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Remove global oversampling (#7228)

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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).
This commit is contained in:
saker
2024-05-05 04:37:43 -04:00
committed by GitHub
parent 9bdc8adf33
commit 9b6e33aa5c
75 changed files with 155 additions and 373 deletions
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12
include/AudioDevice.h
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@@ -89,8 +89,6 @@ public:
virtual void stopProcessing();
void applyQualitySettings();
protected:
// subclasses can re-implement this for being used in conjunction with
// processNextBuffer()
@@ -110,13 +108,6 @@ protected:
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 @@ private:
QMutex m_devMutex;
SRC_DATA m_srcData;
SRC_STATE * m_srcState;
surroundSampleFrame * m_buffer;
};

2
include/AudioDummy.h
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@@ -104,7 +104,7 @@ private:
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 );

55
include/AudioEngine.h
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@@ -108,13 +108,6 @@ public:
struct qualitySettings
{
enum class Mode
{
Draft,
HighQuality,
FinalMix
} ;
enum class Interpolation
{
Linear,
@@ -123,53 +116,11 @@ public:
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 @@ public:
sample_rate_t baseSampleRate() const;
sample_rate_t outputSampleRate() const;
sample_rate_t inputSampleRate() const;
sample_rate_t processingSampleRate() const;
inline float masterGain() const
{

2
include/BandLimitedWave.h
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@@ -107,7 +107,7 @@ public:
*/
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.

8
include/Effect.h
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@@ -111,7 +111,7 @@ public:
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 @@ protected:
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 @@ protected:
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();

2
include/Oscillator.h
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@@ -204,7 +204,7 @@ public:
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;
}

4
include/Sample.h
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@@ -77,8 +77,8 @@ public:
};
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);

4
include/SampleBuffer.h
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@@ -56,7 +56,7 @@ public:
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 @@ public:
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

2
include/SampleLoader.h
View File

@@ -39,7 +39,7 @@ public:
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);
};

2
plugins/AudioFileProcessor/AudioFileProcessor.cpp
View File

@@ -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()

2
plugins/BassBooster/BassBooster.cpp
View File

@@ -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 );

2
plugins/BitInvader/BitInvader.cpp
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@@ -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();

4
plugins/Bitcrush/Bitcrush.cpp
View File

@@ -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;

2
plugins/CarlaBase/Carla.cpp
View File

@@ -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

4
plugins/Compressor/Compressor.cpp
View File

@@ -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);

4
plugins/CrossoverEQ/CrossoverEQ.cpp
View File

@@ -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 );

12
plugins/Delay/DelayEffect.cpp
View File

@@ -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() );
}

2
plugins/Dispersion/Dispersion.cpp
View File

@@ -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)
{
}

4
plugins/DualFilter/DualFilter.cpp
View File

@@ -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;

4
plugins/DualFilter/DualFilterControls.cpp
View File

@@ -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

12
plugins/DynamicsProcessor/DynamicsProcessor.cpp
View File

@@ -64,8 +64,8 @@ DynProcEffect::DynProcEffect( Model * _parent,
m_dpControls( this )
{
m_currentPeak[0] = m_currentPeak[1] = DYN_NOISE_FLOOR;
m_rms[0] = new RmsHelper( 64 * Engine::audioEngine()->processingSampleRate() / 44100 );
m_rms[1] = new RmsHelper( 64 * Engine::audioEngine()->processingSampleRate() / 44100 );
m_rms[0] = new RmsHelper( 64 * Engine::audioEngine()->outputSampleRate() / 44100 );
m_rms[1] = new RmsHelper( 64 * Engine::audioEngine()->outputSampleRate() / 44100 );
calcAttack();
calcRelease();
}
@@ -82,12 +82,12 @@ DynProcEffect::~DynProcEffect()
inline void DynProcEffect::calcAttack()
{
m_attCoeff = std::exp((DNF_LOG / (m_dpControls.m_attackModel.value() * 0.001)) / Engine::audioEngine()->processingSampleRate());
m_attCoeff = std::exp((DNF_LOG / (m_dpControls.m_attackModel.value() * 0.001)) / Engine::audioEngine()->outputSampleRate());
}
inline void DynProcEffect::calcRelease()
{
m_relCoeff = std::exp((DNF_LOG / (m_dpControls.m_releaseModel.value() * 0.001)) / Engine::audioEngine()->processingSampleRate());
m_relCoeff = std::exp((DNF_LOG / (m_dpControls.m_releaseModel.value() * 0.001)) / Engine::audioEngine()->outputSampleRate());
}
@@ -122,8 +122,8 @@ bool DynProcEffect::processAudioBuffer( sampleFrame * _buf,
if( m_needsUpdate )
{
m_rms[0]->setSize( 64 * Engine::audioEngine()->processingSampleRate() / 44100 );
m_rms[1]->setSize( 64 * Engine::audioEngine()->processingSampleRate() / 44100 );
m_rms[0]->setSize( 64 * Engine::audioEngine()->outputSampleRate() / 44100 );
m_rms[1]->setSize( 64 * Engine::audioEngine()->outputSampleRate() / 44100 );
calcAttack();
calcRelease();
m_needsUpdate = false;

12
plugins/Eq/EqCurve.cpp
View File

@@ -201,7 +201,7 @@ bool EqHandle::mousePressed() const
float EqHandle::getPeakCurve( float x )
{
double freqZ = xPixelToFreq( EqHandle::x(), m_width );
const int SR = Engine::audioEngine()->processingSampleRate();
const int SR = Engine::audioEngine()->outputSampleRate();
double w0 = 2 * LD_PI * freqZ / SR ;
double c = cosf( w0 );
double s = sinf( w0 );
@@ -238,7 +238,7 @@ float EqHandle::getPeakCurve( float x )
float EqHandle::getHighShelfCurve( float x )
{
double freqZ = xPixelToFreq( EqHandle::x(), m_width );
const int SR = Engine::audioEngine()->processingSampleRate();
const int SR = Engine::audioEngine()->outputSampleRate();
double w0 = 2 * LD_PI * freqZ / SR;
double c = cosf( w0 );
double s = sinf( w0 );
@@ -274,7 +274,7 @@ float EqHandle::getHighShelfCurve( float x )
float EqHandle::getLowShelfCurve( float x )
{
double freqZ = xPixelToFreq( EqHandle::x(), m_width );
const int SR = Engine::audioEngine()->processingSampleRate();
const int SR = Engine::audioEngine()->outputSampleRate();
double w0 = 2 * LD_PI * freqZ / SR ;
double c = cosf( w0 );
double s = sinf( w0 );
@@ -310,7 +310,7 @@ float EqHandle::getLowShelfCurve( float x )
float EqHandle::getLowCutCurve( float x )
{
double freqZ = xPixelToFreq( EqHandle::x(), m_width );
const int SR = Engine::audioEngine()->processingSampleRate();
const int SR = Engine::audioEngine()->outputSampleRate();
double w0 = 2 * LD_PI * freqZ / SR ;
double c = cosf( w0 );
double s = sinf( w0 );
@@ -353,7 +353,7 @@ float EqHandle::getLowCutCurve( float x )
float EqHandle::getHighCutCurve( float x )
{
double freqZ = xPixelToFreq( EqHandle::x(), m_width );
const int SR = Engine::audioEngine()->processingSampleRate();
const int SR = Engine::audioEngine()->outputSampleRate();
double w0 = 2 * LD_PI * freqZ / SR ;
double c = cosf( w0 );
double s = sinf( w0 );
@@ -527,7 +527,7 @@ void EqHandle::setlp48()
double EqHandle::calculateGain(const double freq, const double a1, const double a2, const double b0, const double b1, const double b2 )
{
const int SR = Engine::audioEngine()->processingSampleRate();
const int SR = Engine::audioEngine()->outputSampleRate();
const double w = 2 * LD_PI * freq / SR ;
const double PHI = pow( sin( w / 2 ), 2 ) * 4;

