Round-half-even or Round-toward-nearest

Started by Kathy_pdx in comp.dsp5 years ago 1 reply

I just finished the reading of An introduction to different rounding algorithms, http://www.eetimes.com/document.asp?doc_id=1274485. I...

I just finished the reading of An introduction to different rounding algorithms, http://www.eetimes.com/document.asp?doc_id=1274485. I must admit I never pay much attention on the trick of the quantization. The approach of Round-toward-nearest is always used in my design. Did anyone have practical experiences to show the DSP filter performance improved or degraded by using Round-half-even ins...


Re: Questions about equivalents of audio/video and digital/analog.

Started by Don Pearce in comp.dsp13 years ago 290 replies

On Sun, 19 Aug 2007 23:26:16 -0700, dplatt@radagast.org (Dave Platt) wrote: > "Digital" and "subject to aliasing" are two different things. >...

On Sun, 19 Aug 2007 23:26:16 -0700, dplatt@radagast.org (Dave Platt) wrote: > "Digital" and "subject to aliasing" are two different things. > > As I believe the term "digital" is usually meant, it implies a > two-state (on/off) storage representation. It's not just that the > signal amplitude is quantized, but that the quantization uses a > power-of-two representation and storage system of so


Cancellation of reconstructed digital signal

Started by alexia in comp.dsp11 years ago 2 replies

Hello, I have an instrumentation amplifier with a gain of 100. The positive input (AO) is connected to a DAC. The negative input is...

Hello, I have an instrumentation amplifier with a gain of 100. The positive input (AO) is connected to a DAC. The negative input is taken from a previous analog stage and also connected to a ADC --> recorded signal (AI1). I put a replicate of the recorded signal (AI1) at AO. However, I got an amplified signal of quantization, noise, etc. I want to ask which is the best way to reconstruc


Sigma Delta Modulation - SNR calculation

Started by bpiletrue in comp.dsp15 years ago 3 replies

Hello All, I have created a simple routine in MATLAB that simulates a second order sigma delta modulator. I am now trying (with some...

Hello All, I have created a simple routine in MATLAB that simulates a second order sigma delta modulator. I am now trying (with some difficulty) to calculate the SNR of the modulator given sinusoidal inputs. My understanding of how this procedure should be carried out is as follows, 1) Build the output of the modulator by running my routine. The quantization error in the ouput would then b...


Fixed point implementation of 4'th order IIR filters

Started by Heureka in comp.dsp17 years ago 7 replies

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...

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...


FIR quantization effect ( implementing in fpga)

Started by sed_y in comp.dsp11 years ago 1 reply

hi, I am trying to implement an FIR filter, fixed point one, using labview and xilinx core generator. while the labview only method gives good...

hi, I am trying to implement an FIR filter, fixed point one, using labview and xilinx core generator. while the labview only method gives good result, the combination dosn't . so, let me explain how i have done computaion,and sb cld point are they correct? i create floating point filter,using kaiser window, of order 54,and 55(nh) coefficients. The max,min values are 0.1544 and -0.1733. to repr...


IIR filter coefficient quantization newbie

Started by Alxpert in comp.dsp15 years ago 2 replies

I designing a IIR digital filter with the following parameters (Fs = 500k,fc = 1k,fs = 10k,wc = 1db,ws = 40db} using matlab. And I got...

I designing a IIR digital filter with the following parameters (Fs = 500k,fc = 1k,fs = 10k,wc = 1db,ws = 40db} using matlab. And I got the following coefficient using the following .m file. A1 A2 B2 B0 B1 { -1.9725 0.9733 0.0001 0.0001 0.0001 -0.9733 0 0 0.0475 0.0475 } Fs = 500000; fc = 1000; fs = 10000; Rp = 1; ...


number of bits to compute DFT

Started by sharanbr in comp.dsp6 years ago 15 replies

Hello All, I am reading DFT topic where quantization effect is described. One of the example in the book calculates the number of bits...

Hello All, I am reading DFT topic where quantization effect is described. One of the example in the book calculates the number of bits needed to compute the DFT of a 1024 point sequence with a SNR of 30 dB. Here b = 15 bits. My question is whether the size of the sample size does not matter at all for computation of the DFT? If input is composed of 24 bits (for example), would number o...


adding dither in audio signals

Started by Anonymous in comp.dsp15 years ago 15 replies

hi, I had certain querries about adding dither in audio signals. 1) I presume one of the reasons to add it is to compensate for quantisation...

hi, I had certain querries about adding dither in audio signals. 1) I presume one of the reasons to add it is to compensate for quantisation error while converting analog to digital. Is there any other reason for adding it? 2) Is it added only when the signal strength is zero, or would it be advantageous to add it always. If quantization error is the primary consideration, I pr...


Increase resolution through decimation

Started by mariking in comp.dsp13 years ago 3 replies

Hi I need to take a standard 8bit ADC which samples at 40Msps increase the resolution by about 4 or 5 bits. I know that oversampling spreads...

Hi I need to take a standard 8bit ADC which samples at 40Msps increase the resolution by about 4 or 5 bits. I know that oversampling spreads the quantization noise over a wider bandwidth thereby decreasing the SQNR. So i can increase the resolution through decimation? How do I increase the resolution through decimation? It decimates the signal frequency but i don't physically know how to increa...


Sigma delta conversion

Started by thom in comp.dsp14 years ago 1 reply

Hi everyone, I have a question about sigma delta conversion. I know that it permits to reduce quantization noise in a receiver. But what about...

