Style and switching to FIR implementation of third octave filter bank

src/Audiosignals/VAAudiofileSignalSource.cpp

@@ -24,13 +24,13 @@ CVAAudiofileSignalSource::CVAAudiofileSignalSource( int iBufferLength, double dS
 
 	m_sfAudioBuffer.init( 2, iBufferLength, true );
 	std::vector< float* > vfBufferPointer;
-	vfBufferPointer.push_back( m_sfAudioBuffer[0].data() );
-	vfBufferPointer.push_back( m_sfAudioBuffer[1].data() );
+	vfBufferPointer.push_back( m_sfAudioBuffer[ 0 ].data() );
+	vfBufferPointer.push_back( m_sfAudioBuffer[ 1 ].data() );
 	m_pBufferDataSource = new ITABufferDatasource( vfBufferPointer, iBufferLength, dSampleRate, iBlockLength );
 
 	m_pBufferDataSource->SetPaused( true );
 	m_pBufferDataSource->SetLoopMode( false );
-	
+
 	SetDatasourceDelegatorTarget( m_pBufferDataSource );
 }
 
@@ -44,33 +44,33 @@ CVAAudiofileSignalSource::CVAAudiofileSignalSource( const std::string& sFileName
 {
 	m_sbOutBuffer.Init( iBlockLength, true );
 
-	CITAAudioSample sfFileBuffer(dSampleRate);
-	sfFileBuffer.LoadWithSampleRateConversion(sFileName);
-		
+	CITAAudioSample sfFileBuffer( dSampleRate );
+	sfFileBuffer.LoadWithSampleRateConversion( sFileName );
+
 	std::vector< float* > vfBufferPointer;
-	
+
 	int iGivenChannels = sfFileBuffer.channels();
 	if( iGivenChannels == 1 )
 	{
 		m_sfAudioBuffer.init( 2, sfFileBuffer.length(), false );
 		m_sfAudioBuffer[ 0 ].write( sfFileBuffer[ 0 ], sfFileBuffer.length() );
 		m_sfAudioBuffer[ 1 ].Zero();
-		vfBufferPointer.push_back( m_sfAudioBuffer[0].data() );
-		vfBufferPointer.push_back( m_sfAudioBuffer[1].data() );
+		vfBufferPointer.push_back( m_sfAudioBuffer[ 0 ].data() );
+		vfBufferPointer.push_back( m_sfAudioBuffer[ 1 ].data() );
 	}
 	else
 	{
 		VA_WARN( "VABufferSignalSource", "More than one channel in file '" << sFileName << "' found, using first." );
 		m_sfAudioBuffer = sfFileBuffer;
 
-		for( int i=0; i<m_sfAudioBuffer.channels(); i++ )
-			vfBufferPointer.push_back( m_sfAudioBuffer[i].data() );
+		for( int i = 0; i < m_sfAudioBuffer.channels(); i++ )
+			vfBufferPointer.push_back( m_sfAudioBuffer[ i ].data() );
 	}
 
 	m_pBufferDataSource = new ITABufferDatasource( vfBufferPointer, m_sfAudioBuffer.length(), dSampleRate, iBlockLength );
 	m_pBufferDataSource->SetPaused( true );
 	m_pBufferDataSource->SetLoopMode( false );
-	
+
 	SetDatasourceDelegatorTarget( m_pBufferDataSource );
 }
 
@@ -108,16 +108,16 @@ void CVAAudiofileSignalSource::SetPlaybackAction( int iPlayStateAction )
 
 void CVAAudiofileSignalSource::SetCursorSeconds( double dPos )
 {
-	double dDuration = m_pBufferDataSource->GetCapacity()/m_pBufferDataSource->GetSampleRate();
+	double dDuration = m_pBufferDataSource->GetCapacity() / m_pBufferDataSource->GetSampleRate();
 	int iPos;
-	if (dPos < 0) 
+	if( dPos < 0 )
 		iPos = 0;
-	else if (dPos > dDuration)
+	else if( dPos > dDuration )
 		iPos = m_pBufferDataSource->GetCapacity();
 	else
-		iPos = (int) floor( dPos * m_pBufferDataSource->GetSampleRate() );
+		iPos = ( int ) floor( dPos * m_pBufferDataSource->GetSampleRate() );
 
-	m_pBufferDataSource->SetCursor(iPos);
+	m_pBufferDataSource->SetCursor( iPos );
 }
 
 int CVAAudiofileSignalSource::GetType() const
@@ -143,16 +143,16 @@ std::string CVAAudiofileSignalSource::GetStateString() const
 std::string CVAAudiofileSignalSource::GetStateString( int iPlayState ) const
 {
 	std::string s;
-	
+
 	if( iPlayState == IVACore::VA_PLAYBACK_STATE_PLAYING )
 		m_pBufferDataSource->GetLoopMode() ? s = "playing and looping" : s = "playing";
 	else if( iPlayState == IVACore::VA_PLAYBACK_STATE_PAUSED )
 		s = "paused";
 	else if( iPlayState == IVACore::VA_PLAYBACK_STATE_STOPPED )
-			s = "stopped";
+		s = "stopped";
 	else
 		s = "unkown";
-	
+
 	return s;
 }
 