2
plugins/Eq/EqEffect.cpp
View File

@@ -66,7 +66,7 @@ EqEffect::EqEffect( Model *parent, const Plugin::Descriptor::SubPluginFeatures::
bool EqEffect::processAudioBuffer( sampleFrame *buf, const fpp_t frames )
{
const int sampleRate = Engine::audioEngine()->processingSampleRate();
const int sampleRate = Engine::audioEngine()->outputSampleRate();
//wet/dry controls
const float dry = dryLevel();

2
plugins/Eq/EqSpectrumView.cpp
View File

@@ -102,7 +102,7 @@ void EqAnalyser::analyze( sampleFrame *buf, const fpp_t frames )
return;
}
m_sampleRate = Engine::audioEngine()->processingSampleRate();
m_sampleRate = Engine::audioEngine()->outputSampleRate();
const int LOWEST_FREQ = 0;
const int HIGHEST_FREQ = m_sampleRate / 2;

16
plugins/Flanger/FlangerEffect.cpp
View File

@@ -58,9 +58,9 @@ FlangerEffect::FlangerEffect( Model *parent, const Plugin::Descriptor::SubPlugin
Effect( &flanger_plugin_descriptor, parent, key ),
m_flangerControls( this )
{
m_lfo = new QuadratureLfo( Engine::audioEngine()->processingSampleRate() );
m_lDelay = new MonoDelay( 1, Engine::audioEngine()->processingSampleRate() );
m_rDelay = new MonoDelay( 1, Engine::audioEngine()->processingSampleRate() );
m_lfo = new QuadratureLfo( Engine::audioEngine()->outputSampleRate() );
m_lDelay = new MonoDelay( 1, Engine::audioEngine()->outputSampleRate() );
m_rDelay = new MonoDelay( 1, Engine::audioEngine()->outputSampleRate() );
m_noise = new Noise;
}
@@ -99,9 +99,9 @@ bool FlangerEffect::processAudioBuffer( sampleFrame *buf, const fpp_t frames )
double outSum = 0.0;
const float d = dryLevel();
const float w = wetLevel();
const float length = m_flangerControls.m_delayTimeModel.value() * Engine::audioEngine()->processingSampleRate();
const float length = m_flangerControls.m_delayTimeModel.value() * Engine::audioEngine()->outputSampleRate();
const float noise = m_flangerControls.m_whiteNoiseAmountModel.value();
float amplitude = m_flangerControls.m_lfoAmountModel.value() * Engine::audioEngine()->processingSampleRate();
float amplitude = m_flangerControls.m_lfoAmountModel.value() * Engine::audioEngine()->outputSampleRate();
bool invertFeedback = m_flangerControls.m_invertFeedbackModel.value();
m_lfo->setFrequency( 1.0/m_flangerControls.m_lfoFrequencyModel.value() );
m_lfo->setOffset( m_flangerControls.m_lfoPhaseModel.value() / 180 * D_PI );
@@ -143,9 +143,9 @@ bool FlangerEffect::processAudioBuffer( sampleFrame *buf, const fpp_t frames )
void FlangerEffect::changeSampleRate()
{
m_lfo->setSampleRate( Engine::audioEngine()->processingSampleRate() );
m_lDelay->setSampleRate( Engine::audioEngine()->processingSampleRate() );
m_rDelay->setSampleRate( Engine::audioEngine()->processingSampleRate() );
m_lfo->setSampleRate( Engine::audioEngine()->outputSampleRate() );
m_lDelay->setSampleRate( Engine::audioEngine()->outputSampleRate() );
m_rDelay->setSampleRate( Engine::audioEngine()->outputSampleRate() );
}

2
plugins/FreeBoy/FreeBoy.cpp
View File

@@ -231,7 +231,7 @@ float FreeBoyInstrument::desiredReleaseTimeMs() const
void FreeBoyInstrument::playNote(NotePlayHandle* nph, sampleFrame* workingBuffer)
{
const f_cnt_t tfp = nph->totalFramesPlayed();
const int samplerate = Engine::audioEngine()->processingSampleRate();
const int samplerate = Engine::audioEngine()->outputSampleRate();
const fpp_t frames = nph->framesLeftForCurrentPeriod();
const f_cnt_t offset = nph->noteOffset();

4
plugins/GigPlayer/GigPlayer.cpp
View File

@@ -323,7 +323,7 @@ void GigInstrument::playNote( NotePlayHandle * _n, sampleFrame * )
void GigInstrument::play( sampleFrame * _working_buffer )
{
const fpp_t frames = Engine::audioEngine()->framesPerPeriod();
const int rate = Engine::audioEngine()->processingSampleRate();
const int rate = Engine::audioEngine()->outputSampleRate();
// Initialize to zeros
std::memset( &_working_buffer[0][0], 0, DEFAULT_CHANNELS * frames * sizeof( float ) );
@@ -746,7 +746,7 @@ void GigInstrument::addSamples( GigNote & gignote, bool wantReleaseSample )
if( gignote.midiNote >= keyLow && gignote.midiNote <= keyHigh )
{
float attenuation = pDimRegion->GetVelocityAttenuation( gignote.velocity );
float length = (float) pSample->SamplesTotal / Engine::audioEngine()->processingSampleRate();
float length = (float) pSample->SamplesTotal / Engine::audioEngine()->outputSampleRate();
// TODO: sample panning? crossfade different layers?

4
plugins/Kicker/Kicker.cpp
View File

@@ -160,7 +160,7 @@ void KickerInstrument::playNote( NotePlayHandle * _n,
{
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
const float decfr = m_decayModel.value() * Engine::audioEngine()->processingSampleRate() / 1000.0f;
const float decfr = m_decayModel.value() * Engine::audioEngine()->outputSampleRate() / 1000.0f;
const f_cnt_t tfp = _n->totalFramesPlayed();
if (!_n->m_pluginData)
@@ -184,7 +184,7 @@ void KickerInstrument::playNote( NotePlayHandle * _n,
}
auto so = static_cast<SweepOsc*>(_n->m_pluginData);
so->update( _working_buffer + offset, frames, Engine::audioEngine()->processingSampleRate() );
so->update( _working_buffer + offset, frames, Engine::audioEngine()->outputSampleRate() );
if( _n->isReleased() )
{

4
plugins/LOMM/LOMM.cpp
View File

@@ -50,7 +50,7 @@ extern "C"
LOMMEffect::LOMMEffect(Model* parent, const Descriptor::SubPluginFeatures::Key* key) :
Effect(&lomm_plugin_descriptor, parent, key),
m_lommControls(this),
m_sampleRate(Engine::audioEngine()->processingSampleRate()),
m_sampleRate(Engine::audioEngine()->outputSampleRate()),
m_lp1(m_sampleRate),
m_lp2(m_sampleRate),
m_hp1(m_sampleRate),
@@ -76,7 +76,7 @@ LOMMEffect::LOMMEffect(Model* parent, const Descriptor::SubPluginFeatures::Key*
void LOMMEffect::changeSampleRate()
{
m_sampleRate = Engine::audioEngine()->processingSampleRate();
m_sampleRate = Engine::audioEngine()->outputSampleRate();
m_lp1.setSampleRate(m_sampleRate);
m_lp2.setSampleRate(m_sampleRate);
m_hp1.setSampleRate(m_sampleRate);

4
plugins/LadspaBrowser/LadspaPortDialog.cpp
View File

@@ -90,11 +90,11 @@ LadspaPortDialog::LadspaPortDialog( const ladspa_key_t & _key )
{
if( min != NOHINT )
{
min *= Engine::audioEngine()->processingSampleRate();
min *= Engine::audioEngine()->outputSampleRate();
}
if( max != NOHINT )
{
max *= Engine::audioEngine()->processingSampleRate();
max *= Engine::audioEngine()->outputSampleRate();
}
}

