IIR audio filterbank

Started by Robert Adams in comp.dsp11 years ago 19 replies

I am looking for a way to design an audio IIR filterbank. I need to split an audio signal into roughly 1/3 octave bands with 6th-order...

I am looking for a way to design an audio IIR filterbank. I need to split an audio signal into roughly 1/3 octave bands with 6th-order or 8th-order IIR bandpass filters and then sum them back together again with a ripple of +/- 1 dB. I can do a "seat-of-the-pants" design that works well until I approach Nyquist where it kind of falls apart. I have very limited MIPs so I can't do anything very...


IFFT problem rephrased...

Started by lucy in comp.dsp17 years ago 9 replies

I want to simulate the Inverse Fourier Transform of a brickwall/rectangle function from -1500Hz to 1500Hz out of [-5000Hz, 5000Hz], the height...

I want to simulate the Inverse Fourier Transform of a brickwall/rectangle function from -1500Hz to 1500Hz out of [-5000Hz, 5000Hz], the height is 1/Fd, where Fd=3000. Theoratically I know the IFT should be sinc(Fd*t). ---------------------------------------------------- %Inverse FFT ... Fs=10; %this is the sampling rate for inverse FFT. Fn=5000; %Nyquist Rate. Fd=3000...


Relation between Bandwidth, Bit Rate and Sampling Rate

Started by Anonymous in comp.dsp16 years ago 9 replies

Hi all, Nyquist showed that minumum bandwidth, B required for baseband transmission of Rs symbols per sec without ISI is B > = Rs/2 while...

Hi all, Nyquist showed that minumum bandwidth, B required for baseband transmission of Rs symbols per sec without ISI is B > = Rs/2 while for bandpass transmission of Rs symbols per sec without is B > = Rs So does it mean that if we use BPSK to transmit 500Mbps at baseband (or passband) then the we require at least 250MHz (or 500MHz) of bandwidth? Equivalently, can we use 16-PSK to


ADC/DAC for Wimax Beamforming

Started by scc in comp.dsp14 years ago 1 reply

I'm working on the desing of an active beamforming antenna. I'm a beginner and I'm not able to explain why if I have a 21Mhz bandwidth I need an...

I'm working on the desing of an active beamforming antenna. I'm a beginner and I'm not able to explain why if I have a 21Mhz bandwidth I need an ADC of 105MSPS to digitalize the received signal. Which is the minimum sample rate and resolution of the ADC and DAC on A/D and D/A conversion of a signal of 21Mhz bandwidth and why? It should be determinated by the Nyquist freq. but...then why (for wi...


An intriguing story about undersampling

Started by erine in comp.dsp16 years ago 53 replies

Read this astonishing story at http://spazioscuola.altervista.org/UndersamplingAR/UndersamplingAR.htm Here it is an excerpt: A STORY ABOUT...

Read this astonishing story at http://spazioscuola.altervista.org/UndersamplingAR/UndersamplingAR.htm Here it is an excerpt: A STORY ABOUT UNDERSAMPLING by Angelo Ricotta - Rome, Italy a.ricotta@isac.cnr.it ITALIAN VERSION In the article "Turning Nyquist upside down by undersampling" by Bonnie Baker, EDN 12 May 2005, are reported the two formulae and to compute an allowab...


Beginner's question on undersampling and aliasing

Started by prad in comp.dsp14 years ago 36 replies

Hi all, I am a newbie to DSP. I have a huge number of data samples that I want to perform FFT on. But due to the enormous amount of the data...

Hi all, I am a newbie to DSP. I have a huge number of data samples that I want to perform FFT on. But due to the enormous amount of the data samples, I'd like to use only 1M samples and perform FFT. I know that this is severe undersampling and that it does not satisfy the Nyquist rate. But all I want is to prevent aliasing so that I can get correct information about the low frequency content...