@@ -161,7 +161,7 @@ IVACore* CVAAudiofileSignalSource::GetAssociatedCore() const
 	return m_pAssociatedCore;
 }
 
-void CVAAudiofileSignalSource::HandleRegistration(IVACore* pParentCore)
+void CVAAudiofileSignalSource::HandleRegistration( IVACore* pParentCore )
 {
 	m_pAssociatedCore = pParentCore;
 }
@@ -175,12 +175,12 @@ const float* CVAAudiofileSignalSource::GetStreamBlock( const CVAAudiostreamState
 {
 	const int iNewPlayAction = m_iRequestedPlaybackAction;
 	const bool bSyncMod = pStreamInfo->bSyncMod; // If true, hold changes back until false
-	
+
 	int iResidualSamples = m_pBufferDataSource->GetCursor() - m_pBufferDataSource->GetCapacity();
 	bool bEndOfFileReached = ( iResidualSamples >= 0 );
 
 	int iNewPlaybackState = IVACore::VA_PLAYBACK_STATE_INVALID;
-	switch ( m_iCurrentPlayState )
+	switch( m_iCurrentPlayState )
 	{
 
 	case IVACore::VA_PLAYBACK_STATE_STOPPED:
@@ -243,7 +243,7 @@ const float* CVAAudiofileSignalSource::GetStreamBlock( const CVAAudiostreamState
 		// Do not apply any changes during synced modification, but process if required
 		if( bSyncMod )
 		{
-			 // Special case: if running into EOF, state change may be required
+			// Special case: if running into EOF, state change may be required
 			if( bEndOfFileReached )
 			{
 				m_pBufferDataSource->Rewind();
@@ -292,13 +292,13 @@ const float* CVAAudiofileSignalSource::GetStreamBlock( const CVAAudiostreamState
 		{
 			m_pBufferDataSource->Rewind();
 			m_pBufferDataSource->SetPaused( false );
-			iNewPlaybackState =IVACore::VA_PLAYBACK_STATE_PLAYING;
+			iNewPlaybackState = IVACore::VA_PLAYBACK_STATE_PLAYING;
 		}
 		else if( iNewPlayAction == IVACore::VA_PLAYBACK_ACTION_PAUSE )
 		{
 			m_pBufferDataSource->Rewind();
 			m_pBufferDataSource->SetPaused( true );
-			iNewPlaybackState =IVACore::VA_PLAYBACK_STATE_PAUSED;
+			iNewPlaybackState = IVACore::VA_PLAYBACK_STATE_PAUSED;
 		}
 		else if( iNewPlayAction == IVACore::VA_PLAYBACK_ACTION_STOP )
 		{
@@ -321,7 +321,7 @@ const float* CVAAudiofileSignalSource::GetStreamBlock( const CVAAudiostreamState
 		assert( iNewPlaybackState != IVACore::VA_PLAYBACK_STATE_INVALID );
 		m_iCurrentPlayState = iNewPlaybackState;
 	}
-	
+
 	// Process audio file signal source: copy samples from buffer
 	if( m_iCurrentPlayState == IVACore::VA_PLAYBACK_STATE_PLAYING )
 	{

src/Rendering/Binaural/FreeField/VABinauralFreeFieldAudioRenderer.cpp

@@ -1124,8 +1124,8 @@ void CVABinauralFreeFieldAudioRenderer::UpdateGlobalAuralizationMode( int iGloba
 CVABFFSoundPath::CVABFFSoundPath( double dSamplerate, int iBlocklength, int iHRIRFilterLength, int iDirFilterLength )
 {
 	pThirdOctaveFilterBank = CITAThirdOctaveFilterbank::Create( dSamplerate, iBlocklength,
-		//CITAThirdOctaveFilterbank::FIR_SPLINE_LINEAR_PHASE);
-		CITAThirdOctaveFilterbank::IIR_BIQUADS_ORDER10 );
+		CITAThirdOctaveFilterbank::FIR_SPLINE_LINEAR_PHASE);
+		//CITAThirdOctaveFilterbank::IIR_BIQUADS_ORDER10 );
 	pThirdOctaveFilterBank->SetIdentity();
 
 	float fReserverdMaxDelaySamples = (float) ( 3 * dSamplerate ); // 3 Sekunden ~ 1km Entfernung