Adaptive Filter | ADPCM | ADSP | ADSP-2181 | Aliasing | AMR | Anti-Aliasing | ARMA | Autocorrelation | AutoCovariance | Beamforming | Bessel | Blackfin | Butterworth | C6713 | CCS | Chebyshev | CIC Filter | Circular Convolution | Code Composer Studio | Comb Filter | Compression | Convolution | Cross Correlation | DCT | Decimation | Deconvolution | Demodulation | DM642 | DSP Boards | DSP/BIOS | DTMF | Echo Cancellation | Equalization | Equalizer | ETSI | EZLITE (Ez-kit Lite) | FFT | FFTW | FIR Filter | Fixed Point | FSK | G.711 | G.723 | G.729 | Gaussian Noise | Goertzel | GPIO | Hilbert Transform | IFFT | IIR Filter | Interpolation | Invariance | JTAG | Kalman | Laplace Transform | Levinson | LPC | McBSP | MIPS | Modulation | MPEG | Multirate | Notch Filter | Nyquist | OFDM | Oversampling | Pink Noise | Pitch | PLL | Polyphase | QAM | QDMA | Quantization | Quantizer | Radar | Random Noise | Reed Solomon | Remez | Resampling | RTDX | Sampling | Sharc | TI C6711 | Undersampling | Viterbi | Wavelets | White Noise | Wiener Filter | Windowing | XDS510PP | Z Transform

Adaptive Equilization

I'm looking at the block diagram on page 247 of "Adaptive Signal Processing" from B.Widrow and S.D.Stearns. Anyway, the block diagram shows an LMS filter in the center, and the Quantizer on the bottom right. The output of the Quantizer goes to a subtraction element (which subtrats the LMS out...

How determine the path of signal?

I am digitizing the sampled signal be using a uniform quantizer, so the step (q) between two consective quantization levels is fixed and is "q = Vin/2^N" where 'q' is the quantization step, 'Vin' is the total range of quantizer and 'N' are the effective number of bits of quantizer. Now I want to ...

snr of quantizer using fixed logic

Hello all, I am doing SNR measurement of 8 bit quantizer implemented using "fix" logic (flooring towards zero for both + and - numbers). i am doing time domain and frequency domain estimation. TSNR = 43.6 dB (correct value). FSNR = 48.0 dB (in-...

Re: Dtihering, gain and again...

In article v5bBb.6707$Ho3.476@newsread1.news.atl.earthlink.net, Randy Yates at yates@ieee.org wrote on 12/08/2003 22:13: > robert bristow-johnson wrote: > > second moment, and we believe psychoacoustically that no one can hear > > couplings of higher moments. > > A belief which I've so...

Re: New Digital Audio Standard?

In article bsnpmq$du0ut$1@ID-167115.news.uni-berlin.de, Glenn Zelniker at glennz@z-sys.com wrote on 12/28/2003 18:41: > robert bristow-johnson wrote: > > > i don't know of any commercial product that actually does DSP in the 1-bit > > domain (except for the first stage of the decimator i...

Re: newbie question on deadzone quantization

Hi, > Can some one help me with deadzone quantization. What is its > difference when compared to normal quantization methods? There's no "normal" quantization, so to say. A quantizer is a mapping from a continuous set A of "signals" to a discrete set "D" of symbols such that every a in A...

Re: Fractional Saving

in article yLudnYt8VMfgj77ZnZ2dnUVZ_sOdnZ2d@giganews.com, snigdha at i_am_snigdha@yahoo.com wrote on 03/23/2006 16:26: > I have seen some posts regarding the implementation of IIR filters on a > fixed-point platform. I understand that using a cascade of DF1 IIR filters > combined with fract...

Re: projects in DSP

> > > > > "Rohit" == Rohit Pydimukkala writes: Rohit> I just started out with DSP and I am quite excited with my learning. Rohit> Now that I am interested in doing some small projects in DSP. Can Rohit> anyone let me know where I can find DSP projects on internet or Rohi...

Re: Dthering....

eric.jacobsen@delete.ieee.org (Eric Jacobsen) wrote in message news: ... > On Sun, 12 Oct 2003 01:12:58 -0400, Jerry Avins wrote: > > > Eric Jacobsen wrote: > > > > Jerry, you've stumped me. What's "fraction saving"? the way i would describe it is: 1st-order noise shaping with ...

Sigma Delta - Modeling s-function

Hey guys, I have a continuous-time sigma-delta converter that includes a continuous-time transfer function H(s) at the input, followed by a sampler, quantizer and a digital-to-analog converter (DAC) in the feedback path. I'm trying to model the transfer function H(s) that includes some non-i...

