mirror of
https://github.com/juce-framework/JUCE.git
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227 lines
6.9 KiB
C++
227 lines
6.9 KiB
C++
/*
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==============================================================================
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This file is part of the JUCE library - "Jules' Utility Class Extensions"
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Copyright 2004-11 by Raw Material Software Ltd.
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------------------------------------------------------------------------------
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JUCE can be redistributed and/or modified under the terms of the GNU General
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Public License (Version 2), as published by the Free Software Foundation.
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A copy of the license is included in the JUCE distribution, or can be found
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online at www.gnu.org/licenses.
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JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
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WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
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A PARTICULAR PURPOSE. See the GNU General Public License for more details.
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------------------------------------------------------------------------------
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To release a closed-source product which uses JUCE, commercial licenses are
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available: visit www.rawmaterialsoftware.com/juce for more information.
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==============================================================================
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*/
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SamplerSound::SamplerSound (const String& name_,
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AudioFormatReader& source,
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const BigInteger& midiNotes_,
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const int midiNoteForNormalPitch,
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const double attackTimeSecs,
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const double releaseTimeSecs,
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const double maxSampleLengthSeconds)
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: name (name_),
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midiNotes (midiNotes_),
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midiRootNote (midiNoteForNormalPitch)
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{
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sourceSampleRate = source.sampleRate;
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if (sourceSampleRate <= 0 || source.lengthInSamples <= 0)
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{
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length = 0;
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attackSamples = 0;
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releaseSamples = 0;
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}
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else
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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 = new AudioSampleBuffer (jmin (2, (int) source.numChannels), length + 4);
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source.read (data, 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 (const int midiNoteNumber)
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{
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return midiNotes [midiNoteNumber];
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}
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bool SamplerSound::appliesToChannel (const 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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: pitchRatio (0.0),
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sourceSamplePosition (0.0),
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lgain (0.0f),
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rgain (0.0f),
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isInAttack (false),
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isInRelease (false)
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{
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}
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SamplerVoice::~SamplerVoice()
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{
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}
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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 (const int midiNoteNumber,
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const float velocity,
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SynthesiserSound* s,
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const int /*currentPitchWheelPosition*/)
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{
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if (const SamplerSound* const sound = dynamic_cast <const SamplerSound*> (s))
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{
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pitchRatio = 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 = 0.0f;
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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 (const 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 (const int /*newValue*/)
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{
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}
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void SamplerVoice::controllerMoved (const int /*controllerNumber*/,
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const int /*newValue*/)
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{
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}
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//==============================================================================
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void SamplerVoice::renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples)
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{
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if (const SamplerSound* const playingSound = static_cast <SamplerSound*> (getCurrentlyPlayingSound().get()))
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{
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const float* const inL = playingSound->data->getSampleData (0, 0);
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const float* const inR = playingSound->data->getNumChannels() > 1
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? playingSound->data->getSampleData (1, 0) : nullptr;
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float* outL = outputBuffer.getSampleData (0, startSample);
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float* outR = outputBuffer.getNumChannels() > 1 ? outputBuffer.getSampleData (1, startSample) : nullptr;
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while (--numSamples >= 0)
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{
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const int pos = (int) sourceSamplePosition;
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const float alpha = (float) (sourceSamplePosition - pos);
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const float 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 (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 (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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