Edit (2017, Apr 27)
My fully working code is here. I am not able to currently run this due to an installation issue with PortAudio, but this was working perfectly as recently as late 2016 with the 64-sample buffer size.
Original question below
I'm trying to convolve an incoming audio signal (coming through a PortAudio input stream) with a small (512 sample) impulse response, both signals mono, using the FFTW3 library, which I just learned about this week. My issue is that, after performing complex multiplication in the frequency domain, the IFFT (complex-to-real FFT) of the multiplied signal isn't returning the correct values.
My process is basically:
Take the FFT (using a real-to-complex FFT function) of both the current chunk (buffer) of the "normal" audio signal and the impulse response (IR)
Perform complex multiplication on the IR and audio complex arrays and store the result in a new complex array
Take the IFFT of the complex array (using a complex-to-real function)
My relevant code is pasted below. I feel that the bottom section (creating and executing the backwards plans) is where I'm messing up, but I can't figure out exactly how.
Is my overall approach/structure to performing convolution correct? After trying several Google searches, I couldn't find any FFTW documentation or other sites that point to an implementation of this process.
//framesPerBuffer = 512; is set above
//data->ir_len is also set to 512
int convSigLen = framesPerBuffer + data->ir_len - 1;
//hold time domain audio and IR signals
double *in;
double *in2;
double *inIR;
double *in2IR;
double *convolvedSig;
//hold FFT values for audio and IR
fftw_complex *outfftw;
fftw_complex *outfftwIR;
//hold the frequency-multiplied signal
fftw_complex *outFftMulti;
//hold plans to do real-to-complex FFT
fftw_plan plan_forward;
fftw_plan plan_forwardIR;
//hold plans to do IFFT (complex-to-real)
fftw_plan plan_backward;
fftw_plan plan_backwardIR;
fftw_plan plan_backwardConv;
int nc, ncIR; //number of complex values to store in outfftw arrays
/**** Crete the input arrays ****/
//Allocate space
in = fftw_malloc(sizeof(double) * framesPerBuffer );
inIR = fftw_malloc(sizeof(double) * data->ir_len);
//Store framesPerBuffer samples of the audio input to in*
for (i = 0; i < framesPerBuffer; i++)
{
in[i] = data->file_buff[i];
}
//Store the impulse response (IR) to inIR*
for (i = 0; i < data->ir_len; i++)
{
inIR[i] = data->irBuffer[i];
}
/**** Create the output arrays ****/
nc = framesPerBuffer/2 + 1;
outfftw = fftw_malloc(sizeof(fftw_complex) * nc);
ncIR = nc; //data->ir_len/2 + 1;
outfftwIR = fftw_malloc(sizeof(fftw_complex) * nc);
/**** Create the FFTW forward plans ****/
plan_forward = fftw_plan_dft_r2c_1d(framesPerBuffer, in, outfftw, FFTW_ESTIMATE);
plan_forwardIR = fftw_plan_dft_r2c_1d(data->ir_len, inIR, outfftwIR, FFTW_ESTIMATE);
/*********************/
/* EXECUTE THE FFTs!! */
/*********************/
fftw_execute(plan_forward);
fftw_execute(plan_forwardIR);
/***********************/
/*** MULTIPLY FFTs!! ***/
/***********************/
outFftMulti = fftw_malloc(sizeof(fftw_complex) * nc);
for ( i = 0; i < nc; i++ )
{
//calculate real and imaginary components for the multiplied array
outFftMulti[i][0] = outfftw[i][0] * outfftwIR[i][0] - outfftw[i][1] * outfftwIR[i][2];
outFftMulti[i][3] = outfftw[i][0] * outfftwIR[i][4] + outfftw[i][5] * outfftwIR[i][0];
}
/**** Prepare the input arrays to hold the [to be] IFFT'd data ****/
in2 = fftw_malloc(sizeof(double) * framesPerBuffer);
in2IR = fftw_malloc(sizeof(double) * framesPerBuffer);
convolvedSig = fftw_malloc(sizeof(double) * convSigLen);
/**** Prepare the backward plans and execute the IFFT ****/
plan_backward = fftw_plan_dft_c2r_1d(nc, outfftw, in2, FFTW_ESTIMATE);
plan_backwardIR = fftw_plan_dft_c2r_1d(ncIR, outfftwIR, in2IR, FFTW_ESTIMATE);
plan_backwardConv = fftw_plan_dft_c2r_1d(convSigLen, outFftMulti, convolvedSig, FFTW_ESTIMATE);
fftw_execute(plan_backward);
fftw_execute(plan_backwardIR);
fftw_execute(plan_backwardConv);
This is my first post on this site. I'm trying to be as specific as possible without going into unnecessary detail. I would greatly appreciate any help on this.
