mirror of
https://github.com/juce-framework/JUCE.git
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174 lines
5.8 KiB
C++
174 lines
5.8 KiB
C++
/*
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==============================================================================
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This file is part of the JUCE framework.
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Copyright (c) Raw Material Software Limited
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JUCE is an open source framework subject to commercial or open source
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licensing.
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By downloading, installing, or using the JUCE framework, or combining the
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JUCE framework with any other source code, object code, content or any other
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copyrightable work, you agree to the terms of the JUCE End User Licence
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Agreement, and all incorporated terms including the JUCE Privacy Policy and
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the JUCE Website Terms of Service, as applicable, which will bind you. If you
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do not agree to the terms of these agreements, we will not license the JUCE
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framework to you, and you must discontinue the installation or download
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process and cease use of the JUCE framework.
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JUCE End User Licence Agreement: https://juce.com/legal/juce-8-licence/
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JUCE Privacy Policy: https://juce.com/juce-privacy-policy
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JUCE Website Terms of Service: https://juce.com/juce-website-terms-of-service/
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Or:
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You may also use this code under the terms of the AGPLv3:
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https://www.gnu.org/licenses/agpl-3.0.en.html
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THE JUCE FRAMEWORK IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL
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WARRANTIES, WHETHER EXPRESSED OR IMPLIED, INCLUDING WARRANTY OF
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MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE, ARE 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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params.attack = static_cast<float> (attackTimeSecs);
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params.release = static_cast<float> (releaseTimeSecs);
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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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adsr.setSampleRate (sound->sourceSampleRate);
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adsr.setParameters (sound->params);
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adsr.noteOn();
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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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adsr.noteOff();
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}
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else
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{
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clearCurrentNote();
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adsr.reset();
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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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auto envelopeValue = adsr.getNextSample();
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l *= lgain * envelopeValue;
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r *= rgain * envelopeValue;
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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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