fix macOS build (following Projucer changes made in Windows, which removed /Applications/JUCE/modules from its headers). move JUCE headers under source control, so that Windows and macOS can both build against same version of JUCE. remove AUv3 target (I think it's an iOS thing, so it will never work with this macOS fluidsynth dylib).
This commit is contained in:
205
modules/juce_audio_formats/sampler/juce_Sampler.cpp
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205
modules/juce_audio_formats/sampler/juce_Sampler.cpp
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/*
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==============================================================================
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This file is part of the JUCE library.
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Copyright (c) 2017 - ROLI Ltd.
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JUCE is an open source library subject to commercial or open-source
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licensing.
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By using JUCE, you agree to the terms of both the JUCE 5 End-User License
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Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
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27th April 2017).
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End User License Agreement: www.juce.com/juce-5-licence
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Privacy Policy: www.juce.com/juce-5-privacy-policy
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Or: You may also use this code under the terms of the GPL v3 (see
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www.gnu.org/licenses).
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JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
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EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
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DISCLAIMED.
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==============================================================================
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*/
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namespace juce
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{
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SamplerSound::SamplerSound (const String& soundName,
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AudioFormatReader& source,
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const BigInteger& notes,
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int midiNoteForNormalPitch,
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double attackTimeSecs,
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double releaseTimeSecs,
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double maxSampleLengthSeconds)
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: name (soundName),
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sourceSampleRate (source.sampleRate),
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midiNotes (notes),
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midiRootNote (midiNoteForNormalPitch)
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{
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if (sourceSampleRate > 0 && source.lengthInSamples > 0)
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{
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length = jmin ((int) source.lengthInSamples,
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(int) (maxSampleLengthSeconds * sourceSampleRate));
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data.reset (new AudioBuffer<float> (jmin (2, (int) source.numChannels), length + 4));
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source.read (data.get(), 0, length + 4, 0, true, true);
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attackSamples = roundToInt (attackTimeSecs * sourceSampleRate);
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releaseSamples = roundToInt (releaseTimeSecs * sourceSampleRate);
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}
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}
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SamplerSound::~SamplerSound()
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{
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}
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bool SamplerSound::appliesToNote (int midiNoteNumber)
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{
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return midiNotes[midiNoteNumber];
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}
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bool SamplerSound::appliesToChannel (int /*midiChannel*/)
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{
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return true;
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}
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//==============================================================================
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SamplerVoice::SamplerVoice() {}
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SamplerVoice::~SamplerVoice() {}
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bool SamplerVoice::canPlaySound (SynthesiserSound* sound)
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{
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return dynamic_cast<const SamplerSound*> (sound) != nullptr;
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}
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void SamplerVoice::startNote (int midiNoteNumber, float velocity, SynthesiserSound* s, int /*currentPitchWheelPosition*/)
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{
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if (auto* sound = dynamic_cast<const SamplerSound*> (s))
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{
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pitchRatio = std::pow (2.0, (midiNoteNumber - sound->midiRootNote) / 12.0)
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* sound->sourceSampleRate / getSampleRate();
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sourceSamplePosition = 0.0;
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lgain = velocity;
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rgain = velocity;
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isInAttack = (sound->attackSamples > 0);
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isInRelease = false;
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if (isInAttack)
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{
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attackReleaseLevel = 0.0f;
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attackDelta = (float) (pitchRatio / sound->attackSamples);
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}
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else
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{
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attackReleaseLevel = 1.0f;
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attackDelta = 0.0f;
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}
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if (sound->releaseSamples > 0)
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releaseDelta = (float) (-pitchRatio / sound->releaseSamples);
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else
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releaseDelta = -1.0f;
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}
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else
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{
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jassertfalse; // this object can only play SamplerSounds!
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}
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}
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void SamplerVoice::stopNote (float /*velocity*/, bool allowTailOff)
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{
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if (allowTailOff)
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{
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isInAttack = false;
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isInRelease = true;
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}
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else
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{
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clearCurrentNote();
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}
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}
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void SamplerVoice::pitchWheelMoved (int /*newValue*/) {}
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void SamplerVoice::controllerMoved (int /*controllerNumber*/, int /*newValue*/) {}
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//==============================================================================
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void SamplerVoice::renderNextBlock (AudioBuffer<float>& outputBuffer, int startSample, int numSamples)
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{
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if (auto* playingSound = static_cast<SamplerSound*> (getCurrentlyPlayingSound().get()))
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{
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auto& data = *playingSound->data;
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const float* const inL = data.getReadPointer (0);
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const float* const inR = data.getNumChannels() > 1 ? data.getReadPointer (1) : nullptr;
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float* outL = outputBuffer.getWritePointer (0, startSample);
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float* outR = outputBuffer.getNumChannels() > 1 ? outputBuffer.getWritePointer (1, startSample) : nullptr;
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while (--numSamples >= 0)
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{
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auto pos = (int) sourceSamplePosition;
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auto alpha = (float) (sourceSamplePosition - pos);
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auto invAlpha = 1.0f - alpha;
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// just using a very simple linear interpolation here..
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float l = (inL[pos] * invAlpha + inL[pos + 1] * alpha);
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float r = (inR != nullptr) ? (inR[pos] * invAlpha + inR[pos + 1] * alpha)
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: l;
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l *= lgain;
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r *= rgain;
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if (isInAttack)
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{
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l *= attackReleaseLevel;
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r *= attackReleaseLevel;
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attackReleaseLevel += attackDelta;
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if (attackReleaseLevel >= 1.0f)
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{
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attackReleaseLevel = 1.0f;
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isInAttack = false;
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}
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}
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else if (isInRelease)
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{
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l *= attackReleaseLevel;
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r *= attackReleaseLevel;
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attackReleaseLevel += releaseDelta;
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if (attackReleaseLevel <= 0.0f)
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{
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stopNote (0.0f, false);
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break;
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}
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}
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if (outR != nullptr)
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{
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*outL++ += l;
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*outR++ += r;
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}
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else
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{
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*outL++ += (l + r) * 0.5f;
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}
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sourceSamplePosition += pitchRatio;
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if (sourceSamplePosition > playingSound->length)
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{
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stopNote (0.0f, false);
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break;
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}
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}
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}
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}
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} // namespace juce
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