Make num harmonics parameter an integer parameter
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parent
ced5a5e22a
commit
65545c96b6
@ -128,9 +128,12 @@ PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()
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{
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{
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return jmap<float>(value, 101.0f, 1.0f, 0.0f, 1.0f);
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return jmap<float>(value, 101.0f, 1.0f, 0.0f, 1.0f);
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};
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};
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addParameter(new AudioParameterFloat("numharmonics0", "Num harmonics",
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/*
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addParameter(new AudioParameterFloat("numharmonics0", "Num harmonics",
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NormalisableRange<float>(1.0f, 101.0f,
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NormalisableRange<float>(1.0f, 101.0f,
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numhar_convertFrom0To1Func, numhar_convertTo0To1Func), 101.0f)); // 11
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numhar_convertFrom0To1Func, numhar_convertTo0To1Func), 101.0f)); // 11
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*/
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addParameter(new AudioParameterInt("numharmonics0", "Num harmonics", 1, 100, 10)); // 11
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addParameter(new AudioParameterFloat("harmonicsfreq0", "Harmonics base freq",
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addParameter(new AudioParameterFloat("harmonicsfreq0", "Harmonics base freq",
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NormalisableRange<float>(1.0f, 5000.0f, 1.00f, 0.5), 128.0f)); // 12
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NormalisableRange<float>(1.0f, 5000.0f, 1.00f, 0.5), 128.0f)); // 12
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addParameter(new AudioParameterFloat("harmonicsbw0", "Harmonics bandwidth", 0.1f, 200.0f, 25.0f)); // 13
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addParameter(new AudioParameterFloat("harmonicsbw0", "Harmonics bandwidth", 0.1f, 200.0f, 25.0f)); // 13
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@ -568,7 +571,7 @@ void PaulstretchpluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, M
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m_ppar.compressor.power = *getFloatParameter(cpi_compress);
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m_ppar.compressor.power = *getFloatParameter(cpi_compress);
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m_ppar.harmonics.nharmonics = *getFloatParameter(cpi_numharmonics);
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m_ppar.harmonics.nharmonics = *getIntParameter(cpi_numharmonics);
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m_ppar.harmonics.freq = *getFloatParameter(cpi_harmonicsfreq);
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m_ppar.harmonics.freq = *getFloatParameter(cpi_harmonicsfreq);
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m_ppar.harmonics.bandwidth = *getFloatParameter(cpi_harmonicsbw);
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m_ppar.harmonics.bandwidth = *getFloatParameter(cpi_harmonicsbw);
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m_ppar.harmonics.gauss = getParameter(cpi_harmonicsgauss);
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m_ppar.harmonics.gauss = getParameter(cpi_harmonicsgauss);
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@ -126,6 +126,10 @@ public:
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{
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{
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return dynamic_cast<AudioParameterFloat*>(getParameters()[index]);
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return dynamic_cast<AudioParameterFloat*>(getParameters()[index]);
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}
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}
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AudioParameterInt* getIntParameter(int index)
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{
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return dynamic_cast<AudioParameterInt*>(getParameters()[index]);
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}
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void setRecordingEnabled(bool b);
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void setRecordingEnabled(bool b);
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bool isRecordingEnabled() { return m_is_recording; }
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bool isRecordingEnabled() { return m_is_recording; }
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double getRecordingPositionPercent();
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double getRecordingPositionPercent();
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@ -44,7 +44,8 @@ History :
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provide the file name of the source of audio file, so audio clip/event/item specific audio won't be imported)
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provide the file name of the source of audio file, so audio clip/event/item specific audio won't be imported)
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-Removed an unnecessary level of buffering (doesn't reduce latency but should help a bit with CPU usage)
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-Removed an unnecessary level of buffering (doesn't reduce latency but should help a bit with CPU usage)
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-Added About window
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-Added About window
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01-21-2018 1.0.0 preview 5
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01-30-2018 1.0.0 preview 5
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-Added buttons to enable/disable spectral processing steps
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-Restored ability to set capture buffer length (via the settings menu)
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-Restored ability to set capture buffer length (via the settings menu)
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-Seek to play range beginning when audio file imported
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-Seek to play range beginning when audio file imported
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-Fixes to waveform display issues
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-Fixes to waveform display issues
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