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JUCE/modules/juce_audio_formats/sampler/juce_Sampler.cpp

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C++

/*
==============================================================================
This file is part of the JUCE framework.
Copyright (c) Raw Material Software Limited
JUCE is an open source framework subject to commercial or open source
licensing.
By downloading, installing, or using the JUCE framework, or combining the
JUCE framework with any other source code, object code, content or any other
copyrightable work, you agree to the terms of the JUCE End User Licence
Agreement, and all incorporated terms including the JUCE Privacy Policy and
the JUCE Website Terms of Service, as applicable, which will bind you. If you
do not agree to the terms of these agreements, we will not license the JUCE
framework to you, and you must discontinue the installation or download
process and cease use of the JUCE framework.
JUCE End User Licence Agreement: https://juce.com/legal/juce-8-licence/
JUCE Privacy Policy: https://juce.com/juce-privacy-policy
JUCE Website Terms of Service: https://juce.com/juce-website-terms-of-service/
Or:
You may also use this code under the terms of the AGPLv3:
https://www.gnu.org/licenses/agpl-3.0.en.html
THE JUCE FRAMEWORK IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL
WARRANTIES, WHETHER EXPRESSED OR IMPLIED, INCLUDING WARRANTY OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE, ARE DISCLAIMED.
==============================================================================
*/
namespace juce
{
SamplerSound::SamplerSound (const String& soundName,
AudioFormatReader& source,
const BigInteger& notes,
int midiNoteForNormalPitch,
double attackTimeSecs,
double releaseTimeSecs,
double maxSampleLengthSeconds)
: name (soundName),
sourceSampleRate (source.sampleRate),
midiNotes (notes),
midiRootNote (midiNoteForNormalPitch)
{
if (sourceSampleRate > 0 && source.lengthInSamples > 0)
{
length = jmin ((int) source.lengthInSamples,
(int) (maxSampleLengthSeconds * sourceSampleRate));
data.reset (new AudioBuffer<float> (jmin (2, (int) source.numChannels), length + 4));
source.read (data.get(), 0, length + 4, 0, true, true);
params.attack = static_cast<float> (attackTimeSecs);
params.release = static_cast<float> (releaseTimeSecs);
}
}
SamplerSound::~SamplerSound()
{
}
bool SamplerSound::appliesToNote (int midiNoteNumber)
{
return midiNotes[midiNoteNumber];
}
bool SamplerSound::appliesToChannel (int /*midiChannel*/)
{
return true;
}
//==============================================================================
SamplerVoice::SamplerVoice() {}
SamplerVoice::~SamplerVoice() {}
bool SamplerVoice::canPlaySound (SynthesiserSound* sound)
{
return dynamic_cast<const SamplerSound*> (sound) != nullptr;
}
void SamplerVoice::startNote (int midiNoteNumber, float velocity, SynthesiserSound* s, int /*currentPitchWheelPosition*/)
{
if (auto* sound = dynamic_cast<const SamplerSound*> (s))
{
pitchRatio = std::pow (2.0, (midiNoteNumber - sound->midiRootNote) / 12.0)
* sound->sourceSampleRate / getSampleRate();
sourceSamplePosition = 0.0;
lgain = velocity;
rgain = velocity;
adsr.setSampleRate (sound->sourceSampleRate);
adsr.setParameters (sound->params);
adsr.noteOn();
}
else
{
jassertfalse; // this object can only play SamplerSounds!
}
}
void SamplerVoice::stopNote (float /*velocity*/, bool allowTailOff)
{
if (allowTailOff)
{
adsr.noteOff();
}
else
{
clearCurrentNote();
adsr.reset();
}
}
void SamplerVoice::pitchWheelMoved (int /*newValue*/) {}
void SamplerVoice::controllerMoved (int /*controllerNumber*/, int /*newValue*/) {}
//==============================================================================
void SamplerVoice::renderNextBlock (AudioBuffer<float>& outputBuffer, int startSample, int numSamples)
{
if (auto* playingSound = static_cast<SamplerSound*> (getCurrentlyPlayingSound().get()))
{
auto& data = *playingSound->data;
const float* const inL = data.getReadPointer (0);
const float* const inR = data.getNumChannels() > 1 ? data.getReadPointer (1) : nullptr;
float* outL = outputBuffer.getWritePointer (0, startSample);
float* outR = outputBuffer.getNumChannels() > 1 ? outputBuffer.getWritePointer (1, startSample) : nullptr;
while (--numSamples >= 0)
{
auto pos = (int) sourceSamplePosition;
auto alpha = (float) (sourceSamplePosition - pos);
auto invAlpha = 1.0f - alpha;
// just using a very simple linear interpolation here
float l = (inL[pos] * invAlpha + inL[pos + 1] * alpha);
float r = (inR != nullptr) ? (inR[pos] * invAlpha + inR[pos + 1] * alpha)
: l;
auto envelopeValue = adsr.getNextSample();
l *= lgain * envelopeValue;
r *= rgain * envelopeValue;
if (outR != nullptr)
{
*outL++ += l;
*outR++ += r;
}
else
{
*outL++ += (l + r) * 0.5f;
}
sourceSamplePosition += pitchRatio;
if (sourceSamplePosition > playingSound->length)
{
stopNote (0.0f, false);
break;
}
}
}
}
} // namespace juce