Re: IIR filter design

Erik de Castro Lopo wrote: > Bernhard Holzmayer wrote: > > ... > > Matlab caused a lot of pain when I tried to design even lower > > order filters (4th order Butterworth), when the relation between > > cutoff-frequency and sampling frequency decreases: > > try this: 4th order HPF and L...

Delta-Sigma Modulator Loop Filter analytical equation not useful?

Dear friends, I am encountering a confusion in Delta-Sigma Modulator. Most of the references use a linearized model to model the DSM, and an analytical transfer function can be derived. Refer to Shenoi's "Digital Signal Processing in Telecommunications", page 492, the denomenator of the trans...

Re: Linear PCM audio: 44.1 KHz, monaural, 1-bit-per-second

Andrew Reilly writes: > On Mon, 23 Jul 2007 10:35:03 -0400, Randy Yates wrote: > > > It's not the sample rate that prevents DSD from being linear PCM, but the > > fact that it does not utilize simple linear quantization. > > Yes, there's some fancy filtering up-stream, to ensure tha...

Re: Compensating DAC for low bit-resolution with high sample rate

The added accuracy by oversampling can be explained easily: The DAC produces the same absolute amount of quantization noise, but it spreads over a wider bandwidth. At least that's what I can show with an ideal quantizer in Matlab, and it's a good rule to remember. With real-world converters ...

Re: sigma-delta

kbc32@yahoo.com (kbc) writes: > Hi, > > Consider a sigma delta adc which first oversamples, does > noise-shaping and then filters and decimates to give final output > at Nyquist rate and with 16 bits per sample. OK, so this is a standard delta sigma ADC. Fine. > For this fin...

Re: Quantized FIR Coefficients and MSE Optimization

"Jim Beasley" wrote in message news:fad45c70.0306261456.30b58633@posting.google.com... > I have an MSE algorithm that optimizes using a closed-form solution > (based on estimates of the statistics). > > However my filter coefficients are quantized. I am sure I can get > further impro...

Sigms Delta Modulator Question

Hi all, If input a step respond into a SDM with a 1 bit quantizer, the output would be a stream of '1' and '0' s. IS that stream PCM or PDM?? Can I tell the dynamic range form the bit stream?? For example a 50nA step input with a output bit strem of "101010111011" and repeat. what ca...

Re: Bifurcations in delta-sigma A-to-D converter

For a first order loop, holding the voltage constant at 1/2 scale (also at 1/8, 1/4, ...) will show some limit cycle. This is a well documented problem. A higher order loop or a quantizer of more than 1 bit will reduced this problem. Another technique is to add random noise in the loop, jus...

Delta-Sigma Modulator question

Dear all, In a Delta-Sigma Modulator, after the quantizer, the output bit stream (+1 and -1) is fed back to the input and being subtract from the input(assuming first order loop). However, none of the references that I read mentioned how large this feedback signal should be. For example, if t...

Re: Digital Delta-Sigma DAC

In sci.electronics.basics Davy wrote: : Hi all, : I want to design a digital delta-sigma DAC. It includes a pulse density : modulated module and RC low-pass filter. : The pulse density modulated module is a Delta-Sigma type (one adder and : one substractor). But why use Delta-Sigma type ...

Re: IIR Form to prevent Limit Cycles?

jeff, approach 1. a second order IIR section that is free of zero input limit cycles is given on page 726 of sanjit k mitra book. fig 12.54. 3rd edition, digital signal processing. I am not too confident about this structure as limit cycle oscillations are caused by the quan...

how can I decide the points of blackman window?

Hi:all I am trying to measure the SNR of a second order sigma-delta ADC. I want to just do a FFT analysis for the output of the quantizer. So, I collect 65536 points for FFT. I was told that I should use a Blackman window here.But how many points of the blackman window should I use? Here are som...

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

Don Pearce wrote: > ... If you are doing digital signal processing, you are > doing arithmetic on the numbers that come out of an AtoD converter. > You can't do that with some voltage levels out of a quantizer. Transversal and recursive filters and correlators have been built that oper...

Re: Question about computing a "moving variance"

in article 35SdnWAbpqYEBqDfRVn-qA@rcn.net, Jerry Avins at jya@ieee.org wrote on 03/20/2005 10:49: > Rick Lyons wrote: > > ... > > > x(n)---> (*)------> (+)-----------+----> y(n) > > ^ ^ | > > | | v > > [1-p] |...

Re: 1-bit ADC for RF, is it possible?

> For RF transmission, I think sigma-delta techniques are a very big deal... From what I understand all S-D converters use a feedback loop. Basically this loop is used to trade sampling frequency with dynamic range. Without this feedback loop the system is a simple sampler with a 1-bit quanti...

Re: Shameless Plug

Tim Wescott wrote: > Embedded Systems Programming Magazine has sunk to a new low -- they made > _my_ article their cover story in the July issue. > > Tee hee. > > http://www.embedded.com/showArticle.jhtml?articleID=22101730 > Nicely done- I was anxious to see a analytical ex...