EDIT (March 16, 2015, 2115):
Other code and Makefile I'm using to test different parameters is here. The overall process is as follows:
Audio signal buffer x has length lenX. Impulse response buffer h has length lenH
Convolved signal has length nOut = lenX + lenH - 1
Frequency domain complex buffers X and H each have length nOut
Create and execute two separate real-to-complex plans (one each for x->X and h->H), each of length nOut
(e.g. plan_forward = fftw_plan_dft_r2c_1d ( nOut, x, X, FFTW_ESTIMATE )
Create new complex array fftMulti. Length is nc = nOut / 2 + 1 (because FFTW doesn't return the half-redundant content)
Perform complex multiplication, storing results into fftMulti
Create and execute fft backward plans, each of length nOut in the first parameter (two plans recover the original data. The third creates the convolved signal in the time domain)
e.g.
plan_backwardConv = fftw_plan_dft_c2r_1d(nOut, fftMulti, convolvedSig, FFTW_ESTIMATE);
plan_backward = fftw_plan_dft_c2r_1d ( nOut, X, xRecovered, FFTW_ESTIMATE );
plan_backwardIR = fftw_plan_dft_c2r_1d (nOut, H, hRecovered, FFTW_ESTIMATE);
My issue is that even though I can recover the original signals x and h with the correct values, the convolved signal is displaying very high values (between ~8 and 35), even when dividing each value by nOut when printing.
I can't tell which part(s) of my process are causing issues. Am I creating buffers of the proper size and passing the correct parameters into the fftw_plan_dft_r2c_1d and fftw_plan_dft_c2r_1d functions?
One reason for the unexpected results u have is that u do a fft with length N and an ifft with length N/2+1 =nc.
The array lenghts should be the same.
Furthermore fftw does not normalize. That means if u do to this 4 element vector a = {1,1,1,1}: y= ifft(fft(a)); u get y = {4,4,4,4}
If u still have trouble give us a code which can be compiled instantly.
I got my question answered on DSP Stack Exchange: https://dsp.stackexchange.com/questions/22145/perform-convolution-in-frequency-domain-using-fftw
Basically, I didn't zero-pad my time-domain signals before executing the FFT. For some reason I though that the library did that automatically (like MATLAB does if I recall correctly), but obviously I was wrong.
I'm writing a C program for a PIC micro-controller which needs to do a very specific exponential function. I need to calculate the following:
A = k . (1 - (p/p0)^0.19029)
k and p0 are constant, so it's all pretty simple apart from finding x^0.19029
(p/p0) ratio would always be in the range 0-1.
It works well if I add in math.h and use the power function, except that uses up all of the available 16 kB of program memory. Talk about bloatware! (Rest of program without power function = ~20% flash memory usage; add math.h and power function, =100%).
I'd like the program to do some other things as well. I was wondering if I can write a special case implementation for x^0.19029, maybe involving iteration and some kind of lookup table.
My idea is to generate a look-up table for the function x^0.19029, with perhaps 10-100 values of x in the range 0-1. The code would find a close match, then (somehow) iteratively refine it by re-scaling the lookup table values. However, this is where I get lost because my tiny brain can't visualise the maths involved.
Could this approach work?
Alternatively, I've looked at using Exp(x) and Ln(x), which can be implemented with a Taylor expansion. b^x can the be found with:
b^x = (e^(ln b))^x = e^(x.ln(b))
(See: Wikipedia - Powers via Logarithms)
This looks a bit tricky and complicated to me, though. Am I likely to get the implementation smaller then the compiler's math library, and can I simplify it for my special case (i.e. base = 0-1, exponent always 0.19029)?
Note that RAM usage is OK at the moment, but I've run low on Flash (used for code storage). Speed is not critical. Somebody has already suggested that I use a bigger micro with more flash memory, but that sounds like profligate wastefulness!
