Searching around on how I can improve my waveform generation code, I've come across SIMD and the libsimdpp library, but I have no idea how to use it. If I got it right using raw SIMD will require me to write code for each architecture while libsimdpp will handle that for me.
What I need to do is, to calculate the squared and rms value of a chunk of samples, which I managed to boost the process using vectorization, which worked perfectly until I introduced the same calculation for both left and right channel of an audio file.
So, my question and what I need help with, is how can I use libsimdpp (or any library that will make simdp easier for me) to improve the bellow code?
// STRAT: vector containing all the audio samples
std::vector<double> samples;
nv_samples = samples.size();
// END
// START: Loop through the samples vector, incrementing ecah time the index with samples_per_pixel
for (int i = 0; i < nb_samples; i+= samples_per_pixel)
{
// START: Create chunk of samples with the size of samples_per_pixel
double* chunk = &samplesL[i];
// END
// START: Calculate rms and sqared sum
float sum = 0;
float squaredsum = 0;
/// there are multiple definitions of above for both channels but I won't include them
//// to make the code easier to be read
for (int j = 0; j < samples_per_pixel; j++)
{
if (chunk[j] < 0)
sum += -chunk[j]
else
sum += chunk[j]
squaredsum += chunk[j] * chunk[j]
}
/// average
float average_point = (sumL * 2) / samples_per_pixel;
// rms
float meanL = squaredsumL / samples_per_pixel;
rms_pointL = qSqrt(meanL);
/// Drawing of both avearge point and rms
//// [...]
// END
}
Related
I am developing a molecular dynamics simulation code in C++, which essentially takes atom positions and other properties as input and simulates their motion under Newton's laws of motion. The core algorithm uses what's called the Velocity Verlet scheme and looks like:
// iterate through time (k=[1,#steps])
double Dt = 0.002; // time step
double Ttot = 1.0; // total time
double halfDt = Dt/2.0;
for (int k = 1; k*Dt <= Ttot; k++){
for (int i = 0; i < number_particles; i++)
vHalf[i] = p[i].velocity + F[i]*halfDt; // step 1
for (int i = 0; i < number_particles; i++)
p[i].position += vHalf[i]*Dt; // step 2
for (int i = 0; i < number_particles; i++)
F[i] = Force(p,i); // recalculate force on all particle i's
for (int i = 0; i < number_particles; i++)
p[i].velocity = vHalf[i] + F[i]*halfDt; // step 3
}
Where p is an array of class objects which store things like particle position, velocity, mass, etc. and Force is a function that calculates the net force on a particle using something like Lennard-Jones potential.
My question regards the time required to complete the calculation; all of my subroutines are optimized in terms of crunching numbers (e.g. using x*x*x to raise to the third power instead of pow(x,3)), but the main issue is the time loop will often be performed for millions of iterations and there are typically close to a million particles. Is there any way to implement this algorithm using multi-threading? From my understanding, multi-threading essentially opens another stream of data to and from a CPU core, which would allow me to run two different simulations at the same time; I would like to use multi-threading to make just one of these simulations run faster
I'd recommend using OpenMP.
Your specific use case is trivially parallelizable.
Prallelization should be as simple as:
double Dt = 0.002; // time step
double Ttot = 1.0; // total time
double halfDt = Dt/2.0;
for (int k = 1; k*Dt <= Ttot; k++){
#pragma omp parallel for
for (int i = 0; i < number_particles; i++)
vHalf[i] = p[i].velocity + F[i]*halfDt; // step 1
p[i].position += vHalf[i]*Dt; // step 2
#pragma omp parallel for
for (int i = 0; i < number_particles; i++)
F[i] = Force(p,i); // recalculate force on all particle i's
p[i].velocity = vHalf[i] + F[i]*halfDt; // step 3
}
Most popular compilers and platforms have support for OpenMP.
I'm trying to use C++ to recreate the spectrogram function used by Matlab. The function uses a Short Time Fourier Transform (STFT). I found some C++ code here that performs a STFT. The code seems to work perfectly for all frequencies but I only want a few. I found this post for a similar question with the following answer:
Just take the inner product of your data with a complex exponential at
the frequency of interest. If g is your data, then just substitute for
f the value of the frequency you want (e.g., 1, 3, 10, ...)
Having no background in mathematics, I can't figure out how to do this. The inner product part seems simple enough from the Wikipedia page but I have absolutely no idea what he means by (with regard to the formula for a DFT)
a complex exponential at frequency of interest
Could someone explain how I might be able to do this? My data structure after the STFT is a matrix filled with complex numbers. I just don't know how to extract my desired frequencies.