Fourier series of sign(cos(.)) in simulation and practical mixer simulation

Started by Peter Mairhofer in comp.dsp7 years ago 38 replies

Hi, I simulate (or want to simulate) an RF system. My simulation Nyquist rate is 50GHz, the signals are 8 Msamples long and there is a...

Hi, I simulate (or want to simulate) an RF system. My simulation Nyquist rate is 50GHz, the signals are 8 Msamples long and there is a 5MHz transmission at 1GHz (using standard I/Q modulation). In any case, I am playing around with modulation. When I create a modulation sequence fc = 1GHz + 6.103515624999999kHz % note the non-integer multiple m1(t) = cos(2*pi*fc*t) then...


The sampling frequency in 802.16Rev2/D2

Started by jia in comp.dsp14 years ago 1 reply

Hi, here I don't understand the definition of the Sampling frequency in 802.16Rev2/D2 It is defined as Fs = floor(n*BW/8000)*8000 ...

Hi, here I don't understand the definition of the Sampling frequency in 802.16Rev2/D2 It is defined as Fs = floor(n*BW/8000)*8000 where, n=8/7 is the sampling factor. Q1: According to the Nyquist Sampling Theorem, the Fs should be larger than 2 times of BW. But here, Fs is similar to BW. Why? Q2: What is the usage of the function floor( ) ? Thanks for your time. -- Jia ...


Arbitrary Digital Tone generation?

Started by Shafik in comp.dsp17 years ago 13 replies

Hello folks, Suppose Im sampling at 8khz. To generate a 4khz tone, its easy, just generate alternating 1,-1,1,-1, etc... To generate a 2khz...

Hello folks, Suppose Im sampling at 8khz. To generate a 4khz tone, its easy, just generate alternating 1,-1,1,-1, etc... To generate a 2khz tone, its also easy. How can one generate a rational fraction of the nyquist freq? I want to able to generate 3.333khz while sampling at 8khz. Any ideas? This method needs to be very computationally cheap, or else its useless in this context. ...


Possible repost - Using least squares cubic spline fitting as a "filter", what is analogus to Nyquist criterion?

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

Back in September, Zevv titled a post "Isolating semi-periodic waveforms in a signal" in which he gave a link to http://tinyurl.com/pmgraph (a...

Back in September, Zevv titled a post "Isolating semi-periodic waveforms in a signal" in which he gave a link to http://tinyurl.com/pmgraph (a plot of household power usage). He was kind enough to send me a week's worth of raw data ( sampled every 2 seconds). As I am visually oriented, the first thing I did was to plot the data. I saw at least two distinct types of noise: 1. ap...


Improving ISI

Started by Randy Yates in comp.dsp8 years ago 9 replies

Hi Folks, I am working on a system that, incredibly, has a known bad TX filter that introduces ISI at the high bit rates. I know some type...

Hi Folks, I am working on a system that, incredibly, has a known bad TX filter that introduces ISI at the high bit rates. I know some type of inverse filter can be used to improve ISI, but I'm not sure which type to use: one in which the composite response is a Nyquist filter, or one in which the composite response is the (near-)ideal filter. I believe this type of distortion can als...


Normalized frequency (freqz function)

Started by Anonymous in comp.dsp16 years ago 2 replies

Lets say I have a filter with a cutoff at 100Hz and my sampling rate is 1kHz. This means my cutoff is at 0.1 in normalized frequency terms....

Lets say I have a filter with a cutoff at 100Hz and my sampling rate is 1kHz. This means my cutoff is at 0.1 in normalized frequency terms. But it seems that the matlab freqz function plots my cutoff as being at 0.2, because it defines normalized as being relative to the nyquist. Is this not really confusing? If I'm referencing the freqz graph, I guess I'll need to say that the cutoff has ...


OFDM and oversampling

Started by 6.20 in comp.dsp15 years ago 8 replies

Hi ! I would like to know if in practice, it is useful to oversample the signal at IDFT output. The sampling rate at the IDFT output is the...

