Remove global oversampling (#7228)

Oversampling can have many different effects to the audio signal such as latency, phase issues, clipping, smearing, etc, so this should really be an option on a per-plugin basis, not globally across all of LMMS (which, in some places, shouldn't really need to oversample at all but were oversampled anyways).
LMMS · May 5, 2024 · 9b6e33a · 9b6e33a
1 parent 9bdc8ad
commit 9b6e33a
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Showing 75 changed files with 155 additions and 373 deletions.
diff --git a/include/AudioDevice.h b/include/AudioDevice.h
@@ -89,8 +89,6 @@ class AudioDevice
 
 	virtual void stopProcessing();
 
-	void applyQualitySettings();
-
 protected:
 	// subclasses can re-implement this for being used in conjunction with
 	// processNextBuffer()
@@ -110,13 +108,6 @@ class AudioDevice
 	void clearS16Buffer( int_sample_t * _outbuf,
 							const fpp_t _frames );
 
-	// resample given buffer from samplerate _src_sr to samplerate _dst_sr
-	fpp_t resample( const surroundSampleFrame * _src,
-					const fpp_t _frames,
-					surroundSampleFrame * _dst,
-					const sample_rate_t _src_sr,
-					const sample_rate_t _dst_sr );
-
 	inline void setSampleRate( const sample_rate_t _new_sr )
 	{
 		m_sampleRate = _new_sr;
@@ -142,9 +133,6 @@ class AudioDevice
 
 	QMutex m_devMutex;
 
-	SRC_DATA m_srcData;
-	SRC_STATE * m_srcState;
-
 	surroundSampleFrame * m_buffer;
 
 };

diff --git a/include/AudioDummy.h b/include/AudioDummy.h
@@ -104,7 +104,7 @@ class AudioDummy : public QThread, public AudioDevice
 				delete[] b;
 			}
 
-			const int microseconds = static_cast<int>( audioEngine()->framesPerPeriod() * 1000000.0f / audioEngine()->processingSampleRate() - timer.elapsed() );
+			const int microseconds = static_cast<int>( audioEngine()->framesPerPeriod() * 1000000.0f / audioEngine()->outputSampleRate() - timer.elapsed() );
 			if( microseconds > 0 )
 			{
 				usleep( microseconds );

diff --git a/include/AudioEngine.h b/include/AudioEngine.h
@@ -108,13 +108,6 @@ class LMMS_EXPORT AudioEngine : public QObject
 
 	struct qualitySettings
 	{
-		enum class Mode
-		{
-			Draft,
-			HighQuality,
-			FinalMix
-		} ;
-
 		enum class Interpolation
 		{
 			Linear,
@@ -123,53 +116,11 @@ class LMMS_EXPORT AudioEngine : public QObject
 			SincBest
 		} ;
 
-		enum class Oversampling
-		{
-			None,
-			X2,
-			X4,
-			X8
-		} ;
-
 		Interpolation interpolation;
-		Oversampling oversampling;
 
-		qualitySettings(Mode m)
+		qualitySettings(Interpolation i) :
+			interpolation(i)
 		{
-			switch (m)
-			{
-				case Mode::Draft:
-					interpolation = Interpolation::Linear;
-					oversampling = Oversampling::None;
-					break;
-				case Mode::HighQuality:
-					interpolation =
-						Interpolation::SincFastest;
-					oversampling = Oversampling::X2;
-					break;
-				case Mode::FinalMix:
-					interpolation = Interpolation::SincBest;
-					oversampling = Oversampling::X8;
-					break;
-			}
-		}
-
-		qualitySettings(Interpolation i, Oversampling o) :
-			interpolation(i),
-			oversampling(o)
-		{
-		}
-
-		int sampleRateMultiplier() const
-		{
-			switch( oversampling )
-			{
-				case Oversampling::None: return 1;
-				case Oversampling::X2: return 2;
-				case Oversampling::X4: return 4;
-				case Oversampling::X8: return 8;
-			}
-			return 1;
 		}
 