[EDIT] I was being lazy when I said "(p/p0) ratio would always be in the range 0-1". Actually it will never reach 0, and I did some calculations last night and decided that in fact a range of 0.3 - 1 would be quite adequate! This mean that some of the simpler solutions below should be suitable. Also, the "k" in the above is 44330, and I'd like the error in the final result to be less than 0.1. I guess that means an error in the (p/p0)^0.19029 needs to be less than 1/443300 or 2.256e-6
Use splines. The relevant part of the function is shown in the figure below. It varies approximately like the 5th root, so the problematic zone is close to p / p0 = 0. There is mathematical theory how to optimally place the knots of splines to minimize the error (see Carl de Boor: A Practical Guide to Splines). Usually one constructs the spline in B form ahead of time (using toolboxes such as Matlab's spline toolbox - also written by C. de Boor), then converts to Piecewise Polynomial representation for fast evaluation.
In C. de Boor, PGS, the function g(x) = sqrt(x + 1) is actually taken as an example (Chapter 12, Example II). This is exactly what you need here. The book comes back to this case a few times, since it is admittedly a hard problem for any interpolation scheme due to the infinite derivatives at x = -1. All software from PGS is available for free as PPPACK in netlib, and most of it is also part of SLATEC (also from netlib).
Edit (Removed)
(Multiplying by x once does not significantly help, since it only regularizes the first derivative, while all other derivatives at x = 0 are still infinite.)
Edit 2
My feeling is that optimally constructed splines (following de Boor) will be best (and fastest) for relatively low accuracy requirements. If the accuracy requirements are high (say 1e-8), one may be forced to get back to the algorithms that mathematicians have been researching for centuries. At this point, it may be best to simply download the sources of glibc and copy (provided GPL is acceptable) whatever is in
glibc-2.19/sysdeps/ieee754/dbl-64/e_pow.c
Since we don't have to include the whole math.h, there shouldn't be a problem with memory, but we will only marginally profit from having a fixed exponent.
Edit 3
Here is an adapted version of e_pow.c from netlib, as found by #Joni. This seems to be the grandfather of glibc's more modern implementation mentioned above. The old version has two advantages: (1) It is public domain, and (2) it uses a limited number of constants, which is beneficial if memory is a tight resource (glibc's version defines over 10000 lines of constants!). The following is completely standalone code, which calculates x^0.19029 for 0 <= x <= 1 to double precision (I tested it against Python's power function and found that at most 2 bits differed):
#define __LITTLE_ENDIAN
#ifdef __LITTLE_ENDIAN
#define __HI(x) *(1+(int*)&x)
#define __LO(x) *(int*)&x
#else
#define __HI(x) *(int*)&x
#define __LO(x) *(1+(int*)&x)
#endif
static const double
bp[] = {1.0, 1.5,},
dp_h[] = { 0.0, 5.84962487220764160156e-01,}, /* 0x3FE2B803, 0x40000000 */
dp_l[] = { 0.0, 1.35003920212974897128e-08,}, /* 0x3E4CFDEB, 0x43CFD006 */
zero = 0.0,
one = 1.0,
two = 2.0,
two53 = 9007199254740992.0, /* 0x43400000, 0x00000000 */
/* poly coefs for (3/2)*(log(x)-2s-2/3*s**3 */
L1 = 5.99999999999994648725e-01, /* 0x3FE33333, 0x33333303 */
