// Call
int fplay( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
double streamTime, RtAudioStreamStatus status, void *userData )
{
// Da wir RTAUDIO_SINT16 als Format angegeben haben, macht dieser Datentyp
// Sinn ! Btw: Bei Inkompatiblen Datentypen können sich ebenfalls "interessante"
// Effekte ergeben !
int16_t *buffer = (int16_t *) outputBuffer;
// Daten zurückcasten ... 'reinterpret_cast' hat keine Sicherheitsbedenken und castet alles
// in alles andere !
SndfileHandle *sndfile = reinterpret_cast<SndfileHandle*>(userData);
// Error handling !
if ( status ){
std::cout << "Stream underflow detected!" << std::endl;
}
// Direkt aus der Soundfile in den OutputBuffer lesen!
// Für das Beispiel reicht das, allerdings sollte man in einem 'ernsthaften' Programm
// eine andere Lösung finden. Disk I/O ist auch in Zeiten der SSDs eher langsam und
// sollte nicht direkt im Audio-Thread stattfinden !
// 'readf()' liest frames
// 'read()' liest einzelne Samples !
// ACHTUNG! Frames != Samples
// ein Frame = Samples für alle Kanäle
// d.h. |Samples| = Kanäle * Frames !
sndfile->readf(buffer, nBufferFrames);
return 0;
}
static void
create_file (const char * fname, int format)
{ static short buffer [BUFFER_LEN] ;
SndfileHandle file ;
int channels = 2 ;
int srate = 48000 ;
printf ("Creating file named '%s'\n", fname) ;
file = SndfileHandle (fname, SFM_WRITE, format, channels, srate) ;
memset (buffer, 0, sizeof (buffer)) ;
const int size = srate*3;
float sample[size];
float current =0;
for(int i =0; i<size;i++) sample[i] = sin(float(i)/size*M_PI*1500);
file.write (&sample[0], size) ;
puts ("") ;
/*
** The SndfileHandle object will automatically close the file and
** release all allocated memory when the object goes out of scope.
** This is the Resource Acquisition Is Initailization idom.
** See : http://en.wikipedia.org/wiki/Resource_Acquisition_Is_Initialization
*/
} /* create_file */
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 CSignalRecorder::StoreCapturedSamples(SndfileHandle &CaptureWave)
{
ulong capturedSamples = 0;
do
{
capturedSamples = m_CaptureDevice->GetSamples(m_SamplesBuffer->get_Frame(0), m_FrameSize*m_MaxCaptureChannels);
capturedSamples /= m_MaxCaptureChannels;
if(capturedSamples > 0)
{
ulong writtenSamples = 0;
if(m_CaptureDevice->get_InputChannelAmount() > 1)
{
writtenSamples = (ulong)CaptureWave.writef(m_SamplesBuffer->get_Frame(0), capturedSamples);
}
else
{
writtenSamples = (ulong)CaptureWave.write(m_SamplesBuffer->get_Frame(0), capturedSamples);
}
if(writtenSamples < capturedSamples)
{
KODI->Log(LOG_ERROR, "Failed to write to capture wave file!");
Set_State(STATE_INVALID);
}
}
}while(capturedSamples > 0);
}
static int buffer_read_verify(SndfileHandle const & sf, size_t min_length, size_t samplerate)
{
if (!sf)
return -1;
if (sf.frames() < min_length)
return -2; /* no more frames to read */
if (sf.samplerate() != samplerate)
return -3; /* sample rate mismatch */
return 0;
}
void AudioManager::testCropping (int sId, int gId) {
Sample cropped = getGroundTruthAround(sId, gId);
SF_INFO sfinfo;
sfinfo.channels = 1;
sfinfo.samplerate = FS;
sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
string file = "output/cropped.wav";
cout << "Saving " << file << endl;
SndfileHandle ofile = SndfileHandle(file, SFM_WRITE, SF_FORMAT_WAV | SF_FORMAT_PCM_16, 1, 8000);
ofile.write(cropped.audio, cropped.length);
}
//--------------------------------------------------------------