 		int libsrcInterpolation() const
@@ -289,8 +240,6 @@ class LMMS_EXPORT AudioEngine : public QObject
 	sample_rate_t baseSampleRate() const;
 	sample_rate_t outputSampleRate() const;
 	sample_rate_t inputSampleRate() const;
-	sample_rate_t processingSampleRate() const;
-
 
 	inline float masterGain() const
 	{

diff --git a/include/BandLimitedWave.h b/include/BandLimitedWave.h
@@ -107,7 +107,7 @@ class LMMS_EXPORT BandLimitedWave
 	 */
 	static inline float freqToLen( float f )
 	{
-		return freqToLen( f, Engine::audioEngine()->processingSampleRate() );
+		return freqToLen( f, Engine::audioEngine()->outputSampleRate() );
 	}
 
 	/*! \brief This method converts frequency to wavelength, but you can use any custom sample rate with it.

diff --git a/include/Effect.h b/include/Effect.h
@@ -111,7 +111,7 @@ class LMMS_EXPORT Effect : public Plugin
 
 	inline f_cnt_t timeout() const
 	{
-		const float samples = Engine::audioEngine()->processingSampleRate() * m_autoQuitModel.value() / 1000.0f;
+		const float samples = Engine::audioEngine()->outputSampleRate() * m_autoQuitModel.value() / 1000.0f;
 		return 1 + ( static_cast<int>( samples ) / Engine::audioEngine()->framesPerPeriod() );
 	}
 
@@ -192,7 +192,7 @@ class LMMS_EXPORT Effect : public Plugin
 							sample_rate_t _dst_sr )
 	{
 		resample( 0, _src_buf,
-				Engine::audioEngine()->processingSampleRate(),
+				Engine::audioEngine()->outputSampleRate(),
 					_dst_buf, _dst_sr,
 					Engine::audioEngine()->framesPerPeriod() );
 	}
@@ -202,9 +202,9 @@ class LMMS_EXPORT Effect : public Plugin
 							sample_rate_t _src_sr )
 	{
 		resample( 1, _src_buf, _src_sr, _dst_buf,
-				Engine::audioEngine()->processingSampleRate(),
+				Engine::audioEngine()->outputSampleRate(),
 			Engine::audioEngine()->framesPerPeriod() * _src_sr /
-				Engine::audioEngine()->processingSampleRate() );
+				Engine::audioEngine()->outputSampleRate() );
 	}
 	void reinitSRC();
 

diff --git a/include/Oscillator.h b/include/Oscillator.h
@@ -204,7 +204,7 @@ class LMMS_EXPORT Oscillator
 					control.f1 + 1 :
 					0;
 		control.band = waveTableBandFromFreq(
-			m_freq * m_detuning_div_samplerate * Engine::audioEngine()->processingSampleRate());
+			m_freq * m_detuning_div_samplerate * Engine::audioEngine()->outputSampleRate());
 		return control;
 	}
 

diff --git a/include/Sample.h b/include/Sample.h
@@ -77,8 +77,8 @@ class LMMS_EXPORT Sample
 	};
 
 	Sample() = default;
-	Sample(const QByteArray& base64, int sampleRate = Engine::audioEngine()->processingSampleRate());
-	Sample(const sampleFrame* data, size_t numFrames, int sampleRate = Engine::audioEngine()->processingSampleRate());
+	Sample(const QByteArray& base64, int sampleRate = Engine::audioEngine()->outputSampleRate());
+	Sample(const sampleFrame* data, size_t numFrames, int sampleRate = Engine::audioEngine()->outputSampleRate());
 	Sample(const Sample& other);
 	Sample(Sample&& other);
 	explicit Sample(const QString& audioFile);

diff --git a/include/SampleBuffer.h b/include/SampleBuffer.h
@@ -56,7 +56,7 @@ class LMMS_EXPORT SampleBuffer
 	SampleBuffer(const QString& base64, int sampleRate);
 	SampleBuffer(std::vector<sampleFrame> data, int sampleRate);
 	SampleBuffer(
-		const sampleFrame* data, size_t numFrames, int sampleRate = Engine::audioEngine()->processingSampleRate());
+		const sampleFrame* data, size_t numFrames, int sampleRate = Engine::audioEngine()->outputSampleRate());
 