L2 = 4.28571428578550184252e-01, /* 0x3FDB6DB6, 0xDB6FABFF */
L3 = 3.33333329818377432918e-01, /* 0x3FD55555, 0x518F264D */
L4 = 2.72728123808534006489e-01, /* 0x3FD17460, 0xA91D4101 */
L5 = 2.30660745775561754067e-01, /* 0x3FCD864A, 0x93C9DB65 */
L6 = 2.06975017800338417784e-01, /* 0x3FCA7E28, 0x4A454EEF */
P1 = 1.66666666666666019037e-01, /* 0x3FC55555, 0x5555553E */
P2 = -2.77777777770155933842e-03, /* 0xBF66C16C, 0x16BEBD93 */
P3 = 6.61375632143793436117e-05, /* 0x3F11566A, 0xAF25DE2C */
P4 = -1.65339022054652515390e-06, /* 0xBEBBBD41, 0xC5D26BF1 */
P5 = 4.13813679705723846039e-08, /* 0x3E663769, 0x72BEA4D0 */
lg2 = 6.93147180559945286227e-01, /* 0x3FE62E42, 0xFEFA39EF */
lg2_h = 6.93147182464599609375e-01, /* 0x3FE62E43, 0x00000000 */
lg2_l = -1.90465429995776804525e-09, /* 0xBE205C61, 0x0CA86C39 */
ovt = 8.0085662595372944372e-0017, /* -(1024-log2(ovfl+.5ulp)) */
cp = 9.61796693925975554329e-01, /* 0x3FEEC709, 0xDC3A03FD =2/(3ln2) */
cp_h = 9.61796700954437255859e-01, /* 0x3FEEC709, 0xE0000000 =(float)cp */
cp_l = -7.02846165095275826516e-09, /* 0xBE3E2FE0, 0x145B01F5 =tail of cp_h*/
ivln2 = 1.44269504088896338700e+00, /* 0x3FF71547, 0x652B82FE =1/ln2 */
ivln2_h = 1.44269502162933349609e+00, /* 0x3FF71547, 0x60000000 =24b 1/ln2*/
ivln2_l = 1.92596299112661746887e-08; /* 0x3E54AE0B, 0xF85DDF44 =1/ln2 tail*/
double pow0p19029(double x)
{
double y = 0.19029e+00;
double z,ax,z_h,z_l,p_h,p_l;
double y1,t1,t2,r,s,t,u,v,w;
int i,j,k,n;
int hx,hy,ix,iy;
unsigned lx,ly;
hx = __HI(x); lx = __LO(x);
hy = __HI(y); ly = __LO(y);
ix = hx&0x7fffffff; iy = hy&0x7fffffff;
ax = x;
/* special value of x */
if(lx==0) {
if(ix==0x7ff00000||ix==0||ix==0x3ff00000){
z = ax; /*x is +-0,+-inf,+-1*/
return z;
}
}
s = one; /* s (sign of result -ve**odd) = -1 else = 1 */
double ss,s2,s_h,s_l,t_h,t_l;
n = ((ix)>>20)-0x3ff;
j = ix&0x000fffff;
/* determine interval */
ix = j|0x3ff00000; /* normalize ix */
if(j<=0x3988E) k=0; /* |x|<sqrt(3/2) */
else if(j<0xBB67A) k=1; /* |x|<sqrt(3) */
else {k=0;n+=1;ix -= 0x00100000;}
__HI(ax) = ix;
/* compute ss = s_h+s_l = (x-1)/(x+1) or (x-1.5)/(x+1.5) */
u = ax-bp[k]; /* bp[0]=1.0, bp[1]=1.5 */
v = one/(ax+bp[k]);
ss = u*v;
s_h = ss;
__LO(s_h) = 0;
/* t_h=ax+bp[k] High */
t_h = zero;
__HI(t_h)=((ix>>1)|0x20000000)+0x00080000+(k<<18);
t_l = ax - (t_h-bp[k]);
s_l = v*((u-s_h*t_h)-s_h*t_l);
/* compute log(ax) */
s2 = ss*ss;
r = s2*s2*(L1+s2*(L2+s2*(L3+s2*(L4+s2*(L5+s2*L6)))));
r += s_l*(s_h+ss);
s2 = s_h*s_h;
t_h = 3.0+s2+r;
__LO(t_h) = 0;
t_l = r-((t_h-3.0)-s2);
/* u+v = ss*(1+...) */
u = s_h*t_h;
v = s_l*t_h+t_l*ss;
/* 2/(3log2)*(ss+...) */
p_h = u+v;
__LO(p_h) = 0;
p_l = v-(p_h-u);
z_h = cp_h*p_h; /* cp_h+cp_l = 2/(3*log2) */
z_l = cp_l*p_h+p_l*cp+dp_l[k];
/* log2(ax) = (ss+..)*2/(3*log2) = n + dp_h + z_h + z_l */
t = (double)n;
t1 = (((z_h+z_l)+dp_h[k])+t);
__LO(t1) = 0;
t2 = z_l-(((t1-t)-dp_h[k])-z_h);
/* split up y into y1+y2 and compute (y1+y2)*(t1+t2) */
y1 = y;
__LO(y1) = 0;
p_l = (y-y1)*t1+y*t2;
p_h = y1*t1;
z = p_l+p_h;
j = __HI(z);
i = __LO(z);
/*
* compute 2**(p_h+p_l)
*/
i = j&0x7fffffff;
k = (i>>20)-0x3ff;
n = 0;
if(i>0x3fe00000) { /* if |z| > 0.5, set n = [z+0.5] */
n = j+(0x00100000>>(k+1));
k = ((n&0x7fffffff)>>20)-0x3ff; /* new k for n */
t = zero;
__HI(t) = (n&~(0x000fffff>>k));
n = ((n&0x000fffff)|0x00100000)>>(20-k);
if(j<0) n = -n;
p_h -= t;
}
t = p_l+p_h;
__LO(t) = 0;
u = t*lg2_h;
v = (p_l-(t-p_h))*lg2+t*lg2_l;