Re: How to decide the number of ADC bits needed for baseband receiver?

> Hi, > I am planning to build a BPSK/QPSK/8PSK/16QAM/32QAM/64QAM baseband > simulator in matlab and will eventually add in equalizer and decoder > for AWGN and fading channels. I have 2 question regarding the ADC > requirements: > 1)What is the minimum # of ADC bits needed to represent the IQ ...

Re: How can I decide the blackman window's points ?

stephenduan4513 wrote: > Hi:all > I am trying to measure the SNR of a second order sigma-delta ADC. I want > to just do a FFT analysis for the output of the quantizer. So, I collect > 65536 points for FFT. I was told that I should use a Blackman window > here.But how many points of the b...

Re: MATLAB BIT PRECISION!!

MuhammadAli.AA@gmail.com wrote: > Hi all !! > > I would like to know if it is possible to run Matlab on 16-bit and 8 > -bit processor !! > > Since it is already running on 32 bit !! > > what is the possible way to quantize to lower bit precision !!! > > Thanking you > > Ali...

Re: sigma-delta : unintuitive

kbc wrote: > > I am finding the sigma-delta adc difficult to understand. > > The only way, i feel, to reduce the quantization noise , > given the bit-resolution , is to sample non-uniformly. > > Is that what happens ? No. The quantization noise power in a digital signal is co...

Functioning of a second order Sigma delta modulator

Hello All, I am a newbie to the Sigma Delta domain. I am trying to simulate a sigma-delta modulator(SDM) (2 order, using 1 bit quantizer) but have been running into troubles all the way along. 1. using a straight forward implementation of the second order SDM as shown by BOSER and WOOLEY...

Re: Quantizing the filter coefficients, what are the criterias?

Greg Berchin wrote: > > Since the coefficients of a FIR filter are also its impulse response, > the set of quantization errors can be thought of as the impulse response > of an "error filter" that is added to the "true filter" to obtain the > "quantized filter". Similarly, the Discrete ...

Re: Sigma Delta Question

On Sep 16, 2:21 am, "doh ..." wrote: > The high data rate bit stream created by a one bit DSM can be filtered, decimated > and turned into a wider bit width with a simple counter. This illustrates, I think, all > the principles about which you are confused. i think the OP needs to under...

Re: Interpolated ADPCM

> > > zhoujinxi wrote: > > > I am developping an real time audio compression codec system.In my system > > I adopted 16-bit 48kHz to 5 bit 48kHz IMA-ADPCM algorithm,but could not > > get the satisfied results,audio result was not good. > > > > I know from someone,there is an tech ca...

Re: IIR filter question

Jerry Avins wrote: > Tim Wescott wrote: > > Martin Eisenberg wrote: > > > > > Joerg wrote: > > > > > > > I am no expert either but from what I have seen IIR is still > > > > used, often with some added measures to muffle their inherent > > > > instability upon signal loss. ...

Re: Mu-Law Algorithm

Tim Wescott wrote: > Mu law maps a 12 (or 13) bit number to 8 bits, so it is, perforce, a > 4096 to 256 mapping (or 4096 to 255, or 8192 to 256 -- look at the > standard). As such it is lossy, and when you go back to 12 (or 13) bits > there will be gaps -- they'll just be gaps th...

Re: mu-law companding

On Jan 23, 3:58=A0pm, "ky.nh...@gmail.com" wrote: > hi all, > I'm doing some tests with mu-law companding. I have a speech signal > with Fs =3D 16 kHz, 16-bit. I test on 2 methods to encoding in 8/4/2 > bit: > - 1st methode: using uniform quantization 256/ 16/ 4 levels. > - 2nd method...

Re: Am I missing somethin? was Re: VMSK - are claims genuine ?

"Jerry Avins" wrote in message news:402cebb7$0$3093$61fed72c@news.rcn.com... > [...] The feedback filters you cite have a great thing going > for them that ordinary FIRs lack: feedback. When their prediction is a > little bit wrong, no matter; the next estimate compensates. That's why > ...

Re: Software clock recovery for biphase demodulation

That time code does look pretty bad! It must have been a pretty low MP3 bitrate. I don't suppose there is a way to go back to the original minidisc and record it into the PC as an uncompressed .WAV file? I would suggest a multi-prong approach: some pre-filtering to boost the highs to get the...

Re: Reducing u-law noise?

Vladimir Vassilevsky writes: > Randy Yates wrote: > > Vladimir Vassilevsky writes: > > > > > [...] > > > 1. Pushing the noise to above 9.5kHz with the noise shaping filter > > > with the fixed coefficients. The frequencies above 0.9 of Nyquist are > > > very well attenuated by the ...