Relevant function, where window is Hamming, and vector of desired frequencies isn't yet an input because I don't know what to do with them:
Matrix<complex<double>> ShortTimeFourierTransform::Calculate(const vector<double> &signal,
const vector<double> &window, int windowSize, int hopSize)
{
int signalLength = signal.size();
int nOverlap = hopSize;
int cols = (signal.size() - nOverlap) / (windowSize - nOverlap);
Matrix<complex<double>> results(window.size(), cols);
int chunkPosition = 0;
int readIndex;
// Should we stop reading in chunks?
bool shouldStop = false;
int numChunksCompleted = 0;
int i;
// Process each chunk of the signal
while (chunkPosition < signalLength && !shouldStop)
{
// Copy the chunk into our buffer
for (i = 0; i < windowSize; i++)
{
readIndex = chunkPosition + i;
if (readIndex < signalLength)
{
// Note the windowing!
data[i][0] = signal[readIndex] * window[i];
data[i][1] = 0.0;
}
else
{
// we have read beyond the signal, so zero-pad it!
data[i][0] = 0.0;
data[i][1] = 0.0;
shouldStop = true;
}
}
// Perform the FFT on our chunk
fftw_execute(plan_forward);
// Copy the first (windowSize/2 + 1) data points into your spectrogram.
// We do this because the FFT output is mirrored about the nyquist
// frequency, so the second half of the data is redundant. This is how
// Matlab's spectrogram routine works.
for (i = 0; i < windowSize / 2 + 1; i++)
{
double real = fft_result[i][0];
double imaginary = fft_result[i][1];
results(i, numChunksCompleted) = complex<double>(real, imaginary);
}
chunkPosition += hopSize;
numChunksCompleted++;
} // Excuse the formatting, the while ends here.
return results;
}
Look up the Goertzel algorithm or filter for example code that uses the computational equivalent of an inner product against a complex exponential to measure the presence or magnitude of a specific stationary sinusoidal frequency in a signal. Performance or resolution will depend on the length of the filter and your signal.
I wrote a program that loads, saves, and performs the fft and ifft on black and white png images. After much debugging headache, I finally got some coherent output only to find that it distorted the original image.
input:
fft:
ifft:
As far as I have tested, the pixel data in each array is stored and converted correctly. Pixels are stored in two arrays, 'data' which contains the b/w value of each pixel and 'complex_data' which is twice as long as 'data' and stores real b/w value and imaginary parts of each pixel in alternating indices. My fft algorithm operates on an array structured like 'complex_data'. After code to read commands from the user, here's the code in question:
if (cmd == "fft")
{
if (height > width) size = height;
else size = width;
N = (int)pow(2.0, ceil(log((double)size)/log(2.0)));
temp_data = (double*) malloc(sizeof(double) * width * 2); //array to hold each row of the image for processing in FFT()
for (i = 0; i < (int) height; i++)
{
for (j = 0; j < (int) width; j++)
{
temp_data[j*2] = complex_data[(i*width*2)+(j*2)];
temp_data[j*2+1] = complex_data[(i*width*2)+(j*2)+1];
}
FFT(temp_data, N, 1);
for (j = 0; j < (int) width; j++)
{
complex_data[(i*width*2)+(j*2)] = temp_data[j*2];
complex_data[(i*width*2)+(j*2)+1] = temp_data[j*2+1];
}
}
transpose(complex_data, width, height); //tested
free(temp_data);
temp_data = (double*) malloc(sizeof(double) * height * 2);
for (i = 0; i < (int) width; i++)
{
for (j = 0; j < (int) height; j++)
{
temp_data[j*2] = complex_data[(i*height*2)+(j*2)];
temp_data[j*2+1] = complex_data[(i*height*2)+(j*2)+1];
}
FFT(temp_data, N, 1);
for (j = 0; j < (int) height; j++)
{
complex_data[(i*height*2)+(j*2)] = temp_data[j*2];
complex_data[(i*height*2)+(j*2)+1] = temp_data[j*2+1];
}
}
transpose(complex_data, height, width);
free(temp_data);
free(data);
data = complex_to_real(complex_data, image.size()/4); //tested
image = bw_data_to_vector(data, image.size()/4); //tested
cout << "*** fft success ***" << endl << endl;
void FFT(double* data, unsigned long nn, int f_or_b){ // f_or_b is 1 for fft, -1 for ifft
unsigned long n, mmax, m, j, istep, i;
double wtemp, w_real, wp_real, wp_imaginary, w_imaginary, theta;
double temp_real, temp_imaginary;
// reverse-binary reindexing to separate even and odd indices
// and to allow us to compute the FFT in place
n = nn<<1;
j = 1;
for (i = 1; i < n; i += 2) {
if (j > i) {
swap(data[j-1], data[i-1]);
swap(data[j], data[i]);
}
m = nn;
while (m >= 2 && j > m) {