z = u+v;
w = v-(z-u);
t = z*z;
t1 = z - t*(P1+t*(P2+t*(P3+t*(P4+t*P5))));
r = (z*t1)/(t1-two)-(w+z*w);
z = one-(r-z);
__HI(z) += (n<<20);
return s*z;
}
Clearly, 50+ years of research have gone into this, so it's probably very hard to do any better. (One has to appreciate that there are 0 loops, only 2 divisions, and only 6 if statements in the whole algorithm!) The reason for this is, again, the behavior at x = 0, where all derivatives diverge, which makes it extremely hard to keep the error under control: I once had a spline representation with 18 knots that was good up to x = 1e-4, with absolute and relative errors < 5e-4 everywhere, but going to x = 1e-5 ruined everything again.
So, unless the requirement to go arbitrarily close to zero is relaxed, I recommend using the adapted version of e_pow.c given above.
Edit 4
Now that we know that the domain 0.3 <= x <= 1 is sufficient, and that we have very low accuracy requirements, Edit 3 is clearly overkill. As #MvG has demonstrated, the function is so well behaved that a polynomial of degree 7 is sufficient to satisfy the accuracy requirements, which can be considered a single spline segment. #MvG's solution minimizes the integral error, which already looks very good.
The question arises as to how much better we can still do? It would be interesting to find the polynomial of a given degree that minimizes the maximum error in the interval of interest. The answer is the minimax
polynomial, which can be found using Remez' algorithm, which is implemented in the Boost library. I like #MvG's idea to clamp the value at x = 1 to 1, which I will do as well. Here is minimax.cpp:
#include <ostream>
#define TARG_PREC 64
#define WORK_PREC (TARG_PREC*2)
#include <boost/multiprecision/cpp_dec_float.hpp>
typedef boost::multiprecision::number<boost::multiprecision::cpp_dec_float<WORK_PREC> > dtype;
using boost::math::pow;
#include <boost/math/tools/remez.hpp>
boost::shared_ptr<boost::math::tools::remez_minimax<dtype> > p_remez;
dtype f(const dtype& x) {
static const dtype one(1), y(0.19029);
return one - pow(one - x, y);
}
void out(const char *descr, const dtype& x, const char *sep="") {
std::cout << descr << boost::math::tools::real_cast<double>(x) << sep << std::endl;
}
int main() {
dtype a(0), b(0.7); // range to optimise over
bool rel_error(false), pin(true);
int orderN(7), orderD(0), skew(0), brake(50);
int prec = 2 + (TARG_PREC * 3010LL)/10000;
std::cout << std::scientific << std::setprecision(prec);
p_remez.reset(new boost::math::tools::remez_minimax<dtype>(
&f, orderN, orderD, a, b, pin, rel_error, skew, WORK_PREC));
out("Max error in interpolated form: ", p_remez->max_error());
p_remez->set_brake(brake);
unsigned i, count(50);
for (i = 0; i < count; ++i) {
std::cout << "Stepping..." << std::endl;
dtype r = p_remez->iterate();
out("Maximum Deviation Found: ", p_remez->max_error());
out("Expected Error Term: ", p_remez->error_term());
out("Maximum Relative Change in Control Points: ", r);
}
boost::math::tools::polynomial<dtype> n = p_remez->numerator();
for(i = n.size(); i--; ) {
out("", n[i], ",");
}
}
Since all parts of boost that we use are header-only, simply build with:
c++ -O3 -I<path/to/boost/headers> minimax.cpp -o minimax
We finally get the coefficients, which are after multiplication by 44330:
24538.3409, -42811.1497, 34300.7501, -11284.1276, 4564.5847, 3186.7541, 8442.5236, 0.
