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dexed/JuceLibraryCode/modules/juce_audio_formats/sampler/juce_Sampler.cpp

225 lines
6.8 KiB

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/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
SamplerSound::SamplerSound (const String& soundName,
AudioFormatReader& source,
const BigInteger& notes,
const int midiNoteForNormalPitch,
const double attackTimeSecs,
const double releaseTimeSecs,
const double maxSampleLengthSeconds)
: name (soundName),
midiNotes (notes),
midiRootNote (midiNoteForNormalPitch)
{
sourceSampleRate = source.sampleRate;
if (sourceSampleRate <= 0 || source.lengthInSamples <= 0)
{
length = 0;
attackSamples = 0;
releaseSamples = 0;
}
else
{
length = jmin ((int) source.lengthInSamples,
(int) (maxSampleLengthSeconds * sourceSampleRate));
data = new AudioSampleBuffer (jmin (2, (int) source.numChannels), length + 4);
source.read (data, 0, length + 4, 0, true, true);
attackSamples = roundToInt (attackTimeSecs * sourceSampleRate);
releaseSamples = roundToInt (releaseTimeSecs * sourceSampleRate);
}
}
SamplerSound::~SamplerSound()
{
}
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bool SamplerSound::appliesToNote (int midiNoteNumber)
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{
return midiNotes [midiNoteNumber];
}
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bool SamplerSound::appliesToChannel (int /*midiChannel*/)
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{
return true;
}
//==============================================================================
SamplerVoice::SamplerVoice()
: pitchRatio (0.0),
sourceSamplePosition (0.0),
lgain (0.0f), rgain (0.0f),
attackReleaseLevel (0), attackDelta (0), releaseDelta (0),
isInAttack (false), isInRelease (false)
{
}
SamplerVoice::~SamplerVoice()
{
}
bool SamplerVoice::canPlaySound (SynthesiserSound* sound)
{
return dynamic_cast<const SamplerSound*> (sound) != nullptr;
}
void SamplerVoice::startNote (const int midiNoteNumber,
const float velocity,
SynthesiserSound* s,
const int /*currentPitchWheelPosition*/)
{
if (const SamplerSound* const sound = dynamic_cast <const SamplerSound*> (s))
{
pitchRatio = pow (2.0, (midiNoteNumber - sound->midiRootNote) / 12.0)
* sound->sourceSampleRate / getSampleRate();
sourceSamplePosition = 0.0;
lgain = velocity;
rgain = velocity;
isInAttack = (sound->attackSamples > 0);
isInRelease = false;
if (isInAttack)
{
attackReleaseLevel = 0.0f;
attackDelta = (float) (pitchRatio / sound->attackSamples);
}
else
{
attackReleaseLevel = 1.0f;
attackDelta = 0.0f;
}
if (sound->releaseSamples > 0)
releaseDelta = (float) (-pitchRatio / sound->releaseSamples);
else
releaseDelta = 0.0f;
}
else
{
jassertfalse; // this object can only play SamplerSounds!
}
}
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void SamplerVoice::stopNote (float /*velocity*/, bool allowTailOff)
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{
if (allowTailOff)
{
isInAttack = false;
isInRelease = true;
}
else
{
clearCurrentNote();
}
}
void SamplerVoice::pitchWheelMoved (const int /*newValue*/)
{
}
void SamplerVoice::controllerMoved (const int /*controllerNumber*/,
const int /*newValue*/)
{
}
//==============================================================================
void SamplerVoice::renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples)
{
if (const SamplerSound* const playingSound = static_cast <SamplerSound*> (getCurrentlyPlayingSound().get()))
{
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const float* const inL = playingSound->data->getReadPointer (0);
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const float* const inR = playingSound->data->getNumChannels() > 1
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? playingSound->data->getReadPointer (1) : nullptr;
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float* outL = outputBuffer.getWritePointer (0, startSample);
float* outR = outputBuffer.getNumChannels() > 1 ? outputBuffer.getWritePointer (1, startSample) : nullptr;
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while (--numSamples >= 0)
{
const int pos = (int) sourceSamplePosition;
const float alpha = (float) (sourceSamplePosition - pos);
const float 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;
l *= lgain;
r *= rgain;
if (isInAttack)
{
l *= attackReleaseLevel;
r *= attackReleaseLevel;
attackReleaseLevel += attackDelta;
if (attackReleaseLevel >= 1.0f)
{
attackReleaseLevel = 1.0f;
isInAttack = false;
}
}
else if (isInRelease)
{
l *= attackReleaseLevel;
r *= attackReleaseLevel;
attackReleaseLevel += releaseDelta;
if (attackReleaseLevel <= 0.0f)
{
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stopNote (0.0f, false);
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break;
}
}
if (outR != nullptr)
{
*outL++ += l;
*outR++ += r;
}
else
{
*outL++ += (l + r) * 0.5f;
}
sourceSamplePosition += pitchRatio;
if (sourceSamplePosition > playingSound->length)
{
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stopNote (0.0f, false);
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break;
}
}
}
}