j -= m;
m >>= 1;
}
j += m;
};
// here begins the Danielson-Lanczos section
mmax = 2;
while (n > mmax) {
istep = mmax<<1;
theta = f_or_b * (2 * M_PI/mmax);
wtemp = sin(0.5 * theta);
wp_real = -2.0 * wtemp * wtemp;
wp_imaginary = sin(theta);
w_real = 1.0;
w_imaginary = 0.0;
for (m = 1; m < mmax; m += 2) {
for (i = m; i <= n; i += istep) {
j = i + mmax;
temp_real = w_real * data[j-1] - w_imaginary * data[j];
temp_imaginary = w_real * data[j] + w_imaginary * data[j-1];
data[j-1] = data[i-1] - temp_real;
data[j] = data[i] - temp_imaginary;
data[i-1] += temp_real;
data[i] += temp_imaginary;
}
wtemp = w_real;
w_real += w_real * wp_real - w_imaginary * wp_imaginary;
w_imaginary += w_imaginary * wp_real + wtemp * wp_imaginary;
}
mmax=istep;
}}
My ifft is the same only with the f_or_b set to -1 instead of 1. My program calls FFT() on each row, transposes the image, calls FFT() on each row again, then transposes back. Is there maybe an error with my indexing?
Not an actual answer as this question is Debug only so some hints instead:
your results are really bad
it should look like this:
first line is the actual DFFT result
Re,Im,Power is amplified by a constant otherwise you would see a black image
the last image is IDFFT of the original not amplified Re,IM result
the second line is the same but the DFFT result is wrapped by half size of image in booth x,y to match the common results in most DIP/CV texts
As you can see if you IDFFT back the wrapped results the result is not correct (checker board mask)
You have just single image as DFFT result
is it power spectrum?
or you forget to include imaginary part? to view only or perhaps also to computation somewhere as well?
is your 1D **DFFT working?**
for real data the result should be symmetric
check the links from my comment and compare the results for some sample 1D array
debug/repair your 1D FFT first and only then move to the next level
do not forget to test Real and complex data ...
your IDFFT looks BW (no gray) saturated
so did you amplify the DFFT results to see the image and used that for IDFFT instead of the original DFFT result?
also check if you do not round to integers somewhere along the computation
beware of (I)DFFT overflows/underflows
If your image pixel intensities are big and the resolution of image too then your computation could loss precision. Newer saw this in images but if your image is HDR then it is possible. This is a common problem with convolution computed by DFFT for big polynomials.
Thank you everyone for your opinions. All that stuff about memory corruption, while it makes a point, is not the root of the problem. The sizes of data I'm mallocing are not overly large, and I am freeing them in the right places. I had a lot of practice with this while learning c. The problem was not the fft algorithm either, nor even my 2D implementation of it.
All I missed was the scaling by 1/(M*N) at the very end of my ifft code. Because the image is 512x512, I needed to scale my ifft output by 1/(512*512). Also, my fft looks like white noise because the pixel data was not rescaled to fit between 0 and 255.
Suggest you look at the article http://www.yolinux.com/TUTORIALS/C++MemoryCorruptionAndMemoryLeaks.html
Christophe has a good point but he is wrong about it not being related to the problem because it seems that in modern times using malloc instead of new()/free() does not initialise memory or select best data type which would result in all problems listed below:-
Possibly causes are:
Sign of a number changing somewhere, I have seen similar issues when a platform invoke has been used on a dll and a value is passed by value instead of reference. It is caused by memory not necessarily being empty so when your image data enters it will have boolean maths performed on its values. I would suggest that you make sure memory is empty before you put your image data there.
Memory rotating right (ROR in assembly langauge) or left (ROL) . This will occur if data types are being used which do not necessarily match, eg. a signed value entering an unsigned data type or if the number of bits is different in one variable to another.
Data being lost due to an unsigned value entering a signed variable. Outcomes are 1 bit being lost because it will be used to determine negative or positive, or at extremes if twos complement takes place the number will become inverted in meaning, look for twos complement on wikipedia.