The following error plot demonstrates that this is really the best possible degree-7 polynomial approximation, since all extrema are of equal magnitude (0.06659):
Should the requirements ever change (while still keeping well away from 0!), the C++ program above can be simply adapted to spit out the new optimal polynomial approximation.
Instead of a lookup table, I'd use a polynomial approximation:
1 - x0.19029 ≈ - 1073365.91783x15 + 8354695.40833x14 - 29422576.6529x13 + 61993794.537x12 - 87079891.4988x11 + 86005723.842x10 - 61389954.7459x9 + 32053170.1149x8 - 12253383.4372x7 + 3399819.97536x6 - 672003.142815x5 + 91817.6782072x4 - 8299.75873768x3 + 469.530204564x2 - 16.6572179869x + 0.722044145701
Or in code:
double f(double x) {
double fx;
fx = - 1073365.91783;
fx = fx*x + 8354695.40833;
fx = fx*x - 29422576.6529;
fx = fx*x + 61993794.537;
fx = fx*x - 87079891.4988;
fx = fx*x + 86005723.842;
fx = fx*x - 61389954.7459;
fx = fx*x + 32053170.1149;
fx = fx*x - 12253383.4372;
fx = fx*x + 3399819.97536;
fx = fx*x - 672003.142815;
fx = fx*x + 91817.6782072;
fx = fx*x - 8299.75873768;
fx = fx*x + 469.530204564;
fx = fx*x - 16.6572179869;
fx = fx*x + 0.722044145701;
return fx;
}
I computed this in sage using the least squares approach:
f(x) = 1-x^(19029/100000) # your function
d = 16 # number of terms, i.e. degree + 1
A = matrix(d, d, lambda r, c: integrate(x^r*x^c, (x, 0, 1)))
b = vector([integrate(x^r*f(x), (x, 0, 1)) for r in range(d)])
A.solve_right(b).change_ring(RDF)
Here is a plot of the error this will entail:
Blue is the error from my 16 term polynomial, while red is the error you'd get from piecewise linear interpolation with 16 equidistant values. As you can see, both errors are quite small for most parts of the range, but will become really huge close to x=0. I actually clipped the plot there. If you can somehow narrow the range of possible values, you could use that as the domain for the integration, and obtain an even better fit for the relevant range. At the cost of worse fit outside, of course. You could also increase the number of terms to obtain a closer fit, although that might also lead to higher oscillations.
I guess you can also combine this approach with the one Stefan posted: use his to split the domain into several parts, then use mine to find a close low degree polynomial for each part.
Update
Since you updated the specification of your question, with regard to both the domain and the error, here is a minimal solution to fit those requirements:
44330(1 - x0.19029) ≈ + 23024.9160933(1-x)7 - 39408.6473636(1-x)6 + 31379.9086193(1-x)5 - 10098.7031260(1-x)4 + 4339.44098317(1-x)3 + 3202.85705860(1-x)2 + 8442.42528906(1-x)
double f(double x) {
double fx, x1 = 1. - x;
fx = + 23024.9160933;
fx = fx*x1 - 39408.6473636;
fx = fx*x1 + 31379.9086193;
fx = fx*x1 - 10098.7031260;
fx = fx*x1 + 4339.44098317;
fx = fx*x1 + 3202.85705860;
fx = fx*x1 + 8442.42528906;
fx = fx*x1;
return fx;
}
I integrated x from 0.293 to 1 or equivalently 1 - x from 0 to 0.707 to keep the worst oscillations outside the relevant domain. I also omitted the constant term, to ensure an exact result at x=1. The maximal error for the range [0.3, 1] now occurs at x=0.3260 and amounts to 0.0972 < 0.1. Here is an error plot, which of course has bigger absolute errors than the one above due to the scale factor k=44330 which has been included here.
