void JuceDemoPluginAudioProcessor::process (AudioBuffer<FloatType>& buffer,
MidiBuffer& midiMessages,
AudioBuffer<FloatType>& delayBuffer)
{
const int numSamples = buffer.getNumSamples();
// apply our gain-change to the incoming data..
applyGain (buffer, delayBuffer);
// 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
keyboardState.processNextMidiBuffer (midiMessages, 0, numSamples, true);
// and now get our synth to process these midi events and generate its output.
synth.renderNextBlock (buffer, midiMessages, 0, numSamples);
// Apply our delay effect to the new output..
applyDelay (buffer, delayBuffer);
// In case we have more outputs than inputs, we'll clear any output
// channels that didn't contain input data, (because these aren't
// guaranteed to be empty - they may contain garbage).
for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i)
buffer.clear (i, 0, numSamples);
// Now ask the host for the current time so we can store it to be displayed later...
updateCurrentTimeInfoFromHost();
}
void AudioSampleBuffer::applyGainRamp (const int channel,
const int startSample,
int numSamples,
float startGain,
float endGain) noexcept
{
if (startGain == endGain)
{
applyGain (channel, startSample, numSamples, startGain);
}
else
{
jassert (isPositiveAndBelow (channel, numChannels));
jassert (startSample >= 0 && startSample + numSamples <= size);
const float increment = (endGain - startGain) / numSamples;
float* d = channels [channel] + startSample;
while (--numSamples >= 0)
{
*d++ *= startGain;
startGain += increment;
}
}
}
void AudioSampleBuffer::applyGain (const int startSample,
const int numSamples,
const float gain) noexcept
{
for (int i = 0; i < numChannels; ++i)
applyGain (i, startSample, numSamples, gain);
}
void AudioMixer2::update(void)
{
audio_block_t *in, *out=NULL;
unsigned int channel;
for (channel=0; channel < 2; channel++) {
if (!out) {
out = receiveWritable(channel);
if (out) {
int32_t mult = multiplier[channel];
if (mult != 65536) applyGain(out->data, mult);
}
} else {
in = receiveReadOnly(channel);
if (in) {
applyGainThenAdd(out->data, in->data, multiplier[channel]);
release(in);
}
}
}
if (out) {
transmit(out);
release(out);
}
}
// static
void SampleUtil::applyAlternatingGain(CSAMPLE* pBuffer, CSAMPLE gain1,
CSAMPLE gain2, int iNumSamples) {
// This handles gain1 == CSAMPLE_GAIN_ONE && gain2 == CSAMPLE_GAIN_ONE as well.
if (gain1 == gain2) {
return applyGain(pBuffer, gain1, iNumSamples);
}
// note: LOOP VECTORIZED.
for (int i = 0; i < iNumSamples / 2; ++i) {
pBuffer[i * 2] *= gain1;
pBuffer[i * 2 + 1] *= gain2;
}
}
// static
void SampleUtil::applyAlternatingGain(CSAMPLE* pBuffer,
CSAMPLE gain1, CSAMPLE gain2,
int iNumSamples) {
// This handles gain1 == 1.0 && gain2 == 1.0f as well.
if (gain1 == gain2) {
return applyGain(pBuffer, gain1, iNumSamples);
}
for (int i = 0; i < iNumSamples; i += 2) {
pBuffer[i] *= gain1;
pBuffer[i+1] *= gain2;
}
}
// static
void SampleUtil::copyWithGain(CSAMPLE* pDest, const CSAMPLE* pSrc,
CSAMPLE gain, int iNumSamples) {
if (pDest == pSrc) {
return applyGain(pDest, gain, iNumSamples);
}
if (gain == 1.0f) {
memcpy(pDest, pSrc, sizeof(pDest[0]) * iNumSamples);
return;
}
if (gain == 0.0f) {
memset(pDest, 0, sizeof(pDest[0]) * iNumSamples);
return;
}
for (int i = 0; i < iNumSamples; ++i) {
pDest[i] = pSrc[i] * gain;
}
// OR! need to test which fares better
// memcpy(pDest, pSrc, sizeof(pDest[0]) * iNumSamples);
// applyGain(pDest, gain);
}
void AudioSampleBuffer::applyGain (const float gain) noexcept
{
applyGain (0, size, gain);
}