Also see how memory should be cleared/assigned before use. http://www.cprogramming.com/tutorial/memory_debugging_parallel_inspector.html
For a project I need to be able to generate a spectrogram from a .WAV file. I've read the following should be done:
Get N (transform size) samples
Apply a window function
Do a Fast Fourier Transform using the samples
Normalise the output
Generate spectrogram
On the image below you see two spectrograms of a 10000 Hz sine wave both using the hanning window function. On the left you see a spectrogram generated by audacity and on the right my version. As you can see my version has a lot more lines/noise. Is this leakage in different bins? How would I get a clear image like the one audacity generates. Should I do some post-processing? I have not yet done any normalisation because do not fully understand how to do so.
update
I found this tutorial explaining how to generate a spectrogram in c++. I compiled the source to see what differences I could find.
My math is very rusty to be honest so I'm not sure what the normalisation does here:
for(i = 0; i < half; i++){
out[i][0] *= (2./transform_size);
out[i][6] *= (2./transform_size);
processed[i] = out[i][0]*out[i][0] + out[i][7]*out[i][8];
//sets values between 0 and 1?
processed[i] =10. * (log (processed[i] + 1e-6)/log(10)) /-60.;
}
after doing this I got this image (btw I've inverted the colors):
I then took a look at difference of the input samples provided by my sound library and the one of the tutorial. Mine were way higher so I manually normalised is by dividing it by the factor 32767.9. I then go this image which looks pretty ok I think. But dividing it by this number seems wrong. And I would like to see a different solution.
Here is the full relevant source code.
void Spectrogram::process(){
int i;
int transform_size = 1024;
int half = transform_size/2;
int step_size = transform_size/2;
double in[transform_size];
double processed[half];
fftw_complex *out;
fftw_plan p;
out = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * transform_size);
for(int x=0; x < wavFile->getSamples()/step_size; x++){
int j = 0;
for(i = step_size*x; i < (x * step_size) + transform_size - 1; i++, j++){
in[j] = wavFile->getSample(i)/32767.9;
}
//apply window function
for(i = 0; i < transform_size; i++){
in[i] *= windowHanning(i, transform_size);
// in[i] *= windowBlackmanHarris(i, transform_size);
}
p = fftw_plan_dft_r2c_1d(transform_size, in, out, FFTW_ESTIMATE);
fftw_execute(p); /* repeat as needed */
for(i = 0; i < half; i++){
out[i][0] *= (2./transform_size);
out[i][11] *= (2./transform_size);
processed[i] = out[i][0]*out[i][0] + out[i][12]*out[i][13];
processed[i] =10. * (log (processed[i] + 1e-6)/log(10)) /-60.;
}
for (i = 0; i < half; i++){
if(processed[i] > 0.99)
processed[i] = 1;
In->setPixel(x,(half-1)-i,processed[i]*255);
}
}
fftw_destroy_plan(p);
fftw_free(out);
}
This is not exactly an answer as to what is wrong but rather a step by step procedure to debug this.
What do you think this line does? processed[i] = out[i][0]*out[i][0] + out[i][12]*out[i][13] Likely that is incorrect: fftw_complex is typedef double fftw_complex[2], so you only have out[i][0] and out[i][1], where the first is the real and the second the imaginary part of the result for that bin. If the array is contiguous in memory (which it is), then out[i][12] is likely the same as out[i+6][0] and so forth. Some of these will go past the end of the array, adding random values.
Is your window function correct? Print out windowHanning(i, transform_size) for every i and compare with a reference version (for example numpy.hanning or the matlab equivalent). This is the most likely cause, what you see looks like a bad window function, kind of.
Print out processed, and compare with a reference version (given the same input, of course you'd have to print the input and reformat it to feed into pylab/matlab etc). However, the -60 and 1e-6 are fudge factors which you don't want, the same effect is better done in a different way. Calculate like this:
power_in_db[i] = 10 * log(out[i][0]*out[i][0] + out[i][1]*out[i][1])/log(10)
Print out the values of power_in_db[i] for the same i but for all x (a horizontal line). Are they approximately the same?
If everything so far is good, the remaining suspect is setting the pixel values. Be very explicit about clipping to range, scaling and rounding.
int pixel_value = (int)round( 255 * (power_in_db[i] - min_db) / (max_db - min_db) );
if (pixel_value < 0) { pixel_value = 0; }
if (pixel_value > 255) { pixel_value = 255; }
Here, again, print out the values in a horizontal line, and compare with the grayscale values in your pgm (by hand, using the colorpicker in photoshop or gimp or similar).
At this point, you will have validated everything from end to end, and likely found the bug.
