {
+void AudioLayer::flushMain (void)
+{
     ost::MutexLock guard (_mutex);
 
     // should pass call id
@@ -41,13 +42,15 @@ void AudioLayer::flushMain (void) {
 }
 
 
-void AudioLayer::flushUrgent (void) {
+void AudioLayer::flushUrgent (void)
+{
     ost::MutexLock guard (_mutex);
     _urgentRingBuffer.flushAll();
 }
 
 
-int AudioLayer::putUrgent (void* buffer, int toCopy) {
+int AudioLayer::putUrgent (void* buffer, int toCopy)
+{
     int a;
 
     ost::MutexLock guard (_mutex);
@@ -62,7 +65,8 @@ int AudioLayer::putUrgent (void* buffer, int toCopy) {
     return 0;
 }
 
-int AudioLayer::putMain (void *buffer, int toCopy, CallID call_id) {
+int AudioLayer::putMain (void *buffer, int toCopy, CallID call_id)
+{
     int a;
 
     ost::MutexLock guard (_mutex);

sflphone-common/src/audio/audioloop.cpp

@@ -36,16 +36,19 @@
 #include <math.h>
 #include <strings.h>
 
-AudioLoop::AudioLoop() :_buffer (0),  _size (0), _pos (0), _sampleRate (0) {
+AudioLoop::AudioLoop() :_buffer (0),  _size (0), _pos (0), _sampleRate (0)
+{
 }
 
-AudioLoop::~AudioLoop() {
+AudioLoop::~AudioLoop()
+{
     delete [] _buffer;
     _buffer = 0;
 }
 
 int
-AudioLoop::getNext (SFLDataFormat* output, int nb, short volume) {
+AudioLoop::getNext (SFLDataFormat* output, int nb, short volume)
+{
     int copied = 0;
     int block;
     int pos = _pos;

sflphone-common/src/audio/audioprocessing.cpp

@@ -38,29 +38,34 @@ AudioProcessing::AudioProcessing (Algorithm *_algo) : _algorithm (_algo) {}
 
 AudioProcessing::~AudioProcessing (void) {}
 
-void AudioProcessing::resetAlgorithm (void) {
+void AudioProcessing::resetAlgorithm (void)
+{
     if (_algorithm)
         _algorithm->reset();
 }
 
-int AudioProcessing::getData (SFLDataFormat *outputData) {
+int AudioProcessing::getData (SFLDataFormat *outputData)
+{
     if (_algorithm)
         return _algorithm->getData (outputData);
     else
         return 0;
 }
 
-void AudioProcessing::putData (SFLDataFormat *inputData, int nbBytes) {
+void AudioProcessing::putData (SFLDataFormat *inputData, int nbBytes)
+{
     if (_algorithm)
         _algorithm->putData (inputData, nbBytes);
 }
 
-void AudioProcessing::processAudio (SFLDataFormat *inputData, int nbBytes) {
+void AudioProcessing::processAudio (SFLDataFormat *inputData, int nbBytes)
+{
     if (_algorithm)
         _algorithm->process (inputData, nbBytes);
 }
 
-int AudioProcessing::processAudio (SFLDataFormat *inputData, SFLDataFormat *outputData, int nbBytes) {
+int AudioProcessing::processAudio (SFLDataFormat *inputData, SFLDataFormat *outputData, int nbBytes)
+{
     if (_algorithm)
         return _algorithm->process (inputData, outputData, nbBytes);
     else
@@ -68,7 +73,8 @@ int AudioProcessing::processAudio (SFLDataFormat *inputData, SFLDataFormat *outp
 }
 
 
-void AudioProcessing::processAudio (SFLDataFormat *micData, SFLDataFormat *spkrData, SFLDataFormat *outputData, int nbBytes) {
+void AudioProcessing::processAudio (SFLDataFormat *micData, SFLDataFormat *spkrData, SFLDataFormat *outputData, int nbBytes)
+{
     if (_algorithm)
         _algorithm->process (micData, spkrData, outputData, nbBytes);
 }

sflphone-common/src/audio/audiorecord.cpp

@@ -49,7 +49,8 @@ struct wavhdr {
 };
 
 
-AudioRecord::AudioRecord() {
+AudioRecord::AudioRecord()
+{
 
     sndSmplRate_ = 8000;
     channels_ = 1;
@@ -65,18 +66,21 @@ AudioRecord::AudioRecord() {
     spkBuffer_ = new SFLDataFormat[nbSamplesMax_];
 }
 
