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Converted AudioSampleBuffer into a templated class that can use either float or double types. Used this to implement 64-bit audio plugin support in VST and AU

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This commit is contained in:
jules 2015-11-02 11:09:41 +00:00
parent ba672f03fb
commit c562cfc3cc
27 changed files with 1713 additions and 1104 deletions
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55
examples/audio plugin demo/Source/PluginProcessor.cpp
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@ -85,7 +85,20 @@ public:
// not interested in controllers in this case.
}
void renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples) override
void renderNextBlock (AudioBuffer<float>& outputBuffer, int startSample, int numSamples) override
{
processBlock (outputBuffer, startSample, numSamples);
}
void renderNextBlock (AudioBuffer<double>& outputBuffer, int startSample, int numSamples) override
{
processBlock (outputBuffer, startSample, numSamples);
}
private:
template <typename FloatType>
void processBlock (AudioBuffer<FloatType>& outputBuffer, int startSample, int numSamples)
{
if (angleDelta != 0.0)
{
@ -93,7 +106,8 @@ public:
{
while (--numSamples >= 0)
{
const float currentSample = (float) (sin (currentAngle) * level * tailOff);
const FloatType currentSample =
static_cast<FloatType> (std::sin (currentAngle) * level * tailOff);
for (int i = outputBuffer.getNumChannels(); --i >= 0;)
outputBuffer.addSample (i, startSample, currentSample);
@ -116,7 +130,7 @@ public:
{
while (--numSamples >= 0)
{
const float currentSample = (float) (sin (currentAngle) * level);
const FloatType currentSample = static_cast<FloatType> (std::sin (currentAngle) * level);
for (int i = outputBuffer.getNumChannels(); --i >= 0;)
outputBuffer.addSample (i, startSample, currentSample);
@ -128,7 +142,6 @@ public:
}
}
private:
double currentAngle, angleDelta, level, tailOff;
};
@ -183,7 +196,6 @@ const float defaultDelay = 0.5f;
//==============================================================================
JuceDemoPluginAudioProcessor::JuceDemoPluginAudioProcessor()
: delayBuffer (2, 12000)
{
// Set up our parameters. The base class will delete them for us.
addParameter (gain = new FloatParameter (defaultGain, "Gain"));
@ -213,7 +225,19 @@ void JuceDemoPluginAudioProcessor::prepareToPlay (double newSampleRate, int /*sa
// initialisation that you need..
synth.setCurrentPlaybackSampleRate (newSampleRate);
keyboardState.reset();
delayBuffer.clear();
if (isUsingDoublePrecision())
{
delayBufferDouble.setSize (2, 12000);
delayBufferFloat.setSize (1, 1);
}
else
{
delayBufferFloat.setSize (2, 12000);
delayBufferDouble.setSize (1, 1);
}
reset();
}
void JuceDemoPluginAudioProcessor::releaseResources()
@ -227,17 +251,21 @@ void JuceDemoPluginAudioProcessor::reset()
{
// Use this method as the place to clear any delay lines, buffers, etc, as it
// means there's been a break in the audio's continuity.
delayBuffer.clear();
delayBufferFloat.clear();
delayBufferDouble.clear();
}
void JuceDemoPluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
template <typename FloatType>
void JuceDemoPluginAudioProcessor::process (AudioBuffer<FloatType>& buffer,
MidiBuffer& midiMessages,
AudioBuffer<FloatType>& delayBuffer)
{
const int numSamples = buffer.getNumSamples();
int channel, dp = 0;
// Go through the incoming data, and apply our gain to it...
for (channel = 0; channel < getNumInputChannels(); ++channel)
buffer.applyGain (channel, 0, buffer.getNumSamples(), gain->getValue());
buffer.applyGain (channel, 0, buffer.getNumSamples(), static_cast<FloatType> (gain->getValue()));
// Now pass any incoming midi messages to our keyboard state object, and let it
// add messages to the buffer if the user is clicking on the on-screen keys
@ -249,15 +277,16 @@ void JuceDemoPluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, Midi
// Apply our delay effect to the new output..
for (channel = 0; channel < getNumInputChannels(); ++channel)
{
float* channelData = buffer.getWritePointer (channel);
float* delayData = delayBuffer.getWritePointer (jmin (channel, delayBuffer.getNumChannels() - 1));
FloatType* channelData = buffer.getWritePointer (channel);
FloatType* delayData = delayBuffer.getWritePointer (jmin (channel, delayBuffer.getNumChannels() - 1));
dp = delayPosition;
for (int i = 0; i < numSamples; ++i)
{
const float in = channelData[i];
const FloatType in = channelData[i];
channelData[i] += delayData[dp];
delayData[dp] = (delayData[dp] + in) * delay->getValue();
delayData[dp] = (delayData[dp] + in) * static_cast<FloatType> (delay->getValue());
if (++dp >= delayBuffer.getNumSamples())
dp = 0;
}