// Note: this function will have half-buffer phase shift effect
void processInput(short *input, short *output, short channel)
{
int i;
// Read in the input buffer
for (i = 0; i < ARRAY_SIZE; i++)
{
inputBuffer[channel][ARRAY_SIZE + i] = input[i];
}
// Apply windows to the midpoint buffer and input buffer
applyWindow(inputBuffer[channel] + (ARRAY_SIZE / 2), windowedCrossSampleArray[channel]);
applyWindow(inputBuffer[channel] + ARRAY_SIZE, windowedInputArray[channel]);
frequencyScale(windowedCrossSampleArray[channel], frequencyScaledSamples[channel][1], ARRAY_SIZE, frequencyScalars);
frequencyScale(windowedInputArray[channel], frequencyScaledSamples[channel][2], ARRAY_SIZE, frequencyScalars);
// Sum overlaps to produce the output array
for (i = 0; i < ARRAY_SIZE / 2; i++)
{
output[i] = frequencyScaledSamples[channel][FSS_PREVIOUS_INPUT][ARRAY_SIZE / 2 + i] + frequencyScaledSamples[channel][FSS_CROSS_INPUT][i];
}
for (i = ARRAY_SIZE / 2; i < ARRAY_SIZE; i++)
{
// float -> short conversion should be handled automatically
output[i] = frequencyScaledSamples[channel][FSS_CROSS_INPUT][i] + frequencyScaledSamples[channel][FSS_THIS_INPUT][i - (ARRAY_SIZE / 2)];
}
// Store input and scaled version of input for use in next iteration
for (i = 0; i < ARRAY_SIZE; i++)
{
inputBuffer[channel][i] = inputBuffer[channel][i + ARRAY_SIZE];
frequencyScaledSamples[channel][FSS_PREVIOUS_INPUT][i] = frequencyScaledSamples[channel][FSS_THIS_INPUT][i];
}
/*
for (i = 0; i < ARRAY_SIZE / 2; i++)
{
printf("%d, %d\n", input[i], output[i + ARRAY_SIZE / 2]);
}
printf("\n\n");
*/
}
void RealtimeAnalyser::doFFTAnalysis() {
DCHECK(isMainThread());
// Unroll the input buffer into a temporary buffer, where we'll apply an
// analysis window followed by an FFT.
size_t fftSize = this->fftSize();
AudioFloatArray temporaryBuffer(fftSize);
float* inputBuffer = m_inputBuffer.data();
float* tempP = temporaryBuffer.data();
// Take the previous fftSize values from the input buffer and copy into the
// temporary buffer.
unsigned writeIndex = m_writeIndex;
if (writeIndex < fftSize) {
memcpy(tempP, inputBuffer + writeIndex - fftSize + InputBufferSize,
sizeof(*tempP) * (fftSize - writeIndex));
memcpy(tempP + fftSize - writeIndex, inputBuffer,
sizeof(*tempP) * writeIndex);
} else {
memcpy(tempP, inputBuffer + writeIndex - fftSize, sizeof(*tempP) * fftSize);
}
// Window the input samples.
applyWindow(tempP, fftSize);
// Do the analysis.
m_analysisFrame->doFFT(tempP);
float* realP = m_analysisFrame->realData();
float* imagP = m_analysisFrame->imagData();
// Blow away the packed nyquist component.
imagP[0] = 0;
// Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT
// scaling factor).
const double magnitudeScale = 1.0 / fftSize;
// A value of 0 does no averaging with the previous result. Larger values
// produce slower, but smoother changes.
double k = m_smoothingTimeConstant;
k = std::max(0.0, k);
k = std::min(1.0, k);
// Convert the analysis data from complex to magnitude and average with the
// previous result.
float* destination = magnitudeBuffer().data();
size_t n = magnitudeBuffer().size();
for (size_t i = 0; i < n; ++i) {
std::complex<double> c(realP[i], imagP[i]);
double scalarMagnitude = abs(c) * magnitudeScale;
destination[i] = float(k * destination[i] + (1 - k) * scalarMagnitude);
}
}
void RealtimeAnalyser::doFFTAnalysis()
{
ASSERT(isMainThread());
// Unroll the input buffer into a temporary buffer, where we'll apply an analysis window followed by an FFT.
size_t fftSize = this->fftSize();
AudioFloatArray temporaryBuffer(fftSize);
float* inputBuffer = m_inputBuffer.data();
float* tempP = temporaryBuffer.data();
// Take the previous fftSize values from the input buffer and copy into the temporary buffer.
