// 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;
}
예제 #2
0
파일: test.cpp 프로젝트: rit-sse/teslacoil
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 */
예제 #3
0
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);
}
예제 #4
0
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;
}
예제 #6
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);

}
예제 #7
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//--------------------------------------------------------------
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;
    
}
예제 #8
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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);
  }
}
예제 #9
0
파일: test.cpp 프로젝트: rit-sse/teslacoil
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 */
예제 #10
0
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;
}
예제 #11
0
파일: bms1B.cpp 프로젝트: xmagam00/BMS
/*
 * 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;
}
예제 #12
0
//--------------------------------------------------------------
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();
}
예제 #13
0
/**
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( &parameters, 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 ;

}
예제 #15
0
파일: bms1B.c 프로젝트: xmagam00/BMS
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;
}