6
plugins/LadspaEffect/LadspaEffect.cpp
View File

@@ -143,13 +143,13 @@ bool LadspaEffect::processAudioBuffer( sampleFrame * _buf,
sampleFrame * o_buf = nullptr;
QVarLengthArray<sample_t> sBuf(_frames * DEFAULT_CHANNELS);
if( m_maxSampleRate < Engine::audioEngine()->processingSampleRate() )
if( m_maxSampleRate < Engine::audioEngine()->outputSampleRate() )
{
o_buf = _buf;
_buf = reinterpret_cast<sampleFrame*>(sBuf.data());
sampleDown( o_buf, _buf, m_maxSampleRate );
frames = _frames * m_maxSampleRate /
Engine::audioEngine()->processingSampleRate();
Engine::audioEngine()->outputSampleRate();
}
// Copy the LMMS audio buffer to the LADSPA input buffer and initialize
@@ -587,7 +587,7 @@ sample_rate_t LadspaEffect::maxSamplerate( const QString & _name )
{
return( __buggy_plugins[_name] );
}
return( Engine::audioEngine()->processingSampleRate() );
return( Engine::audioEngine()->outputSampleRate() );
}

18
plugins/Lb302/Lb302.cpp
View File

@@ -72,7 +72,7 @@ namespace lmms
{
//#define engine::audioEngine()->processingSampleRate() 44100.0f
//#define engine::audioEngine()->outputSampleRate() 44100.0f
const float sampleRateCutoff = 44100.0f;
extern "C"
@@ -111,8 +111,8 @@ void Lb302Filter::recalc()
{
vcf_e1 = exp(6.109 + 1.5876*(fs->envmod) + 2.1553*(fs->cutoff) - 1.2*(1.0-(fs->reso)));
vcf_e0 = exp(5.613 - 0.8*(fs->envmod) + 2.1553*(fs->cutoff) - 0.7696*(1.0-(fs->reso)));
vcf_e0*=M_PI/Engine::audioEngine()->processingSampleRate();
vcf_e1*=M_PI/Engine::audioEngine()->processingSampleRate();
vcf_e0*=M_PI/Engine::audioEngine()->outputSampleRate();
vcf_e1*=M_PI/Engine::audioEngine()->outputSampleRate();
vcf_e1 -= vcf_e0;
vcf_rescoeff = exp(-1.20 + 3.455*(fs->reso));
@@ -233,7 +233,7 @@ void Lb302Filter3Pole::envRecalc()
#ifdef LB_24_IGNORE_ENVELOPE
// kfcn = fs->cutoff;
kfcn = 2.0 * kfco / Engine::audioEngine()->processingSampleRate();
kfcn = 2.0 * kfco / Engine::audioEngine()->outputSampleRate();
#else
kfcn = w;
#endif
@@ -414,7 +414,7 @@ void Lb302Synth::filterChanged()
float d = 0.2 + (2.3*vcf_dec_knob.value());
d *= Engine::audioEngine()->processingSampleRate(); // d *= smpl rate
d *= Engine::audioEngine()->outputSampleRate(); // d *= smpl rate
fs.envdecay = pow(0.1, 1.0/d * ENVINC); // decay is 0.1 to the 1/d * ENVINC
// vcf_envdecay is now adjusted for both
// sampling rate and ENVINC
@@ -448,7 +448,7 @@ void Lb302Synth::recalcFilter()
// THIS IS OLD 3pole/24dB code, I may reintegrate it. Don't need it
// right now. Should be toggled by LB_24_RES_TRICK at the moment.
/*kfcn = 2.0 * (((vcf_cutoff*3000))) / engine::audioEngine()->processingSampleRate();
/*kfcn = 2.0 * (((vcf_cutoff*3000))) / engine::audioEngine()->outputSampleRate();
kp = ((-2.7528*kfcn + 3.0429)*kfcn + 1.718)*kfcn - 0.9984;
kp1 = kp+1.0;
kp1h = 0.5*kp1;
@@ -459,12 +459,12 @@ void Lb302Synth::recalcFilter()
}
inline float GET_INC(float freq) {
return freq/Engine::audioEngine()->processingSampleRate(); // TODO: Use actual sampling rate.
return freq/Engine::audioEngine()->outputSampleRate(); // TODO: Use actual sampling rate.
}
int Lb302Synth::process(sampleFrame *outbuf, const int size)
{
const float sampleRatio = 44100.f / Engine::audioEngine()->processingSampleRate();
const float sampleRatio = 44100.f / Engine::audioEngine()->outputSampleRate();
// Hold on to the current VCF, and use it throughout this period
Lb302Filter *filter = vcf.loadAcquire();
@@ -648,7 +648,7 @@ int Lb302Synth::process(sampleFrame *outbuf, const int size)
// Handle Envelope
if(vca_mode==VcaMode::Attack) {
vca_a+=(vca_a0-vca_a)*vca_attack;
if(sample_cnt>=0.5*Engine::audioEngine()->processingSampleRate())
if(sample_cnt>=0.5*Engine::audioEngine()->outputSampleRate())
vca_mode = VcaMode::Idle;
}
else if(vca_mode == VcaMode::Decay) {

2
plugins/Monstro/Monstro.cpp
View File

@@ -1447,7 +1447,7 @@ void MonstroInstrument::updateLFOAtts()
void MonstroInstrument::updateSamplerate()
{
m_samplerate = Engine::audioEngine()->processingSampleRate();
m_samplerate = Engine::audioEngine()->outputSampleRate();
m_integrator = 0.5f - ( 0.5f - INTEGRATOR ) * 44100.0f / m_samplerate;
m_fmCorrection = 44100.f / m_samplerate * FM_AMOUNT;

2
plugins/MultitapEcho/MultitapEcho.cpp
View File

@@ -55,7 +55,7 @@ MultitapEchoEffect::MultitapEchoEffect( Model* parent, const Descriptor::SubPlug
m_stages( 1 ),
m_controls( this ),
m_buffer( 16100.0f ),
m_sampleRate( Engine::audioEngine()->processingSampleRate() ),
m_sampleRate( Engine::audioEngine()->outputSampleRate() ),
m_sampleRatio( 1.0f / m_sampleRate )
{
m_work = new sampleFrame[Engine::audioEngine()->framesPerPeriod()];

2
plugins/MultitapEcho/MultitapEchoControls.cpp
View File

@@ -172,7 +172,7 @@ void MultitapEchoControls::lengthChanged()
void MultitapEchoControls::sampleRateChanged()
{
m_effect->m_sampleRate = Engine::audioEngine()->processingSampleRate();
m_effect->m_sampleRate = Engine::audioEngine()->outputSampleRate();
m_effect->m_sampleRatio = 1.0f / m_effect->m_sampleRate;
m_effect->updateFilters( 0, 19 );
}

2
plugins/Nes/Nes.cpp
View File

@@ -552,7 +552,7 @@ void NesInstrument::playNote( NotePlayHandle * n, sampleFrame * workingBuffer )
if (!n->m_pluginData)
{
auto nes = new NesObject(this, Engine::audioEngine()->processingSampleRate(), n);
auto nes = new NesObject(this, Engine::audioEngine()->outputSampleRate(), n);
n->m_pluginData = nes;
}

