## Forums comp.dsp

## Low memory footprint decimation

Hello, so I finally have some time to return to the problem of the multichannel decimation on PSOC5LP. The situation is as follows: there are...

Hello, so I finally have some time to return to the problem of the multichannel decimation on PSOC5LP. The situation is as follows: there are 8 channels of 12 bits@100kHz each and a single digital quadrature mixer running at 310kHz, also 12 bits. The hardware is an 80MHz ARM CortexM3 equipped with a coprocessor called DFB, running at the same speed, with single-cycle 24x24-> 48-bit MAC an

## Finding maximum of sinc in 0..1

inHi folks, the audio world is sort of obsessed with sort of "true peak level". Unfortunately it is rather tricky to calculate, so it is...

Hi folks, the audio world is sort of obsessed with sort of "true peak level". Unfortunately it is rather tricky to calculate, so it is calculated by oversampling to about 192kHz and taking the normal peak level. Sadly that's really not true peak level. So how about the (nearly) correct value - if we take say 2 * N + 1 samples around each sample S[0] (hence S[-N] .. S[N]), we can calculate ...

## How can a filter impulse response be interpolated?

inHi, I know how to interpolate a digital signal. It is first interpolated by inserting 0's. For example, one can add 4 0's to each data for a...

Hi, I know how to interpolate a digital signal. It is first interpolated by inserting 0's. For example, one can add 4 0's to each data for a 5 times interpolation. Then, a low pass filtering to eliminate the aliasing frequency. Now, I have a low pass filter from 0 to 10 MHz pass band with a sampling rate of 40 MSPS. I want to get the same 0 to 10 MHz response (it is not a flat pass band...

## Why is there a spurious component in this resampling process?

inHi, Thank all of you for replying to my previous post on filter impulse response. One of the replies said to directly calculate the fir coef...

Hi, Thank all of you for replying to my previous post on filter impulse response. One of the replies said to directly calculate the fir coef as the original filter. I considered this method. The only problem is too time consuming. As it is not a general FIR design, I have to find the background information to write the code from the bottom. OK. I have solved the problem with all of your in...

## 6dB/oct butterworth crossover doesn't have flat response

inHi folks, sorry if I'll be asking a dumb question :). I'd like to build a 6dB/oct crossover (with variable number of bands and crossover...

Hi folks, sorry if I'll be asking a dumb question :). I'd like to build a 6dB/oct crossover (with variable number of bands and crossover points). First order butterworth LP and HP filters seem to be working fine and provide nearly flat magnitude response IF the crossover point isn't high enough, hence close to the nyquist, then it starts forming something that looks like a high-shelf of +6dB. ...

## Kalman filter estimator for Gyro and accelerometer

inI am using a fairly standard approach to estimating angular pitch using a KF. It uses both accelerometer and Gyro angle data. Now it estimated the...

I am using a fairly standard approach to estimating angular pitch using a KF. It uses both accelerometer and Gyro angle data. Now it estimated the angle fine enough and I implement the steady-state KF. Never tried this before but then put a PID or lag-lead controller on this measurement. I find that the Kalman filter bandwidth is stuff all and severely reduces the bandwidth of my cl

## spectral accumulation using phase vocoder

inHi! I've made a simple freeze effect with a phase vocoder that can 'freeze' the sound by repeatedly converting the same spectral frame of that...

Hi! I've made a simple freeze effect with a phase vocoder that can 'freeze' the sound by repeatedly converting the same spectral frame of that sound to the time domain, incrementing the phases each time with the phase difference calculated from that frame and the previous frame. This works fine. What I would like to do though, is take another freeze frame out the incoming sound a

## Sensitivity function - control theory

inI watched this video about the sensitivity function, highly instructive, but at around 9 minutes, the author says that the maximum of the...

I watched this video about the sensitivity function, highly instructive, but at around 9 minutes, the author says that the maximum of the sensitivity function should be between 1.3 and 2. https://www.youtube.com/watch?v=BAWdZvF1O40 What's wrong with having a sensitivity less than 1.3 ? Regards

## Sine Wave autocorrelation, interpolation of phase

inHallo. For the purpose of measuring complex impedances i need to compare the phase of two copies of a sinewave over a number of periods. The...

Hallo. For the purpose of measuring complex impedances i need to compare the phase of two copies of a sinewave over a number of periods. The sinewave is generated in the same µC. Frequency is known and stable. The original and the shifted signals are sampled by a double synchr. ADC. The phase shift is the base for calculation of the complex Z of a load. I have implemented an (auto-)correlatio...

## Hah! Why 5-lug wheels balance

inQuite some time ago I handed y'all a quandary, to wit, proving that sum_{\theta} cos(\theta) = 0, when \theta is evenly distributed on a...

Quite some time ago I handed y'all a quandary, to wit, proving that sum_{\theta} cos(\theta) = 0, when \theta is evenly distributed on a circle and there are an odd number of them. (It's even if there's an even number of them -- you've got this nice pairing of cos(this) + cos(-this) = 0, and the proof is a few lines.) So, I ran across this page:

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