Hi, I searched the internet to find out the difference between a fixed point & a floating point DSP. To my surprise, I did not find a well defined difference. Plz, can anyone differentiate between the two so that a layman can also understand ? Thanx in advance... Sandeep Chikkerur wrote: > > Hi, > > I searched the internet to find out the difference between a fixed > point & a floating point DSP. > To my surprise, I did not find a well defined difference. > > Plz, can anyone differentiate between the two so that a layman can > also understand ...

HI everyone, I was wondering if anyone can recommend me good books or papers on fixed-point IIR implementation strategies on FPGAs or CPUs. Especially regarding fc versus Fs and coefficient quantization. I have issues with an order-1 DC-removal filter with Fs = 100 MHz and Fcut = 250 kHz. Basically, I need lots of precision in the intermediate calculations which impact the number of bits for the multiplier and thus the maximum frequency of my FPGA design. Best regards On 11/14/2011 1:15 PM, Benjamin Couillard wrote: > HI everyone, > > I was wondering if anyone can recommend me good...

Hi I am trying to design a IIR filter using a 16-bit fixed point DSP. I nee to calculate the coeff in the form of second order sections. I would lik to do this with MATLAB. Does anyone have some MATLAB code that will d this. I found a routine on Texas Instrument site but as well as the coef it creates some scaling factors for the inputs. I dont want this as I hav seen other programs such as Filter Express that just give the Coeff. Thanks J This message was sent using the Comp.DSP web interface o www.DSPRelated.com OK. You have several kinds of IIR filter, for instance elliptic (the be...

Hello, I tried implementing a LPF biquad (that works perfectly in floating point) with: a = [1.00000, -1.90887, 0.91129], b = [6.1150e-04, 1.2230e-03, 6.1150e-04] in fixed point / integer, with 16 bit coefficients, 10 bit input signal and a 32 bit accumulator. I suspect the coefficient resolution is not enough. I tried: - the direct form 1 and direct form 2 transposed structures, but the output is zero if the input signal is not very high -- because of the small b coefficients, the signal is attenuated very much in the FIR portion of the filter and is lost; int biquad_ab(d, a0...

Hye, I have a question to try to find a way to improve my processing. I use a multirate filter with decimation. There is a decimation by 40. One FIr is used then decimation, and the IIR with decimation, then a second IIR with decimation, then again one IIR and decimation and at last one FIr . I use a fix point DSP. The filter are designed in 16 bits inputs / output (32 bits for computation). It appears that the results is that i have in output a 12bits resoutio and 4 lsb bits are lost during the stages. How can i improve this? Should i change the IIR design? or thedecimation? thank yo...

Hi Does anyone have some guidelines on how to implement a 4'th order low-pass Butterworth IIR filter in fixed point. My cut-off frequency is relatively close to the DC frequency so high precision is needed for the coefficients. What about realization structure and so on! I have implemented the bit-flipping algorithm in http://www.cmsa.wmin.ac.uk/~artur/pdf/Paper16.pdf for quantization of coefficients and it indeed works, but does some other techniques allow for further reductions of number of bits used to represent the coefficients. Thomas I've found the normalized lattice-ladder...

Hello. I am a professional programmer, but I am doing DSP just as a hobby. Three years ago I designed a morse / PSK31 decoder on a 16bit no- multiplier controller. After rouhgly 3 years of absence in the DSP field I am back to design a HAM shortwave software defined radio using a beautiful new part from Texas Instruments - TLV320AIC3254 codec with miniDSP core. The part could do all the signal processing I need with just a 5x5mm board space including earphone driver. That's an engineer's dream come true. The goal is to downconvert a SSB voice channel down to audio with a quadrature m...

i am designing one 1/3 octave band spectrogram analysis I use multirate filter bank to realize that spectrogram. It goes this way 1) first, from the biggest frequency value, I use three IIR bandpass filter, then calculate the std value. 2) Then decimate the input by 2 through(one 30 order FIR filter, then resampling the input), then use the same three coffeicents to calculate the successive three band, 3) then repeat the step 2) til the lowest frequency. there is 45 channels (one channel one ceter freuquency) I got the right result throught MATLAB But i got some different result ...

Hello, I am sure this is a pretty simple question, but I am having some ambiguity. I need to confirm what is happening exactly. Any information is really appreciated. So this is a Sharc DSP I am working with. Question is pretty simple. Rn = FIX Fx; FIX description from the manual is as follows: "Converts the floating-point operand in Fx to a twos-complement 32-bit fixed-point integer result. If the MODE1 register TRUNC bit=1, the FIX operation truncates the mantissa towards –Infinity. If the TRUNC bit=0, the FIX operation rounds the mantissa towards the nearest integer....

Hi, I have a floating point c code to implement a navigation system algorithm. In order to implement it in the 64x+ fixed point DSP processor, i need to have the code in fixed point. I learnt that there are few processor specific IQmath libraries aiding this purpose. Is it practical to write the code using these libraries or should i manually write codes for each operation by scaling the inputs for each line of the code, in which case i have to create look up table for the trigonometric functions etc which is a harder process to begin with.. Thanks, Divya On 2/13/12 11:38 AM, divya_di...

hello all, I am new in dsp programing and i an using fixed point DSP, as read FIxed point arithmatic it seems to be confusing so plz suggest an link or document having good practical explaination of the topic, or pl give some examples for understanding, Thanks in Advance, Rahul. jadhav_rahul wrote: > hello all, > I am new in dsp programing and i an using fixed point DSP, as i > read FIxed point arithmatic it seems to be confusing so plz suggest any > link or document having good practical explaination of the topic, or plz > give some examples for understan...

Hi, i want to implement the equation p3 = p3*0.5/cos(3*PI/8); into 16bit processor, where p3 is a 16 bit value; any pointers as to how to go about it -vijay n_vijay_anand wrote: > Hi, > > i want to implement the equation > > p3 = p3*0.5/cos(3*PI/8); > > into 16bit processor, where p3 is a 16 bit value; > > any pointers as to how to go about it It looks like '0.5/cos(3*PI/8)' is a constant, so simplify it. cos(3*PI/8) = 0.99978861 0.5/0.99978861 = 0.50010571 Depending on the algorithm you might get away with simply dividing by 2: p3 >...

Hi i am using built-in Matlab functions to design IIR fixed point filter. For example: [b a] = cheby1(2,6,0.25) hd = dfilt.df1(b,a) set(hd,'arithmetic','fixed') Is there a simple method to check the coefficient values for fixed arithmetic? Precisely the ones which will appear in HDL code ( generatehdl(hd) ). Because i want to make a parametrized filter with a coefficient lookup table for different cutoff frequencies. ...

Hi everyone! I need a fixed point implementation for the LMS algorithm;I have to implement it on a TI dsp board (c6711 or a c5000); can anyone suggest me where I can find this code? thank you guys! f Use this: It is a single step for fixed point LMS. c_ptr points to coefficient, r_ptr points to history. muErr = e * mu; mu = 2048(trial), using fixed mu; accA, accB are at least 32 bit accumulators. /* Macro: Performs single step of filter and adapt */ #define ADAPT_AND_FILTER(accA, accB, muErr, c_ptr, r_ptr) \ accB = 16384; \ accB += ((i3...

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