// FIXME : optimize with memcpy().
unsigned writeIndex = m_writeIndex;
for (unsigned i = 0; i < fftSize; ++i)
tempP[i] = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
// Window the input samples.
applyWindow(tempP, fftSize);
// Do the analysis.
m_analysisFrame->doFFT(tempP);
size_t n = DefaultFFTSize / 2;
float* realP = m_analysisFrame->realData();
float* imagP = m_analysisFrame->imagData();
// Blow away the packed nyquist component.
imagP[0] = 0.0f;
// Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
const double MagnitudeScale = 1.0 / DefaultFFTSize;
// A value of 0 does no averaging with the previous result. Larger values produce slower, but smoother changes.
double k = m_smoothingTimeConstant;
k = max(0.0, k);
k = min(1.0, k);
// Convert the analysis data from complex to magnitude and average with the previous result.
float* destination = magnitudeBuffer().data();
for (unsigned i = 0; i < n; ++i) {
Complex c(realP[i], imagP[i]);
double scalarMagnitude = abs(c) * MagnitudeScale;
destination[i] = float(k * destination[i] + (1.0 - k) * scalarMagnitude);
}
}
/* -- main function -- */
int main( int argc, char **argv ) {
PaStreamParameters inputParameters;
float a[2], b[3], mem1[4], mem2[4];
float data[FFT_SIZE];
float datai[FFT_SIZE];
float window[FFT_SIZE];
float freqTable[FFT_SIZE];
char * noteNameTable[FFT_SIZE];
float notePitchTable[FFT_SIZE];
void * fft = NULL;
PaStream *stream = NULL;
PaError err = 0;
struct sigaction action;
// add signal listen so we know when to exit:
action.sa_handler = signalHandler;
sigemptyset (&action.sa_mask);
action.sa_flags = 0;
sigaction (SIGINT, &action, NULL);
sigaction (SIGHUP, &action, NULL);
sigaction (SIGTERM, &action, NULL);
// build the window, fft, etc
/*
buildHanWindow( window, 30 );
for( int i=0; i<30; ++i ) {
for( int j=0; j<window[i]*50; ++j )
printf( "*" );
printf("\n");
}
exit(0);
*/
buildHanWindow( window, FFT_SIZE );
fft = initfft( FFT_EXP_SIZE );
computeSecondOrderLowPassParameters( SAMPLE_RATE, 330, a, b );
mem1[0] = 0;
mem1[1] = 0;
mem1[2] = 0;
mem1[3] = 0;
mem2[0] = 0;
mem2[1] = 0;
mem2[2] = 0;
mem2[3] = 0;
//freq/note tables
for( int i=0; i<FFT_SIZE; ++i ) {
freqTable[i] = ( SAMPLE_RATE * i ) / (float) ( FFT_SIZE );
}
for( int i=0; i<FFT_SIZE; ++i ) {
noteNameTable[i] = NULL;
notePitchTable[i] = -1;
}
for( int i=0; i<127; ++i ) {
float pitch = ( 440.0 / 32.0 ) * pow( 2, (i-9.0)/12.0 ) ;
if( pitch > SAMPLE_RATE / 2.0 )
break;
//find the closest frequency using brute force.
