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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
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@@ -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);
};