Hi everyone, I have a question about sigma delta conversion. I know that it permits to reduce quantization noise in a receiver. But what about the noise at the input of the receiver? (I'm talking about the white noise floor that all RF signals contain) Thanks Thom


Re: DSP Trick: Fixed Point DC Blocking Filter with Noise-Shaping

Started by Darrell in comp.dsp13 years ago 52 replies

r b-j posted a trick back in 1999 to remove the DC of a signal, which I recently ran across in DSP guru. It was basically a...

r b-j posted a trick back in 1999 to remove the DC of a signal, which I recently ran across in DSP guru. It was basically a differentiator followed by a leaky integrator with some tricks to remove the quantization noise. I was wanting to give it a try but I saw something in the 56K assembly code which looked incorrect. The original post can be found at http://groups.google.com/group/comp.d...


Re: Theoretically Highest Quality of PCM Audio

Started by Clay S. Turner in comp.dsp17 years ago

Hello Bob, For uniform quantization, the S/N is 6n+4.77+(Smean/Speak) Smean/Speak is the mean to peak ratio for the signal. For a sine wave...

Hello Bob, For uniform quantization, the S/N is 6n+4.77+(Smean/Speak) Smean/Speak is the mean to peak ratio for the signal. For a sine wave it is just -3 dB; however for speech it will be much higher. It will be more like -13dB. And of course music will probably be closer to the speech value than the sine wave value. Look at Rabiner & Schafer "Digital Processing of Speech Signals" for ...


Reducing u-law noise?

Started by Filip Larsen in comp.dsp13 years ago 8 replies

I have a 200-3400 Hz voice signal in a 22050 Hz sampled 16-bit PCM file that I have to deliver to a sound speaker computer system that only...

I have a 200-3400 Hz voice signal in a 22050 Hz sampled 16-bit PCM file that I have to deliver to a sound speaker computer system that only accepts 22050 Hz u-law. However, the u-law encoding process seems to introduce full spectrum (quantization?) noise so the result is a 200-3400 Hz signal overlaid with 0-11 kHz noise, thus giving very perceptible high frequency noise during playback s...


Phase Vocoder and Vector Quantization

Started by Soren Nielsen in comp.dsp14 years ago

Hi, I've read Dan Ellis article on "Model-Based Monaural Source Separation Using a Vector-Quantized Phase-Vocoder...

Hi, I've read Dan Ellis article on "Model-Based Monaural Source Separation Using a Vector-Quantized Phase-Vocoder Representation". (http://www.ee.columbia.edu/~dpwe/pubs/EllisW06-pvocvq.pdf). with great interest. My problem is that I want to get a good estimate for the phase information after doing some manipulation on the magnitude of the short time fourier transform (STFT) of some spe...


Truncating accumulator output to reduce quantization error

Started by strider in comp.dsp13 years ago

I am performing continuous accumulation of the output of a multiplier block which has samples of two in-phase signals as its inputs. (I get...

I am performing continuous accumulation of the output of a multiplier block which has samples of two in-phase signals as its inputs. (I get samples from a 12 bit ADC which I convert to 16 bits and then feed them to the multiplier;however the input to the accumulator is the truncated 16 bit output of multiplier)I would like to know the width of the accumulator to be used and the method of extractin...


Nyquist, quantization and windowing gotcha's

Started by Richard Owlett in comp.dsp12 years ago 4 replies

I've been experimenting with a 3D version of spectrograms [amplitude vs frequency vs time]. Instead of plotting the spectrum of each time slice...

I've been experimenting with a 3D version of spectrograms [amplitude vs frequency vs time]. Instead of plotting the spectrum of each time slice (cf waterfall displays), I plot contours of equal amplitude across time. Borrowing from traditional spectrograms, each contour's color also indicates amplitude allowing adjacent contours to be distinguished when close together. The observed ar...


rephase my problem: how to reduce round-off error for hardware implementation?

Started by walala in comp.dsp17 years ago 9 replies

Dear all, I have asked question on DCT and quantization error and got some very good answers... Now I have a somewhat redefined problem to...

Dear all, I have asked question on DCT and quantization error and got some very good answers... Now I have a somewhat redefined problem to attack and I hope experts here can give me some thoughts: In ASIC implementation(I am currently using VHDL) of DCT, how to efficiently reduce round-off or finite word length error? For instance, a DCT transform is Y=D*X*D', where D= D = 0.35...


basic questions on jpeg compression

Started by manishp in comp.dsp9 years ago 17 replies

Folks, I have few basic questions on related to compression used in JPEG. 1. Is the DCT applied over Y, CB and CR separately? 2. Is there...

Folks, I have few basic questions on related to compression used in JPEG. 1. Is the DCT applied over Y, CB and CR separately? 2. Is there a specific reason why DCT is chosen over DFT? 3. Can quantization step be roughly seen as a filter since it basically tones down high frequency components? As an observation on point 3 above, we know that filter aims to achieve high attenuation i...


Continuous-time DSP with no sampling

Started by Yannis in comp.dsp15 years ago 72 replies

In principle, sampling is not necessary in order to do filtering digitally. This is discussed in the following paper: Y. Tsividis, "Digital...

In principle, sampling is not necessary in order to do filtering digitally. This is discussed in the following paper: Y. Tsividis, "Digital signal processing in continuous time: a possibility for avoiding aliasing and reducing quantization error", Proc. 2004 IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, vol. II, pp. 589-592, Montreal, May 2004. (If you are interested but can...