4
plugins/OpulenZ/OpulenZ.cpp
View File

@@ -140,7 +140,7 @@ OpulenzInstrument::OpulenzInstrument( InstrumentTrack * _instrument_track ) :
// Create an emulator - samplerate, 16 bit, mono
emulatorMutex.lock();
theEmulator = new CTemuopl(Engine::audioEngine()->processingSampleRate(), true, false);
theEmulator = new CTemuopl(Engine::audioEngine()->outputSampleRate(), true, false);
theEmulator->init();
// Enable waveform selection
theEmulator->write(0x01,0x20);
@@ -231,7 +231,7 @@ OpulenzInstrument::~OpulenzInstrument() {
void OpulenzInstrument::reloadEmulator() {
delete theEmulator;
emulatorMutex.lock();
theEmulator = new CTemuopl(Engine::audioEngine()->processingSampleRate(), true, false);
theEmulator = new CTemuopl(Engine::audioEngine()->outputSampleRate(), true, false);
theEmulator->init();
theEmulator->write(0x01,0x20);
emulatorMutex.unlock();

4
plugins/Organic/Organic.cpp
View File

@@ -605,10 +605,10 @@ void OscillatorObject::updateDetuning()
{
m_detuningLeft = powf( 2.0f, OrganicInstrument::s_harmonics[ static_cast<int>( m_harmModel.value() ) ]
+ (float)m_detuneModel.value() * CENT ) /
Engine::audioEngine()->processingSampleRate();
Engine::audioEngine()->outputSampleRate();
m_detuningRight = powf( 2.0f, OrganicInstrument::s_harmonics[ static_cast<int>( m_harmModel.value() ) ]
- (float)m_detuneModel.value() * CENT ) /
Engine::audioEngine()->processingSampleRate();
Engine::audioEngine()->outputSampleRate();
}

12
plugins/ReverbSC/ReverbSC.cpp
View File

@@ -55,7 +55,7 @@ ReverbSCEffect::ReverbSCEffect( Model* parent, const Descriptor::SubPluginFeatur
m_reverbSCControls( this )
{
sp_create(&sp);
sp->sr = Engine::audioEngine()->processingSampleRate();
sp->sr = Engine::audioEngine()->outputSampleRate();
sp_revsc_create(&revsc);
sp_revsc_init(sp, revsc);
@@ -63,8 +63,8 @@ ReverbSCEffect::ReverbSCEffect( Model* parent, const Descriptor::SubPluginFeatur
sp_dcblock_create(&dcblk[0]);
sp_dcblock_create(&dcblk[1]);
sp_dcblock_init(sp, dcblk[0], Engine::audioEngine()->currentQualitySettings().sampleRateMultiplier() );
sp_dcblock_init(sp, dcblk[1], Engine::audioEngine()->currentQualitySettings().sampleRateMultiplier() );
sp_dcblock_init(sp, dcblk[0], 1);
sp_dcblock_init(sp, dcblk[1], 1);
}
ReverbSCEffect::~ReverbSCEffect()
@@ -132,7 +132,7 @@ bool ReverbSCEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
void ReverbSCEffect::changeSampleRate()
{
// Change sr variable in Soundpipe. does not need to be destroyed
sp->sr = Engine::audioEngine()->processingSampleRate();
sp->sr = Engine::audioEngine()->outputSampleRate();
mutex.lock();
sp_revsc_destroy(&revsc);
@@ -145,8 +145,8 @@ void ReverbSCEffect::changeSampleRate()
sp_dcblock_create(&dcblk[0]);
sp_dcblock_create(&dcblk[1]);
sp_dcblock_init(sp, dcblk[0], Engine::audioEngine()->currentQualitySettings().sampleRateMultiplier() );
sp_dcblock_init(sp, dcblk[1], Engine::audioEngine()->currentQualitySettings().sampleRateMultiplier() );
sp_dcblock_init(sp, dcblk[0], 1);
sp_dcblock_init(sp, dcblk[1], 1);
mutex.unlock();
}

8
plugins/Sf2Player/Sf2Player.cpp
View File

@@ -574,7 +574,7 @@ void Sf2Instrument::reloadSynth()
double tempRate;
// Set & get, returns the true sample rate
fluid_settings_setnum( m_settings, (char *) "synth.sample-rate", Engine::audioEngine()->processingSampleRate() );
fluid_settings_setnum( m_settings, (char *) "synth.sample-rate", Engine::audioEngine()->outputSampleRate() );
fluid_settings_getnum( m_settings, (char *) "synth.sample-rate", &tempRate );
m_internalSampleRate = static_cast<int>( tempRate );
@@ -616,7 +616,7 @@ void Sf2Instrument::reloadSynth()
fluid_synth_set_interp_method( m_synth, -1, FLUID_INTERP_DEFAULT );
}
m_synthMutex.unlock();
if( m_internalSampleRate < Engine::audioEngine()->processingSampleRate() )
if( m_internalSampleRate < Engine::audioEngine()->outputSampleRate() )
{
m_synthMutex.lock();
if( m_srcState != nullptr )
@@ -872,10 +872,10 @@ void Sf2Instrument::renderFrames( f_cnt_t frames, sampleFrame * buf )
{
m_synthMutex.lock();
fluid_synth_get_gain(m_synth); // This flushes voice updates as a side effect
if( m_internalSampleRate < Engine::audioEngine()->processingSampleRate() &&
if( m_internalSampleRate < Engine::audioEngine()->outputSampleRate() &&
m_srcState != nullptr )
{
const fpp_t f = frames * m_internalSampleRate / Engine::audioEngine()->processingSampleRate();
const fpp_t f = frames * m_internalSampleRate / Engine::audioEngine()->outputSampleRate();
#ifdef __GNUC__
sampleFrame tmp[f];
#else

2
plugins/Sfxr/Sfxr.cpp
View File

@@ -444,7 +444,7 @@ QString SfxrInstrument::nodeName() const
void SfxrInstrument::playNote( NotePlayHandle * _n, sampleFrame * _working_buffer )
{
float currentSampleRate = Engine::audioEngine()->processingSampleRate();
float currentSampleRate = Engine::audioEngine()->outputSampleRate();
fpp_t frameNum = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();

2
plugins/Sid/SidInstrument.cpp
View File

@@ -289,7 +289,7 @@ void SidInstrument::playNote( NotePlayHandle * _n,
sampleFrame * _working_buffer )
{
const int clockrate = C64_PAL_CYCLES_PER_SEC;
const int samplerate = Engine::audioEngine()->processingSampleRate();
const int samplerate = Engine::audioEngine()->outputSampleRate();
if (!_n->m_pluginData)
{

4
plugins/SlicerT/SlicerT.cpp
View File

@@ -88,7 +88,7 @@ void SlicerT::playNote(NotePlayHandle* handle, sampleFrame* workingBuffer)
float speedRatio = static_cast<float>(m_originalBPM.value()) / bpm;
if (!m_enableSync.value()) { speedRatio = 1; }
speedRatio *= pitchRatio;
speedRatio *= Engine::audioEngine()->processingSampleRate() / static_cast<float>(m_originalSample.sampleRate());
speedRatio *= Engine::audioEngine()->outputSampleRate() / static_cast<float>(m_originalSample.sampleRate());
float sliceStart, sliceEnd;
if (noteIndex == 0) // full sample at base note
@@ -132,7 +132,7 @@ void SlicerT::playNote(NotePlayHandle* handle, sampleFrame* workingBuffer)
playbackState->setNoteDone(nextNoteDone);
// exponential fade out, applyRelease() not used since it extends the note length
int fadeOutFrames = m_fadeOutFrames.value() / 1000.0f * Engine::audioEngine()->processingSampleRate();
int fadeOutFrames = m_fadeOutFrames.value() / 1000.0f * Engine::audioEngine()->outputSampleRate();
int noteFramesLeft = noteLeft * m_originalSample.sampleSize() * speedRatio;
for (int i = 0; i < frames; i++)
{