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;
}
void AudioManager::saveSamples () {
SF_INFO sfinfo;
sfinfo.channels = 1;
sfinfo.samplerate = FS;
sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
for (vector<Sample>::iterator it = samples.begin(); it != samples.end(); ++it) {
string file = "output/" + to_string(it->offset) + "-" + it->station + ".wav";
cout << "Saving " << file << endl;
SndfileHandle ofile = SndfileHandle(file, SFM_WRITE, SF_FORMAT_WAV | SF_FORMAT_PCM_16, 1, 8000);
ofile.write(it->audio, it->length);
//sf_write_sync(ofile);
//sf_close(ofile);
}
}
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;
}
/*
* Method read input data from file
*/
int* read_file(string file_name, int *frames_length) {
//file handler
SndfileHandle inputFile;
//read file from input WAV file
inputFile = SndfileHandle(file_name);
//get number of frames
int frames = inputFile.frames();
//set input buffer
int *buffer;
buffer = new int[frames];
//read data from input file
inputFile.read(buffer, frames);
*frames_length = frames;
return buffer;
}
//--------------------------------------------------------------
void testApp::guiEvent(nativeWidget & widget){
ofDisableDataPath();
ofEnableDataPath();
if (widget.name == "newColor"){
float hue = ofRandom(0,255);
float sat = ofRandom(190,230);
float bri = ofRandom(220,238);
color.setHsb(hue, sat, bri);
}
if (widget.name == "repeatSound"){
if (audioSamples.size() > 0){
counter = 0;
bPlaying = true;
}
}
if (widget.name == "textBox" || widget.name == "textBox2" || widget.name == "textBox3"
){
string time = ofGetTimestampString();
string fileName = time + ".aiff";
string fileNameMp3 = time + ".mp3";
string toSay = *((string *)widget.variablePtr);
string command = "say -o " + ofToDataPath(fileName) + " " + "\"" + toSay + "\"" + " --data-format=BEI32@44100";
// big endian int 32 bit samples 44100 sample rate
system(command.c_str());
ofSleepMillis(100); // sometimes really long files need time to save out.
SndfileHandle myf = SndfileHandle(ofToDataPath(fileName).c_str());
float * data = new float[int(myf.frames())];
myf.read (data, int(myf.frames()));
audioSamples.clear();
audioSamples.reserve(int(myf.frames()));
for (int i = 0; i < int(myf.frames()); i++){
audioSamples.push_back(data[i]);
}
delete [] data;
bPlaying = true;
counter = 0;
}
computeMessageColors();
}
/**
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 ;
}
int main(int argc, char** argv) {
if (argc != 2) {
printHelp();
return EXIT_FAILURE;
}
SndfileHandle inputFile;
inputFile = SndfileHandle(argv[1]);
int framesCount = inputFile.frames();
int *buffer;
buffer = new int[framesCount];
inputFile.read(buffer, inputFile.frames());
#ifdef DEBUG
cerr<<"Samples size: "<<framesCount<<endl;
#endif
double firstAngle;
double deltaAngle;
string decoded = "";
/* finding angle of a sin between two samples in the input signal */
firstAngle = asin(((double)buffer[0]) / AMPLITUDE);
deltaAngle = asin(((double)buffer[1]) / AMPLITUDE) - firstAngle;
#ifdef DEBUG
cerr<<"deltaAngle: "<<deltaAngle<<" = "<<deltaAngle*180/PI<<"°"<<endl;
#endif
double actAngle = firstAngle;
double deltaX1 = deltaAngle;
double deltaX4 = deltaAngle;
/* QPSK bauds mapping values */
double expectedAngle1 = addAngle(0, 3*PI/4.0);
double expectedAngle2 = addAngle(0, PI/4.0);
double expectedAngle3 = addAngle(0, 5*PI/4.0);
double expectedAngle4 = addAngle(0, 7*PI/4.0);
int changes = 0;
int firstChange = 0;
int secondChange = 0;
int thirdChange = 0;
double expectedAngle1b = addNormalize(0, expectedAngle1);