 	friend void swap(SampleBuffer& first, SampleBuffer& second) noexcept;
 	auto toBase64() const -> QString;
@@ -91,7 +91,7 @@ class LMMS_EXPORT SampleBuffer
 private:
 	std::vector<sampleFrame> m_data;
 	QString m_audioFile;
-	sample_rate_t m_sampleRate = Engine::audioEngine()->processingSampleRate();
+	sample_rate_t m_sampleRate = Engine::audioEngine()->outputSampleRate();
 };
 
 } // namespace lmms

diff --git a/include/SampleLoader.h b/include/SampleLoader.h
@@ -39,7 +39,7 @@ class LMMS_EXPORT SampleLoader
 	static QString openWaveformFile(const QString& previousFile = "");
 	static std::shared_ptr<const SampleBuffer> createBufferFromFile(const QString& filePath);
 	static std::shared_ptr<const SampleBuffer> createBufferFromBase64(
-		const QString& base64, int sampleRate = Engine::audioEngine()->processingSampleRate());
+		const QString& base64, int sampleRate = Engine::audioEngine()->outputSampleRate());
 private:
 	static void displayError(const QString& message);
 };

diff --git a/plugins/AudioFileProcessor/AudioFileProcessor.cpp b/plugins/AudioFileProcessor/AudioFileProcessor.cpp
@@ -284,7 +284,7 @@ auto AudioFileProcessor::beatLen(NotePlayHandle* note) const -> int
 	// Otherwise, use the remaining sample duration
 	const auto baseFreq = instrumentTrack()->baseFreq();
 	const auto freqFactor = baseFreq / note->frequency()
-		* Engine::audioEngine()->processingSampleRate()
+		* Engine::audioEngine()->outputSampleRate()
 		/ Engine::audioEngine()->baseSampleRate();
 
 	const auto startFrame = m_nextPlayStartPoint >= m_sample.endFrame()

diff --git a/plugins/BassBooster/BassBooster.cpp b/plugins/BassBooster/BassBooster.cpp
@@ -118,7 +118,7 @@ bool BassBoosterEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames
 
 inline void BassBoosterEffect::changeFrequency()
 {
-	const sample_t fac = Engine::audioEngine()->processingSampleRate() / 44100.0f;
+	const sample_t fac = Engine::audioEngine()->outputSampleRate() / 44100.0f;
 
 	m_bbFX.leftFX().setFrequency( m_bbControls.m_freqModel.value() * fac );
 	m_bbFX.rightFX().setFrequency( m_bbControls.m_freqModel.value() * fac );

diff --git a/plugins/BitInvader/BitInvader.cpp b/plugins/BitInvader/BitInvader.cpp
@@ -265,7 +265,7 @@ void BitInvader::playNote( NotePlayHandle * _n,
 					const_cast<float*>( m_graph.samples() ),
 					_n,
 					m_interpolation.value(), factor,
-				Engine::audioEngine()->processingSampleRate() );
+				Engine::audioEngine()->outputSampleRate() );
 	}
 
 	const fpp_t frames = _n->framesLeftForCurrentPeriod();

diff --git a/plugins/Bitcrush/Bitcrush.cpp b/plugins/Bitcrush/Bitcrush.cpp
@@ -59,7 +59,7 @@ Plugin::Descriptor PLUGIN_EXPORT bitcrush_plugin_descriptor =
 BitcrushEffect::BitcrushEffect( Model * parent, const Descriptor::SubPluginFeatures::Key * key ) :
 	Effect( &bitcrush_plugin_descriptor, parent, key ),
 	m_controls( this ),
-	m_sampleRate( Engine::audioEngine()->processingSampleRate() ),
+	m_sampleRate( Engine::audioEngine()->outputSampleRate() ),
 	m_filter( m_sampleRate )
 {
 	m_buffer = new sampleFrame[Engine::audioEngine()->framesPerPeriod() * OS_RATE];
@@ -83,7 +83,7 @@ BitcrushEffect::~BitcrushEffect()
 