4
plugins/SpectrumAnalyzer/SaProcessor.cpp
View File

@@ -53,7 +53,7 @@ SaProcessor::SaProcessor(const SaControls *controls) :
m_terminate(false),
m_inBlockSize(FFT_BLOCK_SIZES[0]),
m_fftBlockSize(FFT_BLOCK_SIZES[0]),
m_sampleRate(Engine::audioEngine()->processingSampleRate()),
m_sampleRate(Engine::audioEngine()->outputSampleRate()),
m_framesFilledUp(0),
m_spectrumActive(false),
m_waterfallActive(false),
@@ -166,7 +166,7 @@ void SaProcessor::analyze(LocklessRingBuffer<sampleFrame> &ring_buffer)
#endif
// update sample rate
m_sampleRate = Engine::audioEngine()->processingSampleRate();
m_sampleRate = Engine::audioEngine()->outputSampleRate();
// apply FFT window
for (unsigned int i = 0; i < m_inBlockSize; i++)

6
plugins/Stk/Mallets/Mallets.cpp
View File

@@ -342,7 +342,7 @@ void MalletsInstrument::playNote( NotePlayHandle * _n,
m_vibratoFreqModel.value(),
p,
(uint8_t) m_spreadModel.value(),
Engine::audioEngine()->processingSampleRate() );
Engine::audioEngine()->outputSampleRate() );
}
else if( p == 9 )
{
@@ -355,7 +355,7 @@ void MalletsInstrument::playNote( NotePlayHandle * _n,
m_lfoSpeedModel.value(),
m_adsrModel.value(),
(uint8_t) m_spreadModel.value(),
Engine::audioEngine()->processingSampleRate() );
Engine::audioEngine()->outputSampleRate() );
}
else
{
@@ -368,7 +368,7 @@ void MalletsInstrument::playNote( NotePlayHandle * _n,
m_strikeModel.value() * 128.0,
speed,
(uint8_t) m_spreadModel.value(),
Engine::audioEngine()->processingSampleRate() );
Engine::audioEngine()->outputSampleRate() );
}
m.unlock();
static_cast<MalletsSynth *>(_n->m_pluginData)->setPresetIndex(p);

4
plugins/TripleOscillator/TripleOscillator.cpp
View File

@@ -177,7 +177,7 @@ void OscillatorObject::updateDetuningLeft()
{
m_detuningLeft = powf( 2.0f, ( (float)m_coarseModel.value() * 100.0f
+ (float)m_fineLeftModel.value() ) / 1200.0f )
/ Engine::audioEngine()->processingSampleRate();
/ Engine::audioEngine()->outputSampleRate();
}
@@ -187,7 +187,7 @@ void OscillatorObject::updateDetuningRight()
{
m_detuningRight = powf( 2.0f, ( (float)m_coarseModel.value() * 100.0f
+ (float)m_fineRightModel.value() ) / 1200.0f )
/ Engine::audioEngine()->processingSampleRate();
/ Engine::audioEngine()->outputSampleRate();
}

2
plugins/Vibed/Vibed.cpp
View File

@@ -206,7 +206,7 @@ void Vibed::playNote(NotePlayHandle* n, sampleFrame* workingBuffer)
if (!n->m_pluginData)
{
const auto newContainer = new StringContainer{n->frequency(),
Engine::audioEngine()->processingSampleRate(), s_sampleLength};
Engine::audioEngine()->outputSampleRate(), s_sampleLength};
n->m_pluginData = newContainer;

2
plugins/VstBase/VstPlugin.cpp
View File

@@ -338,7 +338,7 @@ void VstPlugin::updateSampleRate()
{
lock();
sendMessage( message( IdSampleRateInformation ).
addInt( Engine::audioEngine()->processingSampleRate() ) );
addInt( Engine::audioEngine()->outputSampleRate() ) );
waitForMessage( IdInformationUpdated, true );
unlock();
}

2
plugins/Watsyn/Watsyn.cpp
View File

@@ -332,7 +332,7 @@ void WatsynInstrument::playNote( NotePlayHandle * _n,
if (!_n->m_pluginData)
{
auto w = new WatsynObject(&A1_wave[0], &A2_wave[0], &B1_wave[0], &B2_wave[0], m_amod.value(), m_bmod.value(),
Engine::audioEngine()->processingSampleRate(), _n, Engine::audioEngine()->framesPerPeriod(), this);
Engine::audioEngine()->outputSampleRate(), _n, Engine::audioEngine()->framesPerPeriod(), this);
_n->m_pluginData = w;
}

8
plugins/Xpressive/Xpressive.cpp
View File

@@ -204,14 +204,14 @@ void Xpressive::playNote(NotePlayHandle* nph, sampleFrame* working_buffer) {
if (!nph->m_pluginData) {
auto exprO1 = new ExprFront(m_outputExpression[0].constData(),
Engine::audioEngine()->processingSampleRate()); // give the "last" function a whole second
auto exprO2 = new ExprFront(m_outputExpression[1].constData(), Engine::audioEngine()->processingSampleRate());
Engine::audioEngine()->outputSampleRate()); // give the "last" function a whole second
auto exprO2 = new ExprFront(m_outputExpression[1].constData(), Engine::audioEngine()->outputSampleRate());
auto init_expression_step1 = [this, nph](ExprFront* e) { //lambda function to init exprO1 and exprO2
//add the constants and the variables to the expression.
e->add_constant("key", nph->key());//the key that was pressed.
e->add_constant("bnote", nph->instrumentTrack()->baseNote()); // the base note
e->add_constant("srate", Engine::audioEngine()->processingSampleRate());// sample rate of the audio engine
e->add_constant("srate", Engine::audioEngine()->outputSampleRate());// sample rate of the audio engine
e->add_constant("v", nph->getVolume() / 255.0); //volume of the note.
e->add_constant("tempo", Engine::getSong()->getTempo());//tempo of the song.
e->add_variable("A1", m_A1);//A1,A2,A3: general purpose input controls.
@@ -225,7 +225,7 @@ void Xpressive::playNote(NotePlayHandle* nph, sampleFrame* working_buffer) {
m_W2.setInterpolate(m_interpolateW2.value());
m_W3.setInterpolate(m_interpolateW3.value());
nph->m_pluginData = new ExprSynth(&m_W1, &m_W2, &m_W3, exprO1, exprO2, nph,
Engine::audioEngine()->processingSampleRate(), &m_panning1, &m_panning2, m_relTransition.value());
Engine::audioEngine()->outputSampleRate(), &m_panning1, &m_panning2, m_relTransition.value());
}
auto ps = static_cast<ExprSynth*>(nph->m_pluginData);

4
plugins/ZynAddSubFx/ZynAddSubFx.cpp
View File

@@ -451,7 +451,7 @@ void ZynAddSubFxInstrument::initPlugin()
QDir( ConfigManager::inst()->factoryPresetsDir() +
"/ZynAddSubFX" ).absolutePath() ) ) );
m_remotePlugin->updateSampleRate( Engine::audioEngine()->processingSampleRate() );
m_remotePlugin->updateSampleRate( Engine::audioEngine()->outputSampleRate() );
// temporary workaround until the VST synchronization feature gets stripped out of the RemotePluginClient class
// causing not to send buffer size information requests
@@ -463,7 +463,7 @@ void ZynAddSubFxInstrument::initPlugin()
else
{
m_plugin = new LocalZynAddSubFx;
m_plugin->setSampleRate( Engine::audioEngine()->processingSampleRate() );
m_plugin->setSampleRate( Engine::audioEngine()->outputSampleRate() );
m_plugin->setBufferSize( Engine::audioEngine()->framesPerPeriod() );
}