double expectedAngle4b = addNormalize(0, expectedAngle4);
double *expectedAngle = &expectedAngle1b;
/* while 3 changes in sync part of the signal signal */
while (changes < 3) {
#ifdef DEBUG
cerr<<"loop: "<<thirdChange<<endl;
cerr<<"actAngle: "<<actAngle<<" = "<<actAngle*180/PI<<"° "<<sin(actAngle)<<endl;
cerr<<"expected: "<<*expectedAngle<<" = "<<*expectedAngle*180/PI<<"° "<<sin(*expectedAngle)<<endl;
#endif
/* we have a change in a baud */
if (!((actAngle > *expectedAngle - (PI/20))&&(actAngle < *expectedAngle + (PI/20)))) {
if (changes == 0) {
expectedAngle = &expectedAngle4b;
} else if (changes == 1) {
expectedAngle = &expectedAngle1b;
}
changes++;
}
/* counting samples till first change */
if (changes < 1) {
firstChange++;
}
/* counting samples till second change */
if (changes < 2) {
secondChange++;
}
/* counting samples till third change */
if (changes < 3) {
thirdChange++;
}
/* thirdChange is also a counter for this loop */
actAngle = asin(((double)buffer[thirdChange]) / AMPLITUDE);
expectedAngle1b = addNormalizeChangeDelta(expectedAngle1b, &deltaX1);
expectedAngle4b = addNormalizeChangeDelta(expectedAngle4b, &deltaX4);
}
/* preparing values for voting */
secondChange /= 2;
thirdChange /=3;
int samplesPerBaud = 0;
/* voting for samles per baud... we should have at least 2 same
* values to claim it as a samples per baud */
if (firstChange == secondChange) {
samplesPerBaud = firstChange;
} else if (secondChange == thirdChange) {
samplesPerBaud = secondChange;
} else if (firstChange == thirdChange) {
samplesPerBaud = firstChange;
}
#ifdef DEBUG
cerr<<"SPB: "<<samplesPerBaud<< " 1st: "<<firstChange<<" 2nd: "<<secondChange<<" 3rd: "<<thirdChange<<endl;
#endif
double res1 = 0;
double res2 = 0;
double res3 = 0;
double res4 = 0;
/* Iteraes through the all samples. For each baud we are looking
* for the sin values that differs the least from actual baud */
for (int i = 0, s = samplesPerBaud; i < framesCount; i++, s--) {
/* counting differences for each sin */
res1 += fabs(buffer[i] - sin(expectedAngle1));
res2 += fabs(buffer[i] - sin(expectedAngle2));
res3 += fabs(buffer[i] - sin(expectedAngle3));
res4 += fabs(buffer[i] - sin(expectedAngle4));
/* we have processed a baud and we are looking for the
* least change */
if (s == 0 || i == framesCount-1) {
s = samplesPerBaud;
double res12;
double res34;
int resId12 = 0;
int resId34 = 0;
if (res1 < res2) {
res12 = res1;
resId12 = 1;
} else {
res12 = res2;
resId12 = 2;
}
if (res3 < res4) {
res34 = res3;
resId34 = 3;
} else {
res34 = res4;
resId34 = 4;
}
int resId;
if (res12 < res34) {
resId = resId12;
} else {
resId = resId34;
}
switch(resId) {
case 1: decoded += "00"; break;
case 2: decoded += "01"; break;
case 3: decoded += "10"; break;
case 4: decoded += "11"; break;
default: break;
}
res1 = 0;
res2 = 0;
res3 = 0;
res4 = 0;
}
expectedAngle1 = addAngle(expectedAngle1, -deltaAngle);
expectedAngle2 = addAngle(expectedAngle2, -deltaAngle);
expectedAngle3 = addAngle(expectedAngle3, -deltaAngle);
expectedAngle4 = addAngle(expectedAngle4, -deltaAngle);
}
#ifdef DEBUG
cerr<<decoded<<endl;
cerr<<decoded.substr(8);
#endif
/* writting to an output file */
string outName = string(argv[1]);
outName = outName.replace(outName.end()-3, outName.end(), "txt");
ofstream outFile(outName);
if (!outFile.is_open()) {
cerr<<"Can not open output file\n";
return EXIT_FAILURE;
}
outFile<<decoded.substr(8);
outFile.close();
delete [] buffer;
return EXIT_SUCCESS;
}