float min = 1000000000.0;
int index = -1;
for( int j=0; j<FFT_SIZE; ++j ) {
if( fabsf( freqTable[j]-pitch ) < min ) {
min = fabsf( freqTable[j]-pitch );
index = j;
}
}
noteNameTable[index] = NOTES[i%12];
notePitchTable[index] = pitch;
//printf( "%f %d %s\n", pitch, index, noteNameTable[index] );
}
// initialize portaudio
err = Pa_Initialize();
if( err != paNoError ) goto error;
inputParameters.device = Pa_GetDefaultInputDevice();
inputParameters.channelCount = 1;
inputParameters.sampleFormat = paFloat32;
inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultHighInputLatency ;
inputParameters.hostApiSpecificStreamInfo = NULL;
printf( "Opening %s\n",
Pa_GetDeviceInfo( inputParameters.device )->name );
err = Pa_OpenStream( &stream,
&inputParameters,
NULL, //no output
SAMPLE_RATE,
FFT_SIZE,
paClipOff,
NULL,
NULL );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
// this is the main loop where we listen to and
// process audio.
while( running )
{
// read some data
err = Pa_ReadStream( stream, data, FFT_SIZE );
if( err ) goto error; //FIXME: we don't want to err on xrun
// low-pass
//for( int i=0; i<FFT_SIZE; ++i )
// printf( "in %f\n", data[i] );
for( int j=0; j<FFT_SIZE; ++j ) {
data[j] = processSecondOrderFilter( data[j], mem1, a, b );
data[j] = processSecondOrderFilter( data[j], mem2, a, b );
}
// window
applyWindow( window, data, FFT_SIZE );
// do the fft
for( int j=0; j<FFT_SIZE; ++j )
datai[j] = 0;
applyfft( fft, data, datai, false );
//find the peak
float maxVal = -1;
int maxIndex = -1;
for( int j=0; j<FFT_SIZE/2; ++j ) {
float v = data[j] * data[j] + datai[j] * datai[j] ;
/*
printf( "%d: ", j*SAMPLE_RATE/(2*FFT_SIZE) );
for( int i=0; i<sqrt(v)*100000000; ++i )
printf( "*" );
printf( "\n" );
*/
if( v > maxVal ) {
maxVal = v;
maxIndex = j;
}
}
float freq = freqTable[maxIndex];
//find the nearest note:
int nearestNoteDelta=0;
while( true ) {
if( nearestNoteDelta < maxIndex && noteNameTable[maxIndex-nearestNoteDelta] != NULL ) {
nearestNoteDelta = -nearestNoteDelta;
break;
} else if( nearestNoteDelta + maxIndex < FFT_SIZE && noteNameTable[maxIndex+nearestNoteDelta] != NULL ) {
break;
}
++nearestNoteDelta;
}
char * nearestNoteName = noteNameTable[maxIndex+nearestNoteDelta];
float nearestNotePitch = notePitchTable[maxIndex+nearestNoteDelta];
float centsSharp = 1200 * log( freq / nearestNotePitch ) / log( 2.0 );
// now output the results:
printf("\033[2J\033[1;1H"); //clear screen, go to top left
fflush(stdout);
printf( "Tuner listening. Control-C to exit.\n" );
printf( "%f Hz, %d : %f\n", freq, maxIndex, maxVal*1000 );
printf( "Nearest Note: %s\n", nearestNoteName );
if( nearestNoteDelta != 0 ) {
if( centsSharp > 0 )
printf( "%f cents sharp.\n", centsSharp );
if( centsSharp < 0 )
printf( "%f cents flat.\n", -centsSharp );
} else {
printf( "in tune!\n" );
}
printf( "\n" );
int chars = 30;
if( nearestNoteDelta == 0 || centsSharp >= 0 ) {
for( int i=0; i<chars; ++i )
printf( " " );
} else {
for( int i=0; i<chars+centsSharp; ++i )
printf( " " );
for( int i=chars+centsSharp<0?0:chars+centsSharp; i<chars; ++i )
printf( "=" );
}
printf( " %2s ", nearestNoteName );
if( nearestNoteDelta != 0 )
for( int i=0; i<chars && i<centsSharp; ++i )
printf( "=" );
printf("\n");
}
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
// cleanup
destroyfft( fft );
Pa_Terminate();
return 0;
error:
if( stream ) {
Pa_AbortStream( stream );
Pa_CloseStream( stream );
}
destroyfft( fft );
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return 1;
}
