void ofApp::setup() {
ofSetVerticalSync(true);
string filename = "Serato/Serato_CD.aif";// "TraktorMK2/Traktor_MK2_Scribble.wav";
string absoluteFilename = ofToDataPath(filename, true);
SndfileHandle myf = SndfileHandle(absoluteFilename.c_str());
bufferFrames = myf.frames();
int n = bufferFrames * myf.channels();
floatBuffer.resize(n);
curBuffer.resize(n);
myf.read(&floatBuffer[0], n);
bufferPosition = 0;
ttmPosition = 0;
relativePosition = 0;
relativeTtm.resize(ofGetWidth());
absoluteTtm.resize(ofGetWidth());
pitchTtm.resize(ofGetWidth());
string timecode = "serato_cd"; // "serato_cd" "serato_a" "traktor_a"
float speed = 1.0; // 1.0 is 33 1/3, 1.35 is 45 rpm
int sampleRate = 44100; // myf.samplerate()
timecoder_init(&timecoder, timecode.c_str(), speed, sampleRate);
//timecoder_monitor_init(&timecoder, MIXXX_VINYL_SCOPE_SIZE);
bufferSize = 256;
exporting = true;
soundStream.setup(this, 2, 0, sampleRate, bufferSize, 4);
}
//--------------------------------------------------------------
void ofxAudioSample::load(string tmpPath, float _hSampleRate) {
myPath = ofToDataPath(tmpPath,true).c_str();
SndfileHandle sndFile = SndfileHandle(myPath);
myFormat = sndFile.format();
myChannels = sndFile.channels();
mySampleRate = sndFile.samplerate();
resampligFactor = _hSampleRate/mySampleRate;
speed = mainSpeed/resampligFactor;
bufferSize = 4096 * myChannels;
readBuffer = new float[bufferSize];
ofVec2f _wF;
int readcount;
int readpointer;
// convert all multichannel files to mono by averaging the channels
float monoAverage;
while(readcount = sndFile.readf(readBuffer, 4096)){
readpointer = 0;
_wF.set(0,0);
for (int i = 0; i < readcount; i++) {
// for each frame...
monoAverage = 0;
for(int j = 0; j < myChannels; j++) {
monoAverage += readBuffer[readpointer + j];
}
monoAverage /= myChannels;
readpointer += myChannels;
// add the averaged sample to our vector of samples
samples.push_back(monoAverage);
// add to the waveform data
_wF.x = MIN(_wF.x, monoAverage);
_wF.y = MAX(_wF.y, monoAverage);
}
_waveForm.push_back(_wF);
}
position = 0;
}
static void
read_file (const char * fname)
{ static short buffer [BUFFER_LEN] ;
SndfileHandle file ;
file = SndfileHandle (fname) ;
printf ("Opened file '%s'\n", fname) ;
printf (" Sample rate : %d\n", file.samplerate ()) ;
printf (" Channels : %d\n", file.channels ()) ;
file.read (buffer, BUFFER_LEN) ;
puts ("") ;
/* RAII takes care of destroying SndfileHandle object. */
} /* read_file */
void Sound::initData(SndfileHandle sndFile)
{
nChannels = sndFile.channels();
nFrames = sndFile.frames();
int length = nChannels * nFrames;
float buffer [length];
sndFile.read(buffer, length);
data.resize(nChannels, std::vector<float>());
for(int channelNum = 0; channelNum < nChannels; ++channelNum)
{
data[channelNum].resize(nFrames, 0.0);
for(int frameNum = 0; frameNum < nFrames; ++frameNum)
{
int i = frameNum * nChannels + channelNum;
data[channelNum][frameNum] = buffer[i];
}
}
dataInitialized = true;
}
/**
General loading function.
This is used by the both the load Samples and load Groundtruth
*/
void AudioManager::loadFiles (string dirname, vector<Sample> &into) {
vector<string> filenames = list_all_files(dirname);
for (vector<string>::iterator it = filenames.begin(); it != filenames.end(); ++it) {
string filename = *it;
if (filename.find(".wav") == string::npos) continue;
Sample sample;
SndfileHandle ff = SndfileHandle(filename);
assert(ff.channels() == 1);
assert(ff.samplerate() == 8000);
double * signal = new double [ff.frames()];
ff.read(signal, ff.frames());
sample.station = "";
sample.offset = 0;
sample.filename = filename;
sample.audio = signal;
sample.length = ff.frames();
into.push_back(sample);
}
}
int main () {
// Damit das Programm funktioniert, muss eine 16Bit PCM Wave-Datei im
// gleichen Ordner liegen !
const char * fname = "test.flac" ;
// Soundfile-Handle aus der libsndfile-Bibliothek
SndfileHandle file = SndfileHandle (fname) ;
// Alle möglichen Infos über die Audio-Datei ausgeben !
std::cout << "Reading file: " << fname << std::endl;
std::cout << "File format: " << file.format() << std::endl;
std::cout << "PCM 16 BIT: " << (SF_FORMAT_WAV | SF_FORMAT_PCM_16) << std::endl;
std::cout << "Samples in file: " << file.frames() << std::endl;
std::cout << "Samplerate " << file.samplerate() << std::endl;
std::cout << "Channels: " << file.channels() << std::endl;
// Die RtAudio-Klasse ist gleichermassen dac und adc, wird hier aber nur als dac verwendet !
RtAudio dac;
if ( dac.getDeviceCount() < 1 ) {
std::cout << "\nNo audio devices found!\n";
return 0;
}
// Output params ...
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = 2;
parameters.firstChannel = 0;
unsigned int sampleRate = 44100;
// ACHTUNG! Frames != Samples
// ein Frame = Samples für alle Kanäle
// d.h. |Samples| = Kanäle*Frames !
unsigned int bufferFrames = 1024;
// Da wir 16 Bit PCM-Daten lesen, sollte als Datenformat RTAUDIO_SINT16 genutzt
// werden.
// Als Daten wird der Callback-Struktur hier das Soundfile-Handle übergeben.
// Sollte man in einer "ernsthaften" Lösung anders machen !
// Inkompatible Formate können übrigens "interessante" Effekte ergeben !
try {
dac.openStream( ¶meters, NULL, RTAUDIO_SINT16,
sampleRate, &bufferFrames, &fplay, (void *)&file);
dac.startStream();
}
catch ( RtAudioError& e ) {
e.printMessage();
return 0;
}
char input;
std::cout << "\nPlaying ... press <enter> to quit.\n";
std::cin.get( input );
try {
// Stop the stream
dac.stopStream();
}
catch (RtAudioError& e) {
e.printMessage();
}
if ( dac.isStreamOpen() ) dac.closeStream();
return 0 ;
}
