Fix errors

plugins/AudioFileProcessor/AudioFileProcessor.cpp

@@ -276,7 +276,7 @@ QString AudioFileProcessor::nodeName() const
 
 
 
-auto AudioFileProcessor::beatLen(NotePlayHandle* note) const -> int
+auto AudioFileProcessor::beatLen(NotePlayHandle* note) const -> f_cnt_t
 {
 	// If we can play indefinitely, use the default beat note duration
 	if (static_cast<Sample::Loop>(m_loopModel.value()) != Sample::Loop::Off) { return 0; }
@@ -292,7 +292,7 @@ auto AudioFileProcessor::beatLen(NotePlayHandle* note) const -> int
 		: m_nextPlayStartPoint;
 	const auto duration = m_sample.endFrame() - startFrame;
 
-	return static_cast<int>(std::floor(duration * freqFactor));
+	return static_cast<f_cnt_t>(std::floor(duration * freqFactor));
 }
 
 

plugins/AudioFileProcessor/AudioFileProcessor.h

@@ -54,7 +54,7 @@ class AudioFileProcessor : public Instrument
 
 	QString nodeName() const override;
 
-	auto beatLen(NotePlayHandle* note) const -> int override;
+	auto beatLen(NotePlayHandle* note) const -> f_cnt_t override;
 
 	float desiredReleaseTimeMs() const override
 	{
@@ -89,7 +89,7 @@ private slots:
 
 
 signals:
-	void isPlaying( lmms::f_cnt_t _current_frame );
+	void isPlaying(f_cnt_t _current_frame);
 	void sampleUpdated();
 
 private:

plugins/AudioFileProcessor/AudioFileProcessorWaveView.cpp

@@ -27,6 +27,7 @@
 #include "ConfigManager.h"
 #include "gui_templates.h"
 #include "SampleWaveform.h"
+#include "lmms_basics.h"
 
 #include <QPainter>
 #include <QMouseEvent>
@@ -57,7 +58,7 @@ void AudioFileProcessorWaveView::setTo(f_cnt_t to)
 
 void AudioFileProcessorWaveView::setFrom(f_cnt_t from)
 {
-	m_from = std::max(from, 0);
+	m_from = std::max<f_cnt_t>(from, 0);
 }
 
 f_cnt_t AudioFileProcessorWaveView::range() const
@@ -347,19 +348,19 @@ void AudioFileProcessorWaveView::zoom(const bool out)
 	const f_cnt_t d_from = start - m_from;
 	const f_cnt_t d_to = m_to - end;
 
-	const f_cnt_t step = qMax(1, qMax(d_from, d_to) / 10);
+	const f_cnt_t step = std::max<f_cnt_t>(1, std::max(d_from, d_to) / 10);
 	const f_cnt_t step_from = (out ? - step : step);
 	const f_cnt_t step_to = (out ? step : - step);
 
-	const double comp_ratio = double(qMin(d_from, d_to))
-								/ qMax(1, qMax(d_from, d_to));
+	const double comp_ratio = static_cast<double>(std::min(d_from, d_to)) / std::max<f_cnt_t>(1, qMax(d_from, d_to));
 
-	const auto boundedFrom = std::clamp(m_from + step_from, 0, start);
-	const auto boundedTo = std::clamp(m_to + step_to, end, frames);
+	const auto boundedFrom = std::clamp<f_cnt_t>(m_from + step_from, 0, start);
+	const auto boundedTo = std::clamp<f_cnt_t>(m_to + step_to, end, frames);
 
 	const auto toStep = static_cast<f_cnt_t>(step_from * (boundedTo == m_to ? 1 : comp_ratio));
-	const auto newFrom
-		= (out && d_from < d_to) || (!out && d_to < d_from) ? boundedFrom : std::clamp(m_from + toStep, 0, start);
+	const auto newFrom = (out && d_from < d_to) || (!out && d_to < d_from)
+		? boundedFrom
+		: std::clamp<f_cnt_t>(m_from + toStep, 0, start);
 
 	const auto fromStep = static_cast<f_cnt_t>(step_to * (boundedFrom == m_from ? 1 : comp_ratio));
 	const auto newTo

plugins/Lb302/Lb302.cpp

@@ -29,6 +29,7 @@
  */
 
 // Need to include this first to ensure we get M_PI in MinGW with C++11
+#include "lmms_basics.h"
 #define _USE_MATH_DEFINES
 #include <cmath>
 