 void BitcrushEffect::sampleRateChanged()
 {
-	m_sampleRate = Engine::audioEngine()->processingSampleRate();
+	m_sampleRate = Engine::audioEngine()->outputSampleRate();
 	m_filter.setSampleRate( m_sampleRate );
 	m_filter.setLowpass( m_sampleRate * ( CUTOFF_RATIO * OS_RATIO ) );
 	m_needsUpdate = true;

diff --git a/plugins/CarlaBase/Carla.cpp b/plugins/CarlaBase/Carla.cpp
@@ -259,7 +259,7 @@ uint32_t CarlaInstrument::handleGetBufferSize() const
 
 double CarlaInstrument::handleGetSampleRate() const
 {
-    return Engine::audioEngine()->processingSampleRate();
+    return Engine::audioEngine()->outputSampleRate();
 }
 
 bool CarlaInstrument::handleIsOffline() const

diff --git a/plugins/Compressor/Compressor.cpp b/plugins/Compressor/Compressor.cpp
@@ -56,7 +56,7 @@ CompressorEffect::CompressorEffect(Model* parent, const Descriptor::SubPluginFea
 	Effect(&compressor_plugin_descriptor, parent, key),
 	m_compressorControls(this)
 {
-	m_sampleRate = Engine::audioEngine()->processingSampleRate();
+	m_sampleRate = Engine::audioEngine()->outputSampleRate();
 
 	m_yL[0] = m_yL[1] = COMP_NOISE_FLOOR;
 
@@ -560,7 +560,7 @@ inline void CompressorEffect::calcTiltFilter(sample_t inputSample, sample_t &out
 
 void CompressorEffect::changeSampleRate()
 {
-	m_sampleRate = Engine::audioEngine()->processingSampleRate();
+	m_sampleRate = Engine::audioEngine()->outputSampleRate();
 
 	m_coeffPrecalc = COMP_LOG / (m_sampleRate * 0.001f);
 

diff --git a/plugins/CrossoverEQ/CrossoverEQ.cpp b/plugins/CrossoverEQ/CrossoverEQ.cpp
@@ -55,7 +55,7 @@ Plugin::Descriptor PLUGIN_EXPORT crossovereq_plugin_descriptor =
 CrossoverEQEffect::CrossoverEQEffect( Model* parent, const Descriptor::SubPluginFeatures::Key* key ) :
 	Effect( &crossovereq_plugin_descriptor, parent, key ),
 	m_controls( this ),
-	m_sampleRate( Engine::audioEngine()->processingSampleRate() ),
+	m_sampleRate( Engine::audioEngine()->outputSampleRate() ),
 	m_lp1( m_sampleRate ),
 	m_lp2( m_sampleRate ),
 	m_lp3( m_sampleRate ),
@@ -78,7 +78,7 @@ CrossoverEQEffect::~CrossoverEQEffect()
 
 void CrossoverEQEffect::sampleRateChanged()
 {
-	m_sampleRate = Engine::audioEngine()->processingSampleRate();
+	m_sampleRate = Engine::audioEngine()->outputSampleRate();
 	m_lp1.setSampleRate( m_sampleRate );
 	m_lp2.setSampleRate( m_sampleRate );
 	m_lp3.setSampleRate( m_sampleRate );

diff --git a/plugins/Delay/DelayEffect.cpp b/plugins/Delay/DelayEffect.cpp
@@ -58,8 +58,8 @@ DelayEffect::DelayEffect( Model* parent, const Plugin::Descriptor::SubPluginFeat
 	m_delayControls( this )
 {
 	m_delay = 0;
-	m_delay = new StereoDelay( 20, Engine::audioEngine()->processingSampleRate() );
-	m_lfo = new Lfo( Engine::audioEngine()->processingSampleRate() );
+	m_delay = new StereoDelay( 20, Engine::audioEngine()->outputSampleRate() );
+	m_lfo = new Lfo( Engine::audioEngine()->outputSampleRate() );
 	m_outGain = 1.0;
 }
 
