Im doing the pink noise FFT. I calculate the frequency of 1kHz by adding the contribution of the corresponding bins. Is it possible that sometimes...

Im doing the pink noise FFT. I calculate the frequency of 1kHz by adding the contribution of the corresponding bins. Is it possible that sometimes the result is zero? Thanks.

I had some code computing the second difference of a sampled signal as (x(n-1)-2x(n)+x(n+1))/(square of interval) I only needed a crude estimate...

I had some code computing the second difference of a sampled signal as (x(n-1)-2x(n)+x(n+1))/(square of interval) I only needed a crude estimate of the second derivative in this case. Then I came across "five point stencils" By this method, the second difference is (-x(n-2)+16x(n-1)-30n(n)+16x(n+1)-x(n+2))/12/(square of interval) And then 5-sample Savitsky-Golay formula, which seems rath...

I'm confused by some code I wrote to measure signal amplitude from a CODEC. The FPGA sampled for some period of time squaring the samples and...

I'm confused by some code I wrote to measure signal amplitude from a CODEC. The FPGA sampled for some period of time squaring the samples and summing them up. I did not take the square root of the result, at least not in the FPGA. I would expect this result to be proportional to the power in the signal. When I compare this number at multiple frequencies to get the variations,

Hello, I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its...

Hello, I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its amplitude. Without input signal I get FFT values ​​close to 30dB. With maximum input values ​​(overflow) I get values ​​of 90dB, with a scale of 0 to 120dB. Why not get a dynamic range of 100-110dB as is logical

Gentle Readers, Ok, I've been out of DSP for way too long so maybe this is a stupid question, but in the interest of getting my feet wet...

Gentle Readers, Ok, I've been out of DSP for way too long so maybe this is a stupid question, but in the interest of getting my feet wet again, here goes. Let's say we have a tone which we want to detect. Which is going to give better performance, correlation followed by a threshold or a simple bandpass filter followed by an energy detector and threshold? -- Randy Yates, DSP/Embedded ...

A client for unknown reasons wants to emit speech in teh frequency range above 20kHz. I was wondering what is the best way around this to do a...

A client for unknown reasons wants to emit speech in teh frequency range above 20kHz. I was wondering what is the best way around this to do a frequency shift. Then presumably I would need a sound card on a PC with high sampling rate and ultrasonic amplifier and loudspeaker. (this can be bought). So I sample at say 44.1KHz as usual a section of speech and store to wav file. Then I

Recently I attended a coding seminar. The speaker very briefly reviewed the relevant history; first Hamming codes, ~1960: BCH codes, ~1970:...

Recently I attended a coding seminar. The speaker very briefly reviewed the relevant history; first Hamming codes, ~1960: BCH codes, ~1970: convolutional, 1995: polar, 2002: LDPC Clue me in - what was the advance in each case? I worked on a BCH project once, that seemed fairly efficient, how is it, or other algorithms, deficient? -- Rich

A few of us had the great fortune to work with Elwyn Berlekamp during our careers. He passed away on April 9, 2019. The most prolific and...

A few of us had the great fortune to work with Elwyn Berlekamp during our careers. He passed away on April 9, 2019. The most prolific and most genius expert on algebraic coding, Berlekamp also contributed significantly to game theory, cryptology, and quant. He was a member of the USA National Academies of both Engineering and Science, recipient of both the Shannon Award and the Hamming M...

Hi, the input signal is sin(t), I multiply it with cos(t) to get the I component and sin(t) to get the Q component. So...

Hi, the input signal is sin(t), I multiply it with cos(t) to get the I component and sin(t) to get the Q component. So then I(t)=sin(t)*cos(t)=0.5*sin(2t) Q(t)=sin(t)^2 I would expect only the DC component to persist, but instead I have two quickly wiggling waves. What is missing to shift 1kHz to DC without introducing the mirror frequencies? Thanks in advance!

How to record multiple PDM mics at 16KHz? What is a recommended soundcard/HW that can be used?

How to record multiple PDM mics at 16KHz? What is a recommended soundcard/HW that can be used?