16
src/core/AudioEngine.cpp
View File

@@ -83,7 +83,7 @@ AudioEngine::AudioEngine( bool renderOnly ) :
m_workers(),
m_numWorkers( QThread::idealThreadCount()-1 ),
m_newPlayHandles( PlayHandle::MaxNumber ),
m_qualitySettings( qualitySettings::Mode::Draft ),
m_qualitySettings(qualitySettings::Interpolation::Linear),
m_masterGain( 1.0f ),
m_audioDev( nullptr ),
m_oldAudioDev( nullptr ),
@@ -277,17 +277,6 @@ sample_rate_t AudioEngine::inputSampleRate() const
baseSampleRate();
}
sample_rate_t AudioEngine::processingSampleRate() const
{
return outputSampleRate() * m_qualitySettings.sampleRateMultiplier();
}
bool AudioEngine::criticalXRuns() const
{
return cpuLoad() >= 99 && Engine::getSong()->isExporting() == false;
@@ -459,7 +448,7 @@ const surroundSampleFrame *AudioEngine::renderNextBuffer()
renderStageMix(); // STAGE 3: do master mix in mixer
s_renderingThread = false;
m_profiler.finishPeriod(processingSampleRate(), m_framesPerPeriod);
m_profiler.finishPeriod(outputSampleRate(), m_framesPerPeriod);
return m_outputBufferRead;
}
@@ -597,7 +586,6 @@ void AudioEngine::changeQuality(const struct qualitySettings & qs)
stopProcessing();
m_qualitySettings = qs;
m_audioDev->applyQualitySettings();
emit sampleRateChanged();
emit qualitySettingsChanged();

2
src/core/Controller.cpp
View File

@@ -149,7 +149,7 @@ unsigned int Controller::runningFrames()
// Get position in seconds
float Controller::runningTime()
{
return runningFrames() / Engine::audioEngine()->processingSampleRate();
return runningFrames() / Engine::audioEngine()->outputSampleRate();
}

2
src/core/Engine.cpp
View File

@@ -146,7 +146,7 @@ float Engine::framesPerTick(sample_rate_t sampleRate)
void Engine::updateFramesPerTick()
{
s_framesPerTick = s_audioEngine->processingSampleRate() * 60.0f * 4 / DefaultTicksPerBar / s_song->getTempo();
s_framesPerTick = s_audioEngine->outputSampleRate() * 60.0f * 4 / DefaultTicksPerBar / s_song->getTempo();
}

4
src/core/EnvelopeAndLfoParameters.cpp
View File

@@ -410,7 +410,7 @@ void EnvelopeAndLfoParameters::updateSampleVars()
QMutexLocker m(&m_paramMutex);
const float frames_per_env_seg = SECS_PER_ENV_SEGMENT *
Engine::audioEngine()->processingSampleRate();
Engine::audioEngine()->outputSampleRate();
// TODO: Remove the expKnobVals, time should be linear
const auto predelay_frames = static_cast<f_cnt_t>(frames_per_env_seg * expKnobVal(m_predelayModel.value()));
@@ -509,7 +509,7 @@ void EnvelopeAndLfoParameters::updateSampleVars()
const float frames_per_lfo_oscillation = SECS_PER_LFO_OSCILLATION *
Engine::audioEngine()->processingSampleRate();
Engine::audioEngine()->outputSampleRate();
m_lfoPredelayFrames = static_cast<f_cnt_t>( frames_per_lfo_oscillation *
expKnobVal( m_lfoPredelayModel.value() ) );
m_lfoAttackFrames = static_cast<f_cnt_t>( frames_per_lfo_oscillation *

2
src/core/Instrument.cpp
View File

@@ -205,7 +205,7 @@ float Instrument::computeReleaseTimeMsByFrameCount(f_cnt_t frames) const
sample_rate_t Instrument::getSampleRate() const
{
return Engine::audioEngine()->processingSampleRate();
return Engine::audioEngine()->outputSampleRate();
}

2
src/core/InstrumentFunctions.cpp
View File

@@ -369,7 +369,7 @@ void InstrumentFunctionArpeggio::processNote( NotePlayHandle * _n )
const int total_range = range * cnphv.size();
// number of frames that every note should be played
const auto arp_frames = (f_cnt_t)(m_arpTimeModel.value() / 1000.0f * Engine::audioEngine()->processingSampleRate());
const auto arp_frames = (f_cnt_t)(m_arpTimeModel.value() / 1000.0f * Engine::audioEngine()->outputSampleRate());
const auto gated_frames = (f_cnt_t)(m_arpGateModel.value() * arp_frames / 100.0f);
// used for calculating remaining frames for arp-note, we have to add

2
src/core/InstrumentSoundShaping.cpp
View File

@@ -158,7 +158,7 @@ void InstrumentSoundShaping::processAudioBuffer( sampleFrame* buffer,
if( n->m_filter == nullptr )
{
n->m_filter = std::make_unique<BasicFilters<>>( Engine::audioEngine()->processingSampleRate() );
n->m_filter = std::make_unique<BasicFilters<>>( Engine::audioEngine()->outputSampleRate() );
}
n->m_filter->setFilterType( static_cast<BasicFilters<>::FilterType>(m_filterModel.value()) );

2
src/core/LfoController.cpp
View File

@@ -155,7 +155,7 @@ void LfoController::updatePhase()
void LfoController::updateDuration()
{
float newDurationF = Engine::audioEngine()->processingSampleRate() * m_speedModel.value();
float newDurationF = Engine::audioEngine()->outputSampleRate() * m_speedModel.value();
switch(m_multiplierModel.value() )
{

6
src/core/Oscillator.cpp
View File

@@ -79,7 +79,7 @@ Oscillator::Oscillator(const IntModel *wave_shape_model,
void Oscillator::update(sampleFrame* ab, const fpp_t frames, const ch_cnt_t chnl, bool modulator)
{
if (m_freq >= Engine::audioEngine()->processingSampleRate() / 2)
if (m_freq >= Engine::audioEngine()->outputSampleRate() / 2)
{
BufferManager::clear(ab, frames);
return;
@@ -681,7 +681,7 @@ void Oscillator::updateFM( sampleFrame * _ab, const fpp_t _frames,
m_subOsc->update( _ab, _frames, _chnl, true );
recalcPhase();
const float osc_coeff = m_freq * m_detuning_div_samplerate;
const float sampleRateCorrection = 44100.0f / Engine::audioEngine()->processingSampleRate();
const float sampleRateCorrection = 44100.0f / Engine::audioEngine()->outputSampleRate();
for( fpp_t frame = 0; frame < _frames; ++frame )
{
@@ -697,7 +697,7 @@ void Oscillator::updateFM( sampleFrame * _ab, const fpp_t _frames,
template<>
inline sample_t Oscillator::getSample<Oscillator::WaveShape::Sine>(const float sample)
{
const float current_freq = m_freq * m_detuning_div_samplerate * Engine::audioEngine()->processingSampleRate();
const float current_freq = m_freq * m_detuning_div_samplerate * Engine::audioEngine()->outputSampleRate();
if (!m_useWaveTable || current_freq < OscillatorConstants::MAX_FREQ)
{

2
src/core/PeakController.cpp
View File

@@ -80,7 +80,7 @@ void PeakController::updateValueBuffer()
{
if( m_coeffNeedsUpdate )
{
const float ratio = 44100.0f / Engine::audioEngine()->processingSampleRate();
const float ratio = 44100.0f / Engine::audioEngine()->outputSampleRate();
m_attackCoeff = 1.0f - powf( 2.0f, -0.3f * ( 1.0f - m_peakEffect->attackModel()->value() ) * ratio );
m_decayCoeff = 1.0f - powf( 2.0f, -0.3f * ( 1.0f - m_peakEffect->decayModel()->value() ) * ratio );
m_coeffNeedsUpdate = false;

2
src/core/RemotePlugin.cpp
View File

@@ -535,7 +535,7 @@ bool RemotePlugin::processMessage( const message & _m )
case IdSampleRateInformation:
reply = true;
reply_message.addInt( Engine::audioEngine()->processingSampleRate() );
reply_message.addInt( Engine::audioEngine()->outputSampleRate() );
break;
case IdBufferSizeInformation:

8
src/core/RingBuffer.cpp
View File

@@ -34,7 +34,7 @@ namespace lmms
RingBuffer::RingBuffer( f_cnt_t size ) :
m_fpp( Engine::audioEngine()->framesPerPeriod() ),
m_samplerate( Engine::audioEngine()->processingSampleRate() ),
m_samplerate( Engine::audioEngine()->outputSampleRate() ),
m_size( size + m_fpp )
{
m_buffer = new sampleFrame[ m_size ];
@@ -45,7 +45,7 @@ RingBuffer::RingBuffer( f_cnt_t size ) :
RingBuffer::RingBuffer( float size ) :
m_fpp( Engine::audioEngine()->framesPerPeriod() ),
m_samplerate( Engine::audioEngine()->processingSampleRate() )
m_samplerate( Engine::audioEngine()->outputSampleRate() )
{
m_size = msToFrames( size ) + m_fpp;
m_buffer = new sampleFrame[ m_size ];
@@ -307,9 +307,9 @@ void RingBuffer::writeSwappedAddingMultiplied( sampleFrame * src, float offset,
void RingBuffer::updateSamplerate()
{
float newsize = static_cast<float>( ( m_size - m_fpp ) * Engine::audioEngine()->processingSampleRate() ) / m_samplerate;
float newsize = static_cast<float>( ( m_size - m_fpp ) * Engine::audioEngine()->outputSampleRate() ) / m_samplerate;
m_size = static_cast<f_cnt_t>( ceilf( newsize ) ) + m_fpp;
m_samplerate = Engine::audioEngine()->processingSampleRate();
m_samplerate = Engine::audioEngine()->outputSampleRate();
delete[] m_buffer;
m_buffer = new sampleFrame[ m_size ];
memset( m_buffer, 0, m_size * sizeof( sampleFrame ) );

2
src/core/Sample.cpp
View File

@@ -124,7 +124,7 @@ bool Sample::play(sampleFrame* dst, PlaybackState* state, size_t numFrames, floa
const auto pastBounds = state->m_frameIndex >= m_endFrame || (state->m_frameIndex < 0 && state->m_backwards);
if (loopMode == Loop::Off && pastBounds) { return false; }
const auto outputSampleRate = Engine::audioEngine()->processingSampleRate() * m_frequency / desiredFrequency;
const auto outputSampleRate = Engine::audioEngine()->outputSampleRate() * m_frequency / desiredFrequency;
const auto inputSampleRate = m_buffer->sampleRate();
const auto resampleRatio = outputSampleRate / inputSampleRate;
const auto marginSize = s_interpolationMargins[state->resampler().interpolationMode()];

2
src/core/SampleClip.cpp
View File

@@ -307,7 +307,7 @@ void SampleClip::loadSettings( const QDomElement & _this )
if( sampleFile().isEmpty() && _this.hasAttribute( "data" ) )
{
auto sampleRate = _this.hasAttribute("sample_rate") ? _this.attribute("sample_rate").toInt() :
Engine::audioEngine()->processingSampleRate();
Engine::audioEngine()->outputSampleRate();
auto buffer = gui::SampleLoader::createBufferFromBase64(_this.attribute("data"), sampleRate);
m_sample = Sample(std::move(buffer));

2
src/core/SampleDecoder.cpp
View File

@@ -101,7 +101,7 @@ auto decodeSampleDS(const QString& audioFile) -> std::optional<SampleDecoder::Re
int_sample_t* dataPtr = nullptr;
auto ds = DrumSynth{};
const auto engineRate = Engine::audioEngine()->processingSampleRate();
const auto engineRate = Engine::audioEngine()->outputSampleRate();
const auto frames = ds.GetDSFileSamples(audioFile, dataPtr, DEFAULT_CHANNELS, engineRate);
const auto data = std::unique_ptr<int_sample_t[]>{dataPtr}; // NOLINT, we have to use a C-style array here

2
src/core/SamplePlayHandle.cpp
View File

@@ -145,7 +145,7 @@ bool SamplePlayHandle::isFromTrack( const Track * _track ) const
f_cnt_t SamplePlayHandle::totalFrames() const
{
return (m_sample->endFrame() - m_sample->startFrame()) *
(static_cast<float>(Engine::audioEngine()->processingSampleRate()) / m_sample->sampleRate());
(static_cast<float>(Engine::audioEngine()->outputSampleRate()) / m_sample->sampleRate());
}

2
src/core/VstSyncController.cpp
View File

@@ -155,7 +155,7 @@ void VstSyncController::updateSampleRate()
{
if (!m_syncData) { return; }
m_syncData->m_sampleRate = Engine::audioEngine()->processingSampleRate();
m_syncData->m_sampleRate = Engine::audioEngine()->outputSampleRate();
#ifdef VST_SNC_LATENCY
m_syncData->m_latency = m_syncData->m_bufferSize * m_syncData->m_bpm / ( (float) m_syncData->m_sampleRate * 60 );

90
src/core/audio/AudioDevice.cpp
View File

@@ -34,18 +34,11 @@ namespace lmms
AudioDevice::AudioDevice( const ch_cnt_t _channels, AudioEngine* _audioEngine ) :
m_supportsCapture( false ),
m_sampleRate( _audioEngine->processingSampleRate() ),
m_sampleRate( _audioEngine->outputSampleRate() ),
m_channels( _channels ),
m_audioEngine( _audioEngine ),
m_buffer( new surroundSampleFrame[audioEngine()->framesPerPeriod()] )
{
int error;
if( ( m_srcState = src_new(
audioEngine()->currentQualitySettings().libsrcInterpolation(),
SURROUND_CHANNELS, &error ) ) == nullptr )
{
printf( "Error: src_new() failed in audio_device.cpp!\n" );
}
}
@@ -53,9 +46,7 @@ AudioDevice::AudioDevice( const ch_cnt_t _channels, AudioEngine* _audioEngine )
AudioDevice::~AudioDevice()
{
src_delete( m_srcState );
delete[] m_buffer;
m_devMutex.tryLock();
unlock();
}
@@ -73,39 +64,16 @@ void AudioDevice::processNextBuffer()
}
}
fpp_t AudioDevice::getNextBuffer( surroundSampleFrame * _ab )
fpp_t AudioDevice::getNextBuffer(surroundSampleFrame* _ab)
{
fpp_t frames = audioEngine()->framesPerPeriod();
const surroundSampleFrame * b = audioEngine()->nextBuffer();
if( !b )
{
return 0;
}
// make sure, no other thread is accessing device
lock();
const surroundSampleFrame* b = audioEngine()->nextBuffer();
if (!b) { return 0; }
// resample if necessary
if( audioEngine()->processingSampleRate() != m_sampleRate )
{
frames = resample( b, frames, _ab, audioEngine()->processingSampleRate(), m_sampleRate );
}
else
{
memcpy( _ab, b, frames * sizeof( surroundSampleFrame ) );
}
// release lock
unlock();
if( audioEngine()->hasFifoWriter() )
{
delete[] b;
}
memcpy(_ab, b, frames * sizeof(surroundSampleFrame));
if (audioEngine()->hasFifoWriter()) { delete[] b; }
return frames;
}
@@ -141,23 +109,6 @@ void AudioDevice::stopProcessingThread( QThread * thread )
void AudioDevice::applyQualitySettings()
{
src_delete( m_srcState );
int error;
if( ( m_srcState = src_new(
audioEngine()->currentQualitySettings().libsrcInterpolation(),
SURROUND_CHANNELS, &error ) ) == nullptr )
{
printf( "Error: src_new() failed in audio_device.cpp!\n" );
}
}
void AudioDevice::registerPort( AudioPort * )
{
}
@@ -176,35 +127,6 @@ void AudioDevice::renamePort( AudioPort * )
{
}
fpp_t AudioDevice::resample( const surroundSampleFrame * _src,
const fpp_t _frames,
surroundSampleFrame * _dst,
const sample_rate_t _src_sr,
const sample_rate_t _dst_sr )
{
if( m_srcState == nullptr )
{
return _frames;
}
m_srcData.input_frames = _frames;
m_srcData.output_frames = _frames;
m_srcData.data_in = const_cast<float*>(_src[0].data());
m_srcData.data_out = _dst[0].data ();
m_srcData.src_ratio = (double) _dst_sr / _src_sr;
m_srcData.end_of_input = 0;
if (int error = src_process(m_srcState, &m_srcData))
{
printf( "AudioDevice::resample(): error while resampling: %s\n",
src_strerror( error ) );
}
return static_cast<fpp_t>(m_srcData.output_frames_gen);
}
int AudioDevice::convertToS16( const surroundSampleFrame * _ab,
const fpp_t _frames,
int_sample_t * _output_buffer,

