clang-tidy: Apply modernize-avoid-c-arrays everywhere

include/DataFile.h

@@ -143,14 +143,6 @@ class LMMS_EXPORT DataFile : public QDomDocument
 
 	void loadData( const QByteArray & _data, const QString & _sourceFile );
 
-
-	struct LMMS_EXPORT typeDescStruct
-	{
-		Type m_type;
-		QString m_name;
-	} ;
-	static typeDescStruct s_types[TypeCount];
-
 	QString m_fileName; //!< The origin file name or "" if this DataFile didn't originate from a file
 	QDomElement m_content;
 	QDomElement m_head;

include/InstrumentFunctions.h

@@ -52,6 +52,7 @@ class InstrumentFunctionNoteStacking : public Model, public JournallingObject
 
 public:
 	static const int MAX_CHORD_POLYPHONY = 13;
+	static const int NUM_CHORD_TABLES = 95;
 
 private:
 	using ChordSemiTones = std::array<int8_t, MAX_CHORD_POLYPHONY>;
@@ -129,7 +130,7 @@ class InstrumentFunctionNoteStacking : public Model, public JournallingObject
 			ChordSemiTones m_semiTones;
 		};
 
-		static Init s_initTable[];
+		static std::array<Init, NUM_CHORD_TABLES> s_initTable;
 
 	public:
 		static const ChordTable & getInstance()

include/Oscillator.h

@@ -255,7 +255,7 @@ class LMMS_EXPORT Oscillator
 	static fftwf_plan s_fftPlan;
 	static fftwf_plan s_ifftPlan;
 	static fftwf_complex * s_specBuf;
-	static float s_sampleBuffer[OscillatorConstants::WAVETABLE_LENGTH];
+	static std::array<float, OscillatorConstants::WAVETABLE_LENGTH> s_sampleBuffer;
 
 	static void generateSawWaveTable(int bands, sample_t* table, int firstBand = 1);
 	static void generateTriangleWaveTable(int bands, sample_t* table, int firstBand = 1);

include/PianoRoll.h

@@ -343,7 +343,7 @@ protected slots:
 	static QPixmap * s_toolOpen;
 	static QPixmap* s_toolKnife;
 
-	static PianoRollKeyTypes prKeyOrder[];
+	static std::array<PianoRollKeyTypes, 12> prKeyOrder;
 
 	static TextFloat * s_textFloat;
 

include/ProjectRenderer.h

@@ -76,7 +76,7 @@ class LMMS_EXPORT ProjectRenderer : public QThread
 
 	static QString getFileExtensionFromFormat( ExportFileFormats fmt );
 
-	static const FileEncodeDevice fileEncodeDevices[];
+	static const std::array<FileEncodeDevice, 5> fileEncodeDevices;
 
 public slots:
 	void startProcessing();

plugins/Amplifier/Amplifier.cpp

@@ -74,7 +74,7 @@ bool AmplifierEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 	double outSum = 0.0;
 	const float d = dryLevel();
 	const float w = wetLevel();
-	
+
 	const ValueBuffer * volBuf = m_ampControls.m_volumeModel.valueBuffer();
 	const ValueBuffer * panBuf = m_ampControls.m_panModel.valueBuffer();
 	const ValueBuffer * leftBuf = m_ampControls.m_leftModel.valueBuffer();
@@ -83,8 +83,8 @@ bool AmplifierEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 	for( fpp_t f = 0; f < frames; ++f )
 	{
 //		qDebug( "offset %d, value %f", f, m_ampControls.m_volumeModel.value( f ) );
-	
-		sample_t s[2] = { buf[f][0], buf[f][1] };
+
+		auto s = std::array{buf[f][0], buf[f][1]};
 
 		// vol knob
 		if( volBuf )
@@ -99,8 +99,8 @@ bool AmplifierEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 		}
 
 		// convert pan values to left/right values
-		const float pan = panBuf 
-			? panBuf->value( f ) 
+		const float pan = panBuf
+			? panBuf->value( f )
 			: m_ampControls.m_panModel.value();
 		const float left1 = pan <= 0
 			? 1.0
@@ -111,12 +111,12 @@ bool AmplifierEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 
 		// second stage amplification
 		const float left2 = leftBuf
-			? leftBuf->value( f ) 
+			? leftBuf->value( f )
 			: m_ampControls.m_leftModel.value();
 		const float right2 = rightBuf
-			? rightBuf->value( f ) 
+			? rightBuf->value( f )
 			: m_ampControls.m_rightModel.value();
-			
+
 		s[0] *= left1 * left2 * 0.01;
 		s[1] *= right1 * right2 * 0.01;
 

plugins/BassBooster/BassBooster.cpp

@@ -102,7 +102,7 @@ bool BassBoosterEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames
 		m_bbFX.leftFX().setGain( gain );
 		m_bbFX.rightFX().setGain( gain);
 
