JUCE/modules/juce_audio_formats/sampler/juce_Sampler.cpp

/*
  ==============================================================================

   This file is part of the JUCE 6 technical preview.
   Copyright (c) 2020 - Raw Material Software Limited

   You may use this code under the terms of the GPL v3
   (see www.gnu.org/licenses).

   For this technical preview, this file is not subject to commercial licensing.

   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR 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