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View:  Topic list, Topic summary Topics 1 - 10 of 61531  Older »
Description: Digital Signal Processing using computers.
 

Minimum phase filter using real cepstrum 
  Hi, I'm trying to convert a mixed-phase FIR into minimum phase. From quite some googling I figured out I can either do factorization (which doesn't seem feasible) or use some form of cepstrum method. However I'm a little lost in the terms here - I found this paper: [link]... more »
By jungledmnc  - May 24 - 1 new of 1 message    

Interpolation of a Discrete Periodic Signal 
  So, this was an ongoing discussion in another thread, and it's been going on long enough that it probably deserves its own thread. I'm going to switch the notation to use 'n' in the time domain and 'k' in the frequency domain. To summarize the problem: Given a signal x[n] such that x[n+N] = x[n] for all n. x[n] may be... more »
By dszabo  - May 23 - 4 new of 4 messages    

Help with IIIR filters 
  I have a problem understanding IIR filter stability problems. The problem is this; I can't find any stability problems with IIR filters. I have a health problem that prevents me from working, so as a hobby, I write analog and digital filter design software. I have 25 years of RF design experience and I am pretty sure I know when a circuit is unstable,... more »
By FilterDan  - May 22 - 8 new of 8 messages    

Video mixing solution for H.264 streams 
  Hello i need to develop a video mixing solution for h.264 streams.I need a pointer about how to go about it.Is there any Third party toolbox in MATLAB which can take me directly into video codecs internal working and then perhaps implement a video mixing algorithm. i tried to search on internet,but it didn't help much.also please point to... more »
By sandy80  - May 22 - 2 new of 2 messages    

integer multiplication 
  (cross posted to comp.dsp and comp.lang.c.) are compilers now-a-days smart enough to multiply two N-bit numbers to a 2N-bit result without wasting instructions in the generated machine code? like, we know for C, a long times a long is a long. but if you cast one of the multiplicands to (long long), and then multiply, you get a... more »
By robert bristow-johnson  - May 19 - 13 new of 13 messages    

Call for Papers: World Congress on Engineering and Computer Science WCECS 2013 
  Call for Papers: World Congress on Engineering and Computer Science WCECS 2013 CFP: World Congress on Engineering and Computer Science WCECS 2013 Draft Paper Submission Deadline: 2 July, 2013 Camera-Ready Papers Due & Registration Deadline: 30 July, 2013 WCECS 2013: San Francisco, USA, 23-25 October, 2013... more »
By iaeng_imecs_wce_wcecs_a  - May 19 - 2 new of 2 messages    

I love my job 
  I sat down today to get in a little bit of work over the weekend on a communications project I'm doing for a customer. The channel is not only highly dispersive, but I'm stuck with a legacy modulation method that does not cough up a non-correlated mix of symbols, and at any rate the system as a whole is used for feedback where you just can't throw away a... more »
By Tim Wescott  - May 18 - 7 new of 7 messages    

The Delay of Minimum Phase FIR filter 
  Dear all, I would like to ask is there any way to have analytical estimation value of delay of minimum phase FIR filter based on the following assumptions: 1. we regard the position of largest magnitude coefficient as the delay, so for linear symmetric FIR filter the delay = (N+1)/2, where N is number of... more »
By wyonghao  - May 17 - 19 new of 19 messages    

A purely theoretical question 
  Hi group I have a purely theoretical question. When doing a FFT of an unwindowed pure sine signal, if the frequency doesn't match a bin frequency, many bins will have a magnitude > 0. Because the magnitude of each bin fits the same sinc curve, the magnitude of each bin can be "predicted" (calculated) once you know... more »
By Michael P.  - May 16 - 39 new of 39 messages    

Convolving input samples with a filter 
  Hi group I am coding a program to do some filtering on an audio file. So using the method described in Steven Smith's excellent book "The Scientist and Engineer's Guide to Digital Signal Processing", I have designed a filter with 41 coefficients and having a desired frequency response. Now I need to convolve this with a number of input samples.... more »
By Michael P.  - May 15 - 10 new of 10 messages    

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