-		sample_t s[2] = { buf[f][0], buf[f][1] };
+		auto s = std::array{buf[f][0], buf[f][1]};
 		m_bbFX.nextSample( s[0], s[1] );
 
 		buf[f][0] = d * buf[f][0] + w * s[0];

plugins/Bitcrush/Bitcrush.cpp

@@ -35,8 +35,8 @@ const int OS_RATE = 5;
 const float OS_RATIO = 1.0f / OS_RATE;
 const float CUTOFF_RATIO = 0.353553391f;
 const int SILENCEFRAMES = 10;
-const float OS_RESAMPLE [5] = { 0.0001490062883964112, 0.1645978376763992, 0.6705063120704088,
-		0.1645978376763992, 0.0001490062883964112 };
+const auto OS_RESAMPLE = std::array{0.0001490062883964112f, 0.1645978376763992f, 0.6705063120704088f,
+		0.1645978376763992f, 0.0001490062883964112f };
 
 extern "C"
 {
@@ -65,13 +65,13 @@ BitcrushEffect::BitcrushEffect( Model * parent, const Descriptor::SubPluginFeatu
 	m_buffer = MM_ALLOC<sampleFrame>( Engine::audioEngine()->framesPerPeriod() * OS_RATE );
 	m_filter.setLowpass( m_sampleRate * ( CUTOFF_RATIO * OS_RATIO ) );
 	m_needsUpdate = true;
-	
+
 	m_bitCounterL = 0.0f;
 	m_bitCounterR = 0.0f;
-	
+
 	m_left = 0.0f;
 	m_right = 0.0f;
-	
+
 	m_silenceCounter = 0;
 }
 
@@ -125,7 +125,7 @@ bool BitcrushEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 
 		m_rateCoeffL = ( m_sampleRate * OS_RATE ) / ( rate - diff );
 		m_rateCoeffR = ( m_sampleRate * OS_RATE ) / ( rate + diff );
-		
+
 		m_bitCounterL = 0.0f;
 		m_bitCounterR = 0.0f;
 	}
@@ -147,9 +147,9 @@ bool BitcrushEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 		m_outClip = dbfsToAmp( m_controls.m_outClip.value() );
 	}
 	m_needsUpdate = false;
-	
+
 	const float noiseAmt = m_controls.m_inNoise.value() * 0.01f;
-	
+
 	// read input buffer and write it to oversampled buffer
 	if( m_rateEnabled ) // rate crushing enabled so do that
 	{
@@ -164,15 +164,15 @@ bool BitcrushEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 				if( m_bitCounterL > m_rateCoeffL )
 				{
 					m_bitCounterL -= m_rateCoeffL;
-					m_left = m_depthEnabled 
-						? depthCrush( buf[f][0] * m_inGain + noise( buf[f][0] * noiseAmt ) ) 
+					m_left = m_depthEnabled
+						? depthCrush( buf[f][0] * m_inGain + noise( buf[f][0] * noiseAmt ) )
 						: buf[f][0] * m_inGain + noise( buf[f][0] * noiseAmt );
 				}
 				if( m_bitCounterR > m_rateCoeffR )
 				{
 					m_bitCounterR -= m_rateCoeffR;
-					m_right = m_depthEnabled 
-						? depthCrush( buf[f][1] * m_inGain + noise( buf[f][1] * noiseAmt ) ) 
+					m_right = m_depthEnabled
+						? depthCrush( buf[f][1] * m_inGain + noise( buf[f][1] * noiseAmt ) )
 						: buf[f][1] * m_inGain + noise( buf[f][1] * noiseAmt );
 				}
 			}
@@ -185,17 +185,17 @@ bool BitcrushEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 			for( int o = 0; o < OS_RATE; ++o )
 			{
 				m_buffer[f * OS_RATE + o][0] = m_depthEnabled
-					? depthCrush( buf[f][0] * m_inGain + noise( buf[f][0] * noiseAmt ) ) 
+					? depthCrush( buf[f][0] * m_inGain + noise( buf[f][0] * noiseAmt ) )
 					: buf[f][0] * m_inGain + noise( buf[f][0] * noiseAmt );
 				m_buffer[f * OS_RATE + o][1] = m_depthEnabled
-					? depthCrush( buf[f][1] * m_inGain + noise( buf[f][1] * noiseAmt ) ) 
+					? depthCrush( buf[f][1] * m_inGain + noise( buf[f][1] * noiseAmt ) )
 					: buf[f][1] * m_inGain + noise( buf[f][1] * noiseAmt );
 			}
 		}
 	}
-	
+
 	// the oversampled buffer is now written, so filter it to reduce aliasing
-	
+
 	for( int f = 0; f < frames * OS_RATE; ++f )
 	{
 		if( qMax( qAbs( m_buffer[f][0] ), qAbs( m_buffer[f][1] ) ) >= 1.0e-10f )
@@ -218,10 +218,10 @@ bool BitcrushEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 			}
 		}
 	}
-	
-	
+
+
 	// now downsample and write it back to main buffer
-	
+
 	double outSum = 0.0;
 	const float d = dryLevel();
 	const float w = wetLevel();
@@ -238,7 +238,7 @@ bool BitcrushEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 		buf[f][1] = d * buf[f][1] + w * qBound( -m_outClip, rsum, m_outClip ) * m_outGain;
 		outSum += buf[f][0]*buf[f][0] + buf[f][1]*buf[f][1];
 	}
-	
+
 	checkGate( outSum / frames );
 
