void EqualPowerPanner::pan(double azimuth, double /*elevation*/, AudioBus* inputBus, AudioBus* outputBus, size_t framesToProcess)
{
// FIXME: implement stereo sources
bool isInputSafe = inputBus && inputBus->numberOfChannels() == 1 && framesToProcess <= inputBus->length();
ASSERT(isInputSafe);
if (!isInputSafe)
return;
bool isOutputSafe = outputBus && outputBus->numberOfChannels() == 2 && framesToProcess <= outputBus->length();
ASSERT(isOutputSafe);
if (!isOutputSafe)
return;
AudioChannel* channel = inputBus->channel(0);
float* sourceP = channel->data();
float* destinationL = outputBus->channelByType(AudioBus::ChannelLeft)->data();
float* destinationR = outputBus->channelByType(AudioBus::ChannelRight)->data();
if (!sourceP || !destinationL || !destinationR)
return;
// Pan smoothly from left to right with azimuth going from -30 -> +30 degrees.
double desiredPanPosition;
if (azimuth > 30.0)
desiredPanPosition = 1.0;
else if (azimuth < -30.0)
desiredPanPosition = 0.0;
else
desiredPanPosition = (azimuth + 30.0) / 60.0;
double desiredGainL = 0.5 * cos(piDouble * desiredPanPosition) + 0.5;
double desiredGainR = sqrt(1.0 - desiredGainL*desiredGainL);
// Don't de-zipper on first render call.
if (m_isFirstRender) {
m_isFirstRender = false;
m_gainL = desiredGainL;
m_gainR = desiredGainR;
}
// Cache in local variables.
double gainL = m_gainL;
double gainR = m_gainR;
// Get local copy of smoothing constant.
const double SmoothingConstant = m_smoothingConstant;
int n = framesToProcess;
while (n--) {
float input = *sourceP++;
gainL += (desiredGainL - gainL) * SmoothingConstant;
gainR += (desiredGainR - gainR) * SmoothingConstant;
*destinationL++ = static_cast<float>(input * gainL);
*destinationR++ = static_cast<float>(input * gainR);
}
m_gainL = gainL;
m_gainR = gainR;
}
// Pulls on our provider to get rendered audio stream.
OSStatus AudioDestinationMac::render(UInt32 numberOfFrames, AudioBufferList* ioData)
{
AudioBuffer* buffers = ioData->mBuffers;
m_renderBus.setChannelMemory(0, (float*)buffers[0].mData, numberOfFrames);
m_renderBus.setChannelMemory(1, (float*)buffers[1].mData, numberOfFrames);
//@tofix - add support for local/live audio input.
m_callback.render(m_input->m_audioBus, &m_renderBus, numberOfFrames);
// Clamp values at 0db (i.e., [-1.0, 1.0])
for (unsigned i = 0; i < m_renderBus.numberOfChannels(); ++i) {
AudioChannel* channel = m_renderBus.channel(i);
VectorMath::vclip(channel->data(), 1, &kLowThreshold, &kHighThreshold, channel->mutableData(), 1, numberOfFrames);
}
return noErr;
}
void Reverb::process(const AudioBus* sourceBus, AudioBus* destinationBus, size_t framesToProcess)
{
// Do a fairly comprehensive sanity check.
// If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
bool isSafeToProcess = sourceBus && destinationBus && sourceBus->numberOfChannels() > 0 && destinationBus->numberOfChannels() > 0
&& framesToProcess <= MaxFrameSize && framesToProcess <= sourceBus->length() && framesToProcess <= destinationBus->length();
ASSERT(isSafeToProcess);
if (!isSafeToProcess)
return;
// For now only handle mono or stereo output
if (destinationBus->numberOfChannels() > 2) {
destinationBus->zero();
return;
}
AudioChannel* destinationChannelL = destinationBus->channel(0);
const AudioChannel* sourceChannelL = sourceBus->channel(0);
// Handle input -> output matrixing...
size_t numInputChannels = sourceBus->numberOfChannels();
size_t numOutputChannels = destinationBus->numberOfChannels();
size_t numReverbChannels = m_convolvers.size();
if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
// 2 -> 2 -> 2
const AudioChannel* sourceChannelR = sourceBus->channel(1);
AudioChannel* destinationChannelR = destinationBus->channel(1);
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
m_convolvers[1]->process(sourceChannelR, destinationChannelR, framesToProcess);
} else if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
// 1 -> 2 -> 2
for (int i = 0; i < 2; ++i) {
AudioChannel* destinationChannel = destinationBus->channel(i);
m_convolvers[i]->process(sourceChannelL, destinationChannel, framesToProcess);
}
} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
// 1 -> 1 -> 2
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
// simply copy L -> R
AudioChannel* destinationChannelR = destinationBus->channel(1);
bool isCopySafe = destinationChannelL->data() && destinationChannelR->data() && destinationChannelL->length() >= framesToProcess && destinationChannelR->length() >= framesToProcess;
ASSERT(isCopySafe);
if (!isCopySafe)
return;
memcpy(destinationChannelR->mutableData(), destinationChannelL->data(), sizeof(float) * framesToProcess);
} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
// 1 -> 1 -> 1
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
} else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
// 2 -> 4 -> 2 ("True" stereo)
const AudioChannel* sourceChannelR = sourceBus->channel(1);
AudioChannel* destinationChannelR = destinationBus->channel(1);
AudioChannel* tempChannelL = m_tempBuffer->channel(0);
AudioChannel* tempChannelR = m_tempBuffer->channel(1);
// Process left virtual source
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
// Process right virtual source
m_convolvers[2]->process(sourceChannelR, tempChannelL, framesToProcess);
m_convolvers[3]->process(sourceChannelR, tempChannelR, framesToProcess);
destinationBus->sumFrom(*m_tempBuffer);
} else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
// 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
// This is an inefficient use of a four-channel impulse response, but we should handle the case.
AudioChannel* destinationChannelR = destinationBus->channel(1);
AudioChannel* tempChannelL = m_tempBuffer->channel(0);
AudioChannel* tempChannelR = m_tempBuffer->channel(1);
// Process left virtual source
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
// Process right virtual source
m_convolvers[2]->process(sourceChannelL, tempChannelL, framesToProcess);
m_convolvers[3]->process(sourceChannelL, tempChannelR, framesToProcess);
destinationBus->sumFrom(*m_tempBuffer);
} else {
// Handle gracefully any unexpected / unsupported matrixing
// FIXME: add code for 5.1 support...
destinationBus->zero();
}
}