@@ -88,7 +88,7 @@ bool DelayEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 		return( false );
 	}
 	double outSum = 0.0;
-	const float sr = Engine::audioEngine()->processingSampleRate();
+	const float sr = Engine::audioEngine()->outputSampleRate();
 	const float d = dryLevel();
 	const float w = wetLevel();
 	auto dryS = std::array<sample_t, 2>{};
@@ -123,7 +123,7 @@ bool DelayEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 
 		m_delay->setFeedback( *feedbackPtr );
 		m_lfo->setFrequency( *lfoTimePtr );
-		m_currentLength = static_cast<int>(*lengthPtr * Engine::audioEngine()->processingSampleRate());
+		m_currentLength = static_cast<int>(*lengthPtr * Engine::audioEngine()->outputSampleRate());
 		m_delay->setLength( m_currentLength + ( *amplitudePtr * ( float )m_lfo->tick() ) );
 		m_delay->tick( buf[f] );
 
@@ -151,8 +151,8 @@ bool DelayEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
 
 void DelayEffect::changeSampleRate()
 {
-	m_lfo->setSampleRate( Engine::audioEngine()->processingSampleRate() );
-	m_delay->setSampleRate( Engine::audioEngine()->processingSampleRate() );
+	m_lfo->setSampleRate( Engine::audioEngine()->outputSampleRate() );
+	m_delay->setSampleRate( Engine::audioEngine()->outputSampleRate() );
 }
 
 

diff --git a/plugins/Dispersion/Dispersion.cpp b/plugins/Dispersion/Dispersion.cpp
@@ -52,7 +52,7 @@ Plugin::Descriptor PLUGIN_EXPORT dispersion_plugin_descriptor =
 DispersionEffect::DispersionEffect(Model* parent, const Descriptor::SubPluginFeatures::Key* key) :
 	Effect(&dispersion_plugin_descriptor, parent, key),
 	m_dispersionControls(this),
-	m_sampleRate(Engine::audioEngine()->processingSampleRate()),
+	m_sampleRate(Engine::audioEngine()->outputSampleRate()),
 	m_amountVal(0)
 {
 }

diff --git a/plugins/DualFilter/DualFilter.cpp b/plugins/DualFilter/DualFilter.cpp
@@ -57,8 +57,8 @@ DualFilterEffect::DualFilterEffect( Model* parent, const Descriptor::SubPluginFe
 	Effect( &dualfilter_plugin_descriptor, parent, key ),
 	m_dfControls( this )
 {
-	m_filter1 = new BasicFilters<2>( Engine::audioEngine()->processingSampleRate() );
-	m_filter2 = new BasicFilters<2>( Engine::audioEngine()->processingSampleRate() );
+	m_filter1 = new BasicFilters<2>( Engine::audioEngine()->outputSampleRate() );
+	m_filter2 = new BasicFilters<2>( Engine::audioEngine()->outputSampleRate() );
 
 	// ensure filters get updated
 	m_filter1changed = true;

diff --git a/plugins/DualFilter/DualFilterControls.cpp b/plugins/DualFilter/DualFilterControls.cpp
@@ -111,8 +111,8 @@ void DualFilterControls::updateFilters()
 
 	delete m_effect->m_filter1;
 	delete m_effect->m_filter2;
-	m_effect->m_filter1 = new BasicFilters<2>( Engine::audioEngine()->processingSampleRate() );
-	m_effect->m_filter2 = new BasicFilters<2>( Engine::audioEngine()->processingSampleRate() );
+	m_effect->m_filter1 = new BasicFilters<2>( Engine::audioEngine()->outputSampleRate() );
+	m_effect->m_filter2 = new BasicFilters<2>( Engine::audioEngine()->outputSampleRate() );
 
 	// flag filters as needing recalculation