 	return isRunning();

plugins/Compressor/Compressor.cpp

@@ -276,7 +276,7 @@ bool CompressorEffect::processAudioBuffer(sampleFrame* buf, const fpp_t frames)
 	float rOutPeak = 0.0;
 	float lInPeak = 0.0;
 	float rInPeak = 0.0;
-	
+
 	const bool midside = m_compressorControls.m_midsideModel.value();
 	const bool peakmode = m_compressorControls.m_peakmodeModel.value();
 	const float inBalance = m_compressorControls.m_inBalanceModel.value();
@@ -292,8 +292,8 @@ bool CompressorEffect::processAudioBuffer(sampleFrame* buf, const fpp_t frames)
 
 	for(fpp_t f = 0; f < frames; ++f)
 	{
-		sample_t drySignal[2] = {buf[f][0], buf[f][1]};
-		sample_t s[2] = {drySignal[0] * m_inGainVal, drySignal[1] * m_inGainVal};
+		auto drySignal = std::array{buf[f][0], buf[f][1]};
+		auto s = std::array{drySignal[0] * m_inGainVal, drySignal[1] * m_inGainVal};
 
 		// Calculate tilt filters, to bias the sidechain to the low or high frequencies
 		if (m_tiltVal)
@@ -512,7 +512,7 @@ bool CompressorEffect::processAudioBuffer(sampleFrame* buf, const fpp_t frames)
 				m_inputBufLoc = 0;
 			}
 
-			const float temp[2] = {drySignal[0], drySignal[1]};
+			const auto temp = std::array{drySignal[0], drySignal[1]};
 			s[0] = m_inputBuf[0][m_inputBufLoc];
 			s[1] = m_inputBuf[1][m_inputBufLoc];
 
@@ -525,8 +525,8 @@ bool CompressorEffect::processAudioBuffer(sampleFrame* buf, const fpp_t frames)
 			s[1] = drySignal[1];
 		}
 
-		float delayedDrySignal[2] = {s[0], s[1]};
-		
+		auto delayedDrySignal = std::array{s[0], s[1]};
+
 		if (midside)// Convert left/right to mid/side
 		{
 			const float temp = s[0];

plugins/Compressor/CompressorControlDialog.cpp

@@ -206,7 +206,7 @@ CompressorControlDialog::CompressorControlDialog(CompressorControls* controls) :
 	peakButton->setActiveGraphic(PLUGIN_NAME::getIconPixmap("peak_sel"));
 	peakButton->setInactiveGraphic(PLUGIN_NAME::getIconPixmap("peak_unsel"));
 	peakButton->setToolTip(tr("Use absolute value of the input"));
-	
+
 	rmsPeakGroup = new automatableButtonGroup(this);
 	rmsPeakGroup->addButton(rmsButton);
 	rmsPeakGroup->addButton(peakButton);
@@ -221,7 +221,7 @@ CompressorControlDialog::CompressorControlDialog(CompressorControls* controls) :
 	midSideButton->setActiveGraphic(PLUGIN_NAME::getIconPixmap("midside_sel"));
 	midSideButton->setInactiveGraphic(PLUGIN_NAME::getIconPixmap("midside_unsel"));
 	midSideButton->setToolTip(tr("Compress mid and side audio"));
-	
+
 	leftRightMidSideGroup = new automatableButtonGroup(this);
 	leftRightMidSideGroup->addButton(leftRightButton);
 	leftRightMidSideGroup->addButton(midSideButton);
@@ -514,8 +514,8 @@ void CompressorControlDialog::redrawKnee()
 	{
 		m_p.setPen(QPen(m_kneeColor2, 3));
 
-		float prevPoint[2] = {kneePoint1, kneePoint1};
-		float newPoint[2] = {0, 0};
+		auto prevPoint = std::array{kneePoint1, kneePoint1};
+		auto newPoint = std::array{0.f, 0.f};
 
