Chromagram SpectrumAnalyser::chromagramOfFirstFrame(
const AudioData& audio
) const {
if (audio.getChannels() != 1)
throw Exception("Audio must be monophonic to be analysed");
Chromagram c(1, octaves, bandsPerSemitone);
unsigned int fftFrameSize = fft->getFrameSize();
for (unsigned int sample = 0; sample < fftFrameSize; sample++) {
fft->setInput(sample, audio.getSample(sample) * temporalWindow[sample]);
}
fft->execute();
std::vector<float> cv = ct->chromaVector(fft);
for (unsigned int band = 0; band < c.getBands(); band++) {
c.setMagnitude(0, band, cv[band]);
}
return c;
}
void LowPassFilterPrivate::filter(AudioData& audio, Workspace& workspace, unsigned int shortcutFactor) const {
if (audio.getChannels() > 1) {
throw Exception("Monophonic audio only");
}
std::vector<double>* buffer = workspace.lpfBuffer;
if (buffer == NULL) {
workspace.lpfBuffer = new std::vector<double>(impulseLength, 0.0);
buffer = workspace.lpfBuffer;
} else {
// clear delay buffer
std::vector<double>::iterator bufferIterator = buffer->begin();
while (bufferIterator < buffer->end()) {
*bufferIterator = 0.0;
std::advance(bufferIterator, 1);
}
}
std::vector<double>::iterator bufferFront = buffer->begin();
std::vector<double>::iterator bufferBack;
std::vector<double>::iterator bufferTemp;
unsigned int sampleCount = audio.getSampleCount();
audio.resetIterators();
double sum;
// for each frame (running off the end of the sample stream by delay)
for (unsigned int inSample = 0; inSample < sampleCount + delay; inSample++) {
// shuffle old samples along delay buffer
bufferBack = bufferFront;
std::advance(bufferFront, 1);
if (bufferFront == buffer->end()) {
bufferFront = buffer->begin();
}
// load new sample into back of delay buffer
if (audio.readIteratorWithinUpperBound()) {
*bufferBack = audio.getSampleAtReadIterator() / gain;
audio.advanceReadIterator();
} else {
*bufferBack = 0.0; // zero pad once we're past the end of the file
}
// start doing the maths once the delay has passed
int outSample = (signed)inSample - (signed)delay;
if (outSample < 0) {
continue;
}
// and, if shortcut != 1, only do the maths for the useful samples (this is mathematically dodgy, but it's faster and it usually works)
if (outSample % shortcutFactor > 0) {
continue;
}
sum = 0.0;
bufferTemp = bufferFront;
std::vector<double>::const_iterator coefficientIterator = coefficients.begin();
while (coefficientIterator < coefficients.end()) {
sum += *coefficientIterator * *bufferTemp;
std::advance(coefficientIterator, 1);
std::advance(bufferTemp, 1);
if (bufferTemp == buffer->end()) {
bufferTemp = buffer->begin();
}
}
audio.setSampleAtWriteIterator(sum);
audio.advanceWriteIterator(shortcutFactor);
}
}
//Decodes the data in a WAV file and saves it into an AudioData class
AudioData * coder::decode(string filename){
fstream audioFile;
audioFile.open(filename, ios_base::in | ios_base::binary);
//File does not exist
if(audioFile.fail()){
return NULL;
}
AudioData * data = NULL;
if(audioFile.is_open()){
data = new AudioData;
//Read RIFF
int smallBuffer;
audioFile.read((char *)&smallBuffer, 4);
data->setRIFF((char *)&smallBuffer);
//Read file size
audioFile.read((char *)&smallBuffer, 4);
data->setFileSize(smallBuffer);
//Read file type
audioFile.read((char *)&smallBuffer, 4);
data->setFileType((char *)&smallBuffer);
//Read format
audioFile.read((char *)&smallBuffer, 4);
data->setFormat((char *)&smallBuffer);
//Read format info size
unsigned int infoSize = 0;
audioFile.read((char *)&infoSize, 4);
data->setFormatInfoSize(infoSize);
//Read format info
char * largeBuffer = (char *)malloc(sizeof(char) * infoSize); //remember to free on close
memset(largeBuffer, 0x00, infoSize);
audioFile.read(largeBuffer, infoSize);
data->setFormatInfo(largeBuffer);
//Read data chunk
audioFile.read((char *)&smallBuffer, 4);
data->setDataChunk((char *)&smallBuffer);
//Read data size
unsigned int dataSize = 0;
audioFile.read((char *)&dataSize, 4);
data->setDataSize(dataSize);
//Read audio data
largeBuffer = (char *)malloc(sizeof(char) * dataSize);
memset(largeBuffer, 0x00, dataSize);
audioFile.read(largeBuffer, dataSize);
//Seperate audio data into multiple channels
short ** channelData = seperateChannels((short *)largeBuffer, data->getChannels(), data->getNumberOfSamples());
data->setChannelData(channelData);
free(largeBuffer);
audioFile.close();
}
return data;
}