The code you produced, was almost correct. So, you didn't left me much to correct:
void Spectrogram::process(){
int transform_size = 1024;
int half = transform_size/2;
int step_size = transform_size/2;
double in[transform_size];
double processed[half];
fftw_complex *out;
fftw_plan p;
out = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * transform_size);
for (int x=0; x < wavFile->getSamples()/step_size; x++) {
// Fill the transformation array with a sample frame and apply the window function.
// Normalization is performed later
// (One error was here: you didn't set the last value of the array in)
for (int j = 0, int i = x * step_size; i < x * step_size + transform_size; i++, j++)
in[j] = wavFile->getSample(i) * windowHanning(j, transform_size);
p = fftw_plan_dft_r2c_1d(transform_size, in, out, FFTW_ESTIMATE);
fftw_execute(p); /* repeat as needed */
for (int i=0; i < half; i++) {
// (Here were some flaws concerning the access of the complex values)
out[i][0] *= (2./transform_size); // real values
out[i][1] *= (2./transform_size); // complex values
processed[i] = out[i][0]*out[i][0] + out[i][1]*out[i][1]; // power spectrum
processed[i] = 10./log(10.) * log(processed[i] + 1e-6); // dB
// The resulting spectral values in 'processed' are in dB and related to a maximum
// value of about 96dB. Normalization to a value range between 0 and 1 can be done
// in several ways. I would suggest to set values below 0dB to 0dB and divide by 96dB:
// Transform all dB values to a range between 0 and 1:
if (processed[i] <= 0) {
processed[i] = 0;
} else {
processed[i] /= 96.; // Reduce the divisor if you prefer darker peaks
if (processed[i] > 1)
processed[i] = 1;
}
In->setPixel(x,(half-1)-i,processed[i]*255);
}
// This should be called each time fftw_plan_dft_r2c_1d()
// was called to avoid a memory leak:
fftw_destroy_plan(p);
}
fftw_free(out);
}
The two corrected bugs were most probably responsible for the slight variation of successive transformation results. The Hanning window is very vell suited to minimize the "noise" so a different window would not have solved the problem (actually #Alex I already pointed to the 2nd bug in his point 2. But in his point 3. he added a -Inf-bug as log(0) is not defined which can happen if your wave file containts a stretch of exact 0-values. To avoid this the constant 1e-6 is good enough).
Not asked, but there are some optimizations:
put p = fftw_plan_dft_r2c_1d(transform_size, in, out, FFTW_ESTIMATE); outside the main loop,
precalculate the window function outside the main loop,
abandon the array processed and just use a temporary variable to hold one spectral line at a time,
the two multiplications of out[i][0] and out[i][1] can be abandoned in favour of one multiplication with a constant in the following line. I left this (and other things) for you to improve
Thanks to #Maxime Coorevits additionally a memory leak could be avoided: "Each time you call fftw_plan_dft_rc2_1d() memory are allocated by FFTW3. In your code, you only call fftw_destroy_plan() outside the outer loop. But in fact, you need to call this each time you request a plan."
Audacity typically doesn't map one frequency bin to one horizontal line, nor one sample period to one vertical line. The visual effect in Audacity may be due to resampling of the spectrogram picture in order to fit the drawing area.
I have a method which calculates an integral image (description here) commonly used in computer vision applications.
float *Integral(unsigned char *grayscaleSource, int height, int width, int widthStep)
{
// convert the image to single channel 32f
unsigned char *img = grayscaleSource;
// set up variables for data access
int step = widthStep/sizeof(float);
uint8_t *data = (uint8_t *)img;
float *i_data = (float *)malloc(height * width * sizeof(float));
// first row only
float rs = 0.0f;
for(int j=0; j<width; j++)
{
rs += (float)data[j];
i_data[j] = rs;
}
// remaining cells are sum above and to the left
for(int i=1; i<height; ++i)
{
rs = 0.0f;
for(int j=0; j<width; ++j)
{
rs += data[i*step+j];
i_data[i*step+j] = rs + i_data[(i-1)*step+j];
}
}
// return the integral image
return i_data;
}
I am trying to make it as fast as possible. It seems to me like this should be able to take advantage of Apple's Accelerate.framework, or perhaps ARMs neon intrinsics, but I can't see exactly how. It seems like that nested loop is potentially quite slow (for real time applications at least).
Does anyone think this is possible to speed up using any other techniques??
You can certainly vectorize the row by row summation. That is vDSP_vadd(). The horizontal direction is vDSP_vrsum().
If you want to write your own vector code, the horizontal sum might be sped up by something like psadbw, but that is Intel. Also, take a look at prefix sum algorithms, which are famously parallelizable.