Greetings. I have been using neural networks and other machine learning tools for sometime time. Yesterday the following question popped up in...

Greetings. I have been using neural networks and other machine learning tools for sometime time. Yesterday the following question popped up in my mind however: Why do we use machine learning tools when we could achive similar results with plain interpolation? Let's assume a noise free regression scenario (not classification and no measurement errors). In the case of infinite samples and...

I'd like to implement a first-order delta-sigma power amplifier in Verilog. The input stream is N=16-bit wide (signed). How many bits should the...

I'd like to implement a first-order delta-sigma power amplifier in Verilog. The input stream is N=16-bit wide (signed). How many bits should the integrator have? Common sense says it would be enough for the worst-case delta (=N+1) + the actual content (also N+1), so N+2 bits. Is it correct? Can it be done with just N? Best regards, Piotr

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

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...

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 ...

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...

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...

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...

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...

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...

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. ...

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...

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

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...

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

I 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