-AudioRecord::~AudioRecord() {
+AudioRecord::~AudioRecord()
+{
     delete [] mixBuffer_;
     delete [] micBuffer_;
     delete [] spkBuffer_;
 }
 
 
-void AudioRecord::setSndSamplingRate (int smplRate) {
+void AudioRecord::setSndSamplingRate (int smplRate)
+{
     sndSmplRate_ = smplRate;
 }
 
-void AudioRecord::setRecordingOption (FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate, std::string path) {
+void AudioRecord::setRecordingOption (FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate, std::string path)
+{
 
 
     fileType_ = type;
@@ -90,7 +94,8 @@ void AudioRecord::setRecordingOption (FILE_TYPE type, SOUND_FORMAT format, int s
 
 
 
-void AudioRecord::initFileName (std::string peerNumber) {
+void AudioRecord::initFileName (std::string peerNumber)
+{
 
     std::string fName;
 
@@ -112,7 +117,8 @@ void AudioRecord::initFileName (std::string peerNumber) {
     savePath_.append (fName);
 }
 
-void AudioRecord::openFile() {
+void AudioRecord::openFile()
+{
 
     bool result = false;
 
@@ -139,7 +145,8 @@ void AudioRecord::openFile() {
 }
 
 
-void AudioRecord::closeFile() {
+void AudioRecord::closeFile()
+{
 
     if (fp == 0) return;
 
@@ -153,7 +160,8 @@ void AudioRecord::closeFile() {
 }
 
 
-bool AudioRecord::isOpenFile() {
+bool AudioRecord::isOpenFile()
+{
 
     if (fp) {
         return true;
@@ -163,7 +171,8 @@ bool AudioRecord::isOpenFile() {
 }
 
 
-bool AudioRecord::isFileExist() {
+bool AudioRecord::isFileExist()
+{
     _info ("AudioRecord: Try to open name : %s ", fileName_);
 
     if (fopen (fileName_,"rb") ==0) {
@@ -173,7 +182,8 @@ bool AudioRecord::isFileExist() {
     return false;
 }
 
-bool AudioRecord::isRecording() {
+bool AudioRecord::isRecording()
+{
 
     if (recordingEnabled_)
         return true;
@@ -182,7 +192,8 @@ bool AudioRecord::isRecording() {
 }
 
 
-bool AudioRecord::setRecording() {
+bool AudioRecord::setRecording()
+{
 
     if (isOpenFile()) {
         if (!recordingEnabled_) {
@@ -203,7 +214,8 @@ bool AudioRecord::setRecording() {
 
 }
 
-void AudioRecord::stopRecording() {
+void AudioRecord::stopRecording()
+{
     _info ("AudioRecording: Stop recording");
 
     if (recordingEnabled_)
@@ -211,7 +223,8 @@ void AudioRecord::stopRecording() {
 }
 
 
-void AudioRecord::createFilename() {
+void AudioRecord::createFilename()
+{
 
     time_t rawtime;
 
@@ -263,7 +276,8 @@ void AudioRecord::createFilename() {
     _info ("AudioRecord: create filename for this call %s ", fileName_);
 }
 
-bool AudioRecord::setRawFile() {
+bool AudioRecord::setRawFile()
+{
 
     fp = fopen (savePath_.c_str(), "wb");
 
@@ -283,7 +297,8 @@ bool AudioRecord::setRawFile() {
 }
 
 
-bool AudioRecord::setWavFile() {
+bool AudioRecord::setWavFile()
+{
 
     fp = fopen (savePath_.c_str(), "wb");
 
@@ -326,7 +341,8 @@ bool AudioRecord::setWavFile() {
 }
 
 
-bool AudioRecord::openExistingRawFile() {
+bool AudioRecord::openExistingRawFile()
+{
     fp = fopen (fileName_, "ab+");
 
     if (!fp) {
@@ -338,7 +354,8 @@ bool AudioRecord::openExistingRawFile() {
 }
 
 
-bool AudioRecord::openExistingWavFile() {
+bool AudioRecord::openExistingWavFile()
+{
     _info ("AudioRecord: Open existing wave file");
 
     fp = fopen (fileName_, "rb+");
@@ -380,7 +397,8 @@ bool AudioRecord::openExistingWavFile() {
 }