8
src/core/lv2/Lv2Proc.cpp
View File

@@ -439,7 +439,7 @@ void Lv2Proc::initPlugin()
m_features.createFeatureVectors();
m_instance = lilv_plugin_instantiate(m_plugin,
Engine::audioEngine()->processingSampleRate(),
Engine::audioEngine()->outputSampleRate(),
m_features.featurePointers());
if (m_instance)
@@ -507,7 +507,7 @@ void Lv2Proc::initMOptions()
re-initialize, and this code section will be
executed again, creating a new option vector.
*/
float sampleRate = Engine::audioEngine()->processingSampleRate();
float sampleRate = Engine::audioEngine()->outputSampleRate();
int32_t blockLength = Engine::audioEngine()->framesPerPeriod();
int32_t sequenceSize = defaultEvbufSize();
@@ -568,7 +568,7 @@ void Lv2Proc::createPort(std::size_t portNum)
{
AutoLilvNode node(lilv_port_get_name(m_plugin, lilvPort));
QString dispName = lilv_node_as_string(node.get());
sample_rate_t sr = Engine::audioEngine()->processingSampleRate();
sample_rate_t sr = Engine::audioEngine()->outputSampleRate();
if(meta.def() < meta.min(sr) || meta.def() > meta.max(sr))
{
qWarning() << "Warning: Plugin"
@@ -871,7 +871,7 @@ void Lv2Proc::dumpPort(std::size_t num)
qDebug() << " visualization: " << Lv2Ports::toStr(port.m_vis);
if (port.m_type == Lv2Ports::Type::Control || port.m_type == Lv2Ports::Type::Cv)
{
sample_rate_t sr = Engine::audioEngine()->processingSampleRate();
sample_rate_t sr = Engine::audioEngine()->outputSampleRate();
qDebug() << " default:" << port.def();
qDebug() << " min:" << port.min(sr);
qDebug() << " max:" << port.max(sr);

33
src/core/main.cpp
View File

@@ -210,7 +210,6 @@ void printHelp()
" -p, --profile <out> Dump profiling information to file <out>\n"
" -s, --samplerate <samplerate> Specify output samplerate in Hz\n"
" Range: 44100 (default) to 192000\n"
" -x, --oversampling <value> Specify oversampling\n"
" Possible values: 1, 2, 4, 8\n"
" Default: 2\n\n",
LMMS_VERSION, LMMS_PROJECT_COPYRIGHT );
@@ -361,7 +360,7 @@ int main( int argc, char * * argv )
new QCoreApplication( argc, argv ) :
new gui::MainApplication(argc, argv);
AudioEngine::qualitySettings qs( AudioEngine::qualitySettings::Mode::HighQuality );
AudioEngine::qualitySettings qs(AudioEngine::qualitySettings::Interpolation::Linear);
OutputSettings os( 44100, OutputSettings::BitRateSettings(160, false), OutputSettings::BitDepth::Depth16Bit, OutputSettings::StereoMode::JointStereo );
ProjectRenderer::ExportFileFormat eff = ProjectRenderer::ExportFileFormat::Wave;
@@ -646,36 +645,6 @@ int main( int argc, char * * argv )
return usageError( QString( "Invalid interpolation method %1" ).arg( argv[i] ) );
}
}
else if( arg == "--oversampling" || arg == "-x" )
{
++i;
if( i == argc )
{
return usageError( "No oversampling specified" );
}
int o = QString( argv[i] ).toUInt();
switch( o )
{
case 1:
qs.oversampling = AudioEngine::qualitySettings::Oversampling::None;
break;
case 2:
qs.oversampling = AudioEngine::qualitySettings::Oversampling::X2;
break;
case 4:
qs.oversampling = AudioEngine::qualitySettings::Oversampling::X4;
break;
case 8:
qs.oversampling = AudioEngine::qualitySettings::Oversampling::X8;
break;
default:
return usageError( QString( "Invalid oversampling %1" ).arg( argv[i] ) );
}
}
else if( arg == "--import" )
{
++i;

2
src/gui/Lv2ViewBase.cpp
View File

@@ -74,7 +74,7 @@ Lv2ViewProc::Lv2ViewProc(QWidget* parent, Lv2Proc* proc, int colNum) :
break;
case PortVis::Integer:
{
sample_rate_t sr = Engine::audioEngine()->processingSampleRate();
sample_rate_t sr = Engine::audioEngine()->outputSampleRate();
auto pMin = port.min(sr);
auto pMax = port.max(sr);
int numDigits = std::max(numDigitsAsInt(pMin), numDigitsAsInt(pMax));

7
src/gui/modals/ExportProjectDialog.cpp
View File

@@ -154,11 +154,8 @@ OutputSettings::StereoMode mapToStereoMode(int index)
void ExportProjectDialog::startExport()
{
AudioEngine::qualitySettings qs =
AudioEngine::qualitySettings(
static_cast<AudioEngine::qualitySettings::Interpolation>(interpolationCB->currentIndex()),
static_cast<AudioEngine::qualitySettings::Oversampling>(oversamplingCB->currentIndex()) );
auto qs = AudioEngine::qualitySettings(
static_cast<AudioEngine::qualitySettings::Interpolation>(interpolationCB->currentIndex()));
const auto samplerates = std::array{44100, 48000, 88200, 96000, 192000};
const auto bitrates = std::array{64, 128, 160, 192, 256, 320};

2
src/gui/modals/SetupDialog.cpp
View File

@@ -1223,7 +1223,7 @@ void SetupDialog::setBufferSize(int value)
m_bufferSize = value * BUFFERSIZE_RESOLUTION;
m_bufferSizeLbl->setText(tr("Frames: %1\nLatency: %2 ms").arg(m_bufferSize).arg(
1000.0f * m_bufferSize / Engine::audioEngine()->processingSampleRate(), 0, 'f', 1));
1000.0f * m_bufferSize / Engine::audioEngine()->outputSampleRate(), 0, 'f', 1));
updateBufferSizeWarning(m_bufferSize);
}

31
src/gui/modals/export_project.ui
View File

@@ -404,37 +404,6 @@
</item>
</widget>
</item>
<item>
<widget class="QLabel" name="label_4">
<property name="text">
<string>Oversampling:</string>
</property>
</widget>
</item>
<item>
<widget class="QComboBox" name="oversamplingCB">
<item>
<property name="text">
<string>1x (None)</string>
</property>
</item>
<item>
<property name="text">
<string>2x</string>
</property>
</item>
<item>
<property name="text">
<string>4x</string>
</property>
</item>
<item>
<property name="text">
<string>8x</string>
</property>
</item>
</widget>
</item>
<item>
<spacer>
<property name="orientation">