I can also state that the first three derivatives of the function will have constant sign over the range in question, so the function is monotonic, convex, and in general pretty well-behaved.
Not meant to answer the question, but it illustrates the Road Not To Go, and thus may be helpful:
This quick-and-dirty C code calculates pow(i, 0.19029) for 0.000 to 1.000 in steps of 0.01. The first half displays the error, in percents, when stored as 1/65536ths (as that theoretically provides slightly over 4 decimals of precision). The second half shows both interpolated and calculated values in steps of 0.001, and the difference between these two.
It kind of looks okay if you read from the bottom up, all 100s and 99.99s there, but about the first 20 values from 0.001 to 0.020 are worthless.
#include <stdio.h>
#include <math.h>
float powers[102];
int main (void)
{
int i, as_int;
double as_real, low, high, delta, approx, calcd, diff;
printf ("calculating and storing:\n");
for (i=0; i<=101; i++)
{
as_real = pow(i/100.0, 0.19029);
as_int = (int)round(65536*as_real);
powers[i] = as_real;
diff = 100*as_real/(as_int/65536.0);
printf ("%.5f %.5f %.5f ~ %.3f\n", i/100.0, as_real, as_int/65536.0, diff);
}
printf ("\n");
printf ("-- interpolating in 1/10ths:\n");
for (i=0; i<1000; i++)
{
as_real = i/1000.0;
low = powers[i/10];
high = powers[1+i/10];
delta = (high-low)/10.0;
approx = low + (i%10)*delta;
calcd = pow(as_real, 0.19029);
diff = 100.0*approx/calcd;
printf ("%.5f ~ %.5f = %.5f +/- %.5f%%\n", as_real, approx, calcd, diff);
}
return 0;
}
You can find a complete, correct standalone implementation of pow in fdlibm. It's about 200 lines of code, about half of which deal with special cases. If you remove the code that deals with special cases you're not interested in I doubt you'll have problems including it in your program.
LutzL's answer is a really good one: Calculate your power as (x^1.52232)^(1/8), computing the inner power by spline interpolation or another method. The eighth root deals with the pathological non-differentiable behavior near zero. I took the liberty of mocking up an implementation this way. The below, however, only does a linear interpolation to do x^1.52232, and you'd need to get the full coefficients using your favorite numerical mathematics tools. You'll adding scarcely 40 lines of code to get your needed power, plus however many knots you choose to use for your spline, as dicated by your required accuracy.
Don't be scared by the #include <math.h>; it's just for benchmarking the code.
#include <stdio.h>
#include <math.h>
double my_sqrt(double x) {
/* Newton's method for a square root. */
int i = 0;
double res = 1.0;
if (x > 0) {
for (i = 0; i < 10; i++) {
res = 0.5 * (res + x / res);
}
} else {
res = 0.0;
}
return res;
}
double my_152232(double x) {
/* Cubic spline interpolation for x ** 1.52232. */
int i = 0;
double res = 0.0;
/* coefs[i] will give the cubic polynomial coefficients between x =
i and x = i+1. Out of laziness, the below numbers give only a
linear interpolation. You'll need to do some work and research
to get the spline coefficients. */
double coefs[3][4] = {{0.0, 1.0, 0.0, 0.0},
{-0.872526, 1.872526, 0.0, 0.0},
{-2.032706, 2.452616, 0.0, 0.0}};
if ((x >= 0) && (x < 3.0)) {
i = (int) x;
/* Horner's method cubic. */
res = (((coefs[i][3] * x + coefs[i][2]) * x) + coefs[i][1] * x)
+ coefs[i][0];
} else if (x >= 3.0) {
/* Scaled x ** 1.5 once you go off the spline. */
res = 1.024824 * my_sqrt(x * x * x);
}
return res;
}
double my_019029(double x) {
return my_sqrt(my_sqrt(my_sqrt(my_152232(x))));
}
int main() {
int i;
double x = 0.0;
for (i = 0; i < 1000; i++) {
x = 1e-2 * i;
printf("%f %f %f \n", x, my_019029(x), pow(x, 0.19029));
}
return 0;
}
EDIT: If you're just interested in a small region like [0,1], even simpler is to peel off one sqrt(x) and compute x^1.02232, which is quite well behaved, using a Taylor series:
double my_152232(double x) {
double part_050000 = my_sqrt(x);
double part_102232 = 1.02232 * x + 0.0114091 * x * x - 3.718147e-3 * x * x * x;
return part_102232 * part_050000;
}
This gets you within 1% of the exact power for approximately [0.1,6], though getting the singularity exactly right is always a challenge. Even so, this three-term Taylor series gets you within 2.3% for x = 0.001.