Hi ! I would like to know if in practice, it is useful to oversample the signal at IDFT output. The sampling rate at the IDFT output is the Nyquist rate (2 times the frequency of the baseband signal) but I believe that it is necessary to oversample in order to perform the Root Raised Cosine Filtering. If I am right, what is the method to oversample : - Interpolation (using an IDFT size 2 times ...


Beating Nyquist?

Started by Andor in comp.dsp14 years ago 48 replies

Friends, just now I stumbled upon this webpage: http://www.edi.lv/dasp-web/ Specifically, in this...

Friends, just now I stumbled upon this webpage: http://www.edi.lv/dasp-web/ Specifically, in this chapter http://www.edi.lv/dasp-web/sec-5.htm they state that they can sample a 1.2GHz signal using a pseudo-random sampling instants with an average rate of 80MHz (in the last line of section "5.2 Aliasing, how to avoid it"). I know that for nonuniform sampling, a generalization of...


QPSK carrier frequency ??

Started by Daku in comp.dsp11 years ago 8 replies

Could some DSP guru please provide some hints to my problem - my DSP is a bit rusty ? I am experimenting with a QPSK design for which the input...

Could some DSP guru please provide some hints to my problem - my DSP is a bit rusty ? I am experimenting with a QPSK design for which the input data streams (I, Q) are synchronized with a 1 MHz clock. In this case what should be the carrier frequency ? I understand that the Nyquist criteria must hold, and so in this case what should be the carrier signal frequency ? Any hints would be valuab...


Who attended Embedded Sys Conference in San Francisco

Started by Rick Lyons in comp.dsp18 years ago 1 reply

Hi Guys, I'm wondering if any of you attended the Embedded Systems Conference (also called "electronicaUSA") in San Francisco last...

Hi Guys, I'm wondering if any of you attended the Embedded Systems Conference (also called "electronicaUSA") in San Francisco last month. I have a question about a paper that may be on the CD distributed at the conference. The lecture paper that interests me is titled: "Undersampling: Avoiding Misconceptions about the Nyquist Criterion", Paper# 467, by Teodor Neagoe. ...


How to perform efficient sampling

Started by renaudin in comp.dsp16 years ago 11 replies

Why use multiple sampling rates? The most important reason is for efficiency of calculation and/or memory. But what is the lowest sampling rate...

Why use multiple sampling rates? The most important reason is for efficiency of calculation and/or memory. But what is the lowest sampling rate possible? Well, the Nyquist theorem tells us that the sampling rate must be at least twice the bandwidth of the signal. Therefore, for maximum efficiency, you would like to do DSP processing at a rate which is a small multiple of the signal bandwidth. ...


Complex samples real samples

Started by fizzix in comp.dsp14 years ago 1 reply

I have a simple question regarding equivalent complex and real sequences. I know that by sampling in quadrature (I and Q complex samples) you...

I have a simple question regarding equivalent complex and real sequences. I know that by sampling in quadrature (I and Q complex samples) you can satisfy the Nyquist criteria at the sample rate (fs) and if I took real samples only and sampled at 2fs, then I should get equivalent power spectra in the bandwidth 0


sigma-delta

Started by kbc in comp.dsp17 years ago 8 replies

Hi, Consider a sigma delta adc which first oversamples, does noise-shaping and then filters and decimates to give final output at...

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. For this final output, note that quantisation stepsize is uniform. Now, is it possible to get this same output using a normal adc for some sampling phase and uniform sampling of the analog signal ? Or will i...


Godard algorithm in presence of Doppler

Started by shandilyasaurabh in comp.dsp10 years ago

Hi All I have implemented Godard method to estimate symbol phase and frequency offset. lower and upper bands of Nyquist shaped spectrum are...

Hi All I have implemented Godard method to estimate symbol phase and frequency offset. lower and upper bands of Nyquist shaped spectrum are used to extract phase and frequency information from the received signal. Performance in presence of Multipath and Doppler: In presence of Doppler(5-10Hz) on one of the echoes, an amplitude modulation is observed on symbol timing feedback, which crea...