 		// Draw knee curve using many straight lines.
 		for (int i = 0; i < COMP_KNEE_LINES; ++i)
@@ -542,7 +542,7 @@ void CompressorControlDialog::redrawKnee()
 	m_p.end();
 
 	m_p.begin(&m_kneePixmap2);
-	
+
 	m_p.setCompositionMode(QPainter::CompositionMode_Source);
 	m_p.fillRect(0, 0, m_windowSizeX, m_kneeWindowSizeY, QColor("transparent"));
 	m_p.setCompositionMode(QPainter::CompositionMode_SourceOver);

plugins/Delay/DelayEffect.cpp

@@ -91,7 +91,7 @@ bool DelayEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 	const float sr = Engine::audioEngine()->processingSampleRate();
 	const float d = dryLevel();
 	const float w = wetLevel();
-	sample_t dryS[2];
+	auto dryS = std::array<sample_t, 2>{};
 	float lPeak = 0.0;
 	float rPeak = 0.0;
 	float length = m_delayControls.m_delayTimeModel.value();

plugins/DualFilter/DualFilter.cpp

@@ -134,8 +134,8 @@ bool DualFilterEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames
 	const bool enabled1 = m_dfControls.m_enabled1Model.value();
 	const bool enabled2 = m_dfControls.m_enabled2Model.value();
 
-	
-	
+
+
 
 	// buffer processing loop
 	for( fpp_t f = 0; f < frames; ++f )
@@ -145,9 +145,9 @@ bool DualFilterEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames
 		const float mix1 = 1.0f - mix2;
 		const float gain1 = *gain1Ptr * 0.01f;
 		const float gain2 = *gain2Ptr * 0.01f;
-		sample_t s[2] = { 0.0f, 0.0f };	// mix
-		sample_t s1[2] = { buf[f][0], buf[f][1] };	// filter 1
-		sample_t s2[2] = { buf[f][0], buf[f][1] };	// filter 2
+		auto s = std::array{0.0f, 0.0f};	// mix
+		auto s1 = std::array{buf[f][0], buf[f][1]};	// filter 1
+		auto s2 = std::array{buf[f][0], buf[f][1]};	// filter 2
 
 		// update filter 1
 		if( enabled1 )

plugins/DynamicsProcessor/DynamicsProcessor.cpp

@@ -105,17 +105,17 @@ bool DynProcEffect::processAudioBuffer( sampleFrame * _buf,
 // variables for effect
 	int i = 0;
 
-	float sm_peak[2] = { 0.0f, 0.0f };
+	auto sm_peak = std::array{0.0f, 0.0f};
 	float gain;
 
 	double out_sum = 0.0;
 	const float d = dryLevel();
 	const float w = wetLevel();
-	
+
 	const int stereoMode = m_dpControls.m_stereomodeModel.value();
 	const float inputGain = m_dpControls.m_inputModel.value();
 	const float outputGain = m_dpControls.m_outputModel.value();
-	
+
 	const float * samples = m_dpControls.m_wavegraphModel.samples();
 
 // debug code
@@ -143,7 +143,7 @@ bool DynProcEffect::processAudioBuffer( sampleFrame * _buf,
 
 	for( fpp_t f = 0; f < _frames; ++f )
 	{
-		double s[2] = { _buf[f][0], _buf[f][1] };
+		auto s = std::array{_buf[f][0], _buf[f][1]};
 
 // apply input gain
 		s[0] *= inputGain;
@@ -211,7 +211,7 @@ bool DynProcEffect::processAudioBuffer( sampleFrame * _buf,
 					gain = samples[199];
 				};
 
-				s[i] *= gain; 
+				s[i] *= gain;
 				s[i] /= sm_peak[i];
 			}
 		}

plugins/DynamicsProcessor/DynamicsProcessorControls.cpp

@@ -116,7 +116,7 @@ void DynProcControls::saveSettings( QDomDocument & _doc,
 
 void DynProcControls::setDefaultShape()
 {
-	float shp [200] = { };
+	auto shp = std::array<float, 200>{};
 	for ( int i = 0; i<200; i++)
 	{
 		shp[i] = ((float)i + 1.0f) / 200.0f;

plugins/Eq/EqEffect.cpp

@@ -71,7 +71,7 @@ bool EqEffect::processAudioBuffer( sampleFrame *buf, const fpp_t frames )
 	//wet/dry controls
 	const float dry = dryLevel();
 	const float wet = wetLevel();
-	sample_t dryS[2];
+	auto dryS = std::array<sample_t, 2>{};
 	// setup sample exact controls
 	float hpRes = m_eqControls.m_hpResModel.value();
 	float lowShelfRes = m_eqControls.m_lowShelfResModel.value();