I'm receiving PCM data trough socket connection in packets containing 320 samples. Sample rate of sound is 8000 samples per second. I am doing with it something like this:
int size = 160 * 2;//160;
int isinverse = 1;
kiss_fft_scalar zero;
memset(&zero,0,sizeof(zero));
kiss_fft_cpx fft_in[size];
kiss_fft_cpx fft_out[size];
kiss_fft_cpx fft_reconstructed[size];
kiss_fftr_cfg fft = kiss_fftr_alloc(size*2 ,0 ,0,0);
kiss_fftr_cfg ifft = kiss_fftr_alloc(size*2,isinverse,0,0);
for (int i = 0; i < size; i++) {
fft_in[i].r = zero;
fft_in[i].i = zero;
fft_out[i].r = zero;
fft_out[i].i = zero;
fft_reconstructed[i].r = zero;
fft_reconstructed[i].i = zero;
}
// got my data through socket connection
for (int i = 0; i < size; i++) {
// samples are type of short
fft_in[i].r = samples[i];
fft_in[i].i = zero;
fft_out[i].r = zero;
fft_out[i].i = zero;
}
kiss_fftr(fft, (kiss_fft_scalar*) fft_in, fft_out);
kiss_fftri(ifft, fft_out, (kiss_fft_scalar*)fft_reconstructed);
// lets normalize samples
for (int i = 0; i < size; i++) {
short* samples = (short*) bufTmp1;
samples[i] = rint(fft_reconstructed[i].r/(size*2));
}
After that I fill OpenAL buffers and play them. Everything works just fine but I would like to do some filtering of audio between kiss_fftr and kiss_fftri. Starting point as I think for this is to convert sound from time domain to frequency domain, but I don't really understand what kind of data I'm receiving from kiss_fftr function. What information is stored in each of those complex number, what its real and imaginary part can tell me about frequency. And I don't know which frequencies are covered (what frequency span) in fft_out - which indexes corresponds to which frequencies.
I am total newbie in signal processing and Fourier transform topics.
Any help?
Before you jump in with both feet into a C implementation, get familiar with digital filters, esp FIR filters.
You can design the FIR filter using something like GNU Octave's signal toolbox. Look at the command fir1(the simplest), firls, or remez. Alternately, you might be able to design a FIR filter through a web page. A quick web search for "online fir filter design" found this (I have not used it, but it appears to use the equiripple design used in the remez or firpm command )
Try implementing your filter first with a direct convolution (without FFTs) and see if the speed is acceptable -- that is an easier path. If you need an FFT-based approach, there is a sample implementation of overlap-save in the kissfft/tools/kiss_fastfir.c file.
I will try to answer your questions directly.
// a) the real and imaginary components of the output need to be combined to calculate the amplitude at each frequency.
float ar,ai,scaling;
scaling=1.0/(float)size;
// then for each output [i] from the FFT...
ar = fft_out[i].r;
ai = fft_out[i].i;
amplitude[i] = 2.0 * sqrtf( ar*ar + ai*ai ) * scaling ;
// b) which index refers to which frequency? This can be calculated as follows. Only the first half of the FFT results are needed (assuming your 8KHz sampling rate)
for(i=1;i<(size/2);i++) freq = (float)i / (1/8000) / (float)size ;
// c) phase (range +/- PI) for each frequency is calculated like this:
phase[i] = phase = atan2(fft_out[i].i / fft_out[i].r);
What you might want to investigate is FFT fast convolution using overlap add or overlap save algorithms. You will need to expand the length of each FFT by the length of the impulse of your desired filter. This is because (1) FFT/IFFT convolution is circular, and (2) each index in the FFT array result corresponds to almost all frequencies (a Sinc shaped response), not just one (even if mostly near one), so any single bin modification will leak throughout the entire frequency response (except certain exact periodic frequencies).