@@ -750,8 +751,8 @@ void Lb302Synth::playNote( NotePlayHandle * _n, SampleFrame* _working_buffer )
 		m_notes.prepend( _n );
 	}
 	m_notesMutex.unlock();
-	
-	release_frame = qMax( release_frame, _n->framesLeft() + _n->offset() );
+
+	release_frame = std::max<f_cnt_t>(release_frame, _n->framesLeft() + _n->offset());
 }
 
 

plugins/MultitapEcho/MultitapEcho.cpp

@@ -25,6 +25,7 @@
 
 #include "MultitapEcho.h"
 #include "embed.h"
+#include "lmms_basics.h"
 #include "plugin_export.h"
 
 namespace lmms
@@ -118,8 +119,8 @@ bool MultitapEchoEffect::processAudioBuffer( SampleFrame* buf, const fpp_t frame
 	}
 
 	// add dry buffer - never swap inputs for dry
-	m_buffer.writeAddingMultiplied( buf, 0, frames, dryGain );
-	
+	m_buffer.writeAddingMultiplied(buf, static_cast<f_cnt_t>(0), frames, dryGain);
+
 	// swapped inputs?
 	if( swapInputs )
 	{
@@ -176,4 +177,4 @@ PLUGIN_EXPORT Plugin * lmms_plugin_main( Model* parent, void* data )
 }
 
 
-} // namespace lmms
+} // namespace lmms

plugins/Vibed/VibratingString.cpp

@@ -26,6 +26,7 @@
 #include "interpolation.h"
 #include "AudioEngine.h"
 #include "Engine.h"
+#include "lmms_basics.h"
 
 #include <algorithm>
 #include <cstdlib>
@@ -99,7 +100,7 @@ void VibratingString::resample(const float* src, f_cnt_t srcFrames, f_cnt_t dstF
 	{
 		const float srcFrameFloat = frame * static_cast<float>(srcFrames) / dstFrames;
 		const float fracPos = srcFrameFloat - static_cast<f_cnt_t>(srcFrameFloat);
-		const f_cnt_t srcFrame = std::clamp(static_cast<f_cnt_t>(srcFrameFloat), 1, srcFrames - 3);
+		const f_cnt_t srcFrame = std::clamp<f_cnt_t>(static_cast<f_cnt_t>(srcFrameFloat), 1, srcFrames - 3);
 		m_impulse[frame] = cubicInterpolate(
 			src[srcFrame - 1],
 			src[srcFrame + 0],

src/core/AudioEngine.cpp

@@ -27,6 +27,7 @@
 #include "MixHelpers.h"
 #include "denormals.h"
 
+#include "lmms_basics.h"
 #include "lmmsconfig.h"
 
 #include "AudioEngineWorkerThread.h"
@@ -288,7 +289,7 @@ void AudioEngine::pushInputFrames( SampleFrame* _ab, const f_cnt_t _frames )
 
 	if( frames + _frames > size )
 	{
-		size = std::max(size * 2, frames + _frames);
+		size = std::max<f_cnt_t>(size * 2, frames + _frames);
 		auto ab = new SampleFrame[size];
 		memcpy( ab, buf, frames * sizeof( SampleFrame ) );
 		delete [] buf;

src/core/Instrument.cpp

@@ -28,6 +28,7 @@
 
 #include "DummyInstrument.h"
 #include "InstrumentTrack.h"
+#include "lmms_basics.h"
 #include "lmms_constants.h"
 
 
@@ -185,7 +186,7 @@ void Instrument::applyRelease( SampleFrame* buf, const NotePlayHandle * _n )
 	const auto releaseFrames = desiredReleaseFrames();
 
 	const auto endFrame = _n->framesLeft();
-	const auto startFrame = std::max(0, endFrame - releaseFrames);
+	const auto startFrame = std::max<f_cnt_t>(0, endFrame - releaseFrames);
 
 	for (auto f = startFrame; f < endFrame && f < fpp; f++)
 	{

src/core/audio/AudioSdl.cpp

@@ -69,7 +69,7 @@ AudioSdl::AudioSdl( bool & _success_ful, AudioEngine*  _audioEngine ) :
 						// to convert the buffers
 #endif
 	m_audioHandle.channels = channels();
-	m_audioHandle.samples = std::max(1024, audioEngine()->framesPerPeriod() * 2);
+	m_audioHandle.samples = std::max<fpp_t>(1024, audioEngine()->framesPerPeriod() * 2);
 
 	m_audioHandle.callback = sdlAudioCallback;
 	m_audioHandle.userdata = this;