I am using Superpowered for various real-time FX and they all work very straightforward. However the pitch shifting is a whole other story, I think in fact because it's based on the time-stretching algorithm that of course has to deal with output that changes in time which is a lot more complex than applying FX like EQ or reverb. However I'm only interested in change the pitch of my mic input.
I looked at the only example I could find on GitHub and I slightly adapted it to fit my work:
static bool audioProcessing(void *clientdata,
float **buffers,
unsigned int inputChannels,
unsigned int outputChannels,
unsigned int numberOfSamples,
unsigned int samplerate,
uint64_t hostTime) {
__unsafe_unretained Superpowered *self = (__bridge Superpowered *)clientdata;
SuperpoweredAudiobufferlistElement inputBuffer;
inputBuffer.startSample = 0;
inputBuffer.samplesUsed = 0;
inputBuffer.endSample = self->timeStretcher->numberOfInputSamplesNeeded;
inputBuffer.buffers[0] = SuperpoweredAudiobufferPool::getBuffer(self->timeStretcher->numberOfInputSamplesNeeded * 8 + 64);
inputBuffer.buffers[1] = inputBuffer.buffers[2] = inputBuffer.buffers[3] = NULL;
self->outputBuffers->clear();
self->timeStretcher->process(&inputBuffer, self->outputBuffers);
int samples = self->timeStretcher->numberOfInputSamplesNeeded;
float *timeStretchedAudio = (float *)self->outputBuffers->nextSliceItem(&samples);
if (timeStretchedAudio != 0) {
SuperpoweredDeInterleave(timeStretchedAudio, buffers[0], buffers[1], numberOfSamples);
}
//self->outputBuffers->rewindSlice();
return true;
}
I have removed most of the code that I thought wasn't necessary. For example there was a while loop that seemed to deal with time-stretch scenarios, I'm just outputting the same time as I input.
Some observations:
clear
the outputBuffers
my memory usage goes through the roofself->outputBuffers->rewindSlice();
the app becomes silent, probably meaning the buffers are getting overwritten with silenceself->outputBuffers->rewindSlice();
I can hear my own voice coming back, but timeStretchedAudio
is always 0
except the very first timeI finally got it working:
static bool audioProcessing(void *clientdata,
float **buffers,
unsigned int inputChannels,
unsigned int outputChannels,
unsigned int numberOfSamples,
unsigned int samplerate,
uint64_t hostTime) {
__unsafe_unretained Superpowered *self = (__bridge Superpowered *)clientdata;
//timeStretching->setRateAndPitchShift(realTimeRate, realTimePitch);
SuperpoweredAudiobufferlistElement inputBuffer;
inputBuffer.startSample = 0;
inputBuffer.samplesUsed = 0;
inputBuffer.endSample = numberOfSamples;
inputBuffer.buffers[0] = SuperpoweredAudiobufferPool::getBuffer((unsigned int) (numberOfSamples * 8 + 64));
inputBuffer.buffers[1] = inputBuffer.buffers[2] = inputBuffer.buffers[3] = NULL;
// Converting the 16-bit integer samples to 32-bit floating point.
SuperpoweredInterleave(buffers[0], buffers[1], (float *)inputBuffer.buffers[0], numberOfSamples);
//SuperpoweredShortIntToFloat(audioInputOutput, (float *)inputBuffer.buffers[0], (unsigned int) numberOfSamples);
self->timeStretcher->process(&inputBuffer, self->outputBuffers);
// Do we have some output?
if (self->outputBuffers->makeSlice(0, self->outputBuffers->sampleLength)) {
while (true) { // Iterate on every output slice.
// Get pointer to the output samples.
int numSamples = 0;
float *timeStretchedAudio = (float *)self->outputBuffers->nextSliceItem(&numSamples);
if (!timeStretchedAudio || *timeStretchedAudio == 0) {
break;
}
// Convert the time stretched PCM samples from 32-bit floating point to 16-bit integer.
//SuperpoweredFloatToShortInt(timeStretchedAudio, audioInputOutput,
// (unsigned int) numSamples);
SuperpoweredDeInterleave(timeStretchedAudio, buffers[0], buffers[1], numSamples);
self->recorder->process(timeStretchedAudio, numSamples);
// Write the audio to disk.
//fwrite(audioInputOutput, 1, numSamples * 4, fd);
}
// Clear the output buffer list.
self->outputBuffers->clear();
// If we have enough samples in the fifo output buffer, pass them to the audio output.
//SuperpoweredFloatToShortInt((float *)inputBuffer.buffers[0], audioInputOutput, (unsigned int) numberOfSamples);
}
return true;
}
I am not sure if changing the rate also works, but I don't care for this application. YMMV.