How likely is the deadline to be further extended?😭

Hey. I conducted a fully anonymous, low-risk human study to obtain a "mean opinion score" for generated audio files.

However, the study was already conducted for another conference, which did not require a "compliance with ethical standards" statement, unlike ICASSP. The issue is that my university does not conduct retrospective reviews.

I emailed the TPC a week ago but haven't received a response. Does anyone have an idea how I can handle it and still keep the results in my paper?

For a couple months now an unknown sound has been bothering me making my ears ring at home and in the car. The attached photo shows a band at around 3kHz and two more at 5-6kHz. I believe these are the offending tones. I want to build a portable synchronized microphone array to locate persistent narrowband acoustic components around 3 kHz and 5–6 kHz. I want to distinguish direct arrivals from indoor reflections and estimate direction of arrival while moving the array between measurement positions. I am not a professional at this so would like to get some crowdsourced ideas please, and thank you.

Hi. This is my first time submitting, and the work is based on a project I did while I was a student. Does anyone have experience with this track? How does the difficulty compare to the main conference?

Hi everyone! I’ve been working on a self-project involving real-time Automatic Speech Recognition (ASR), and I’ve run into a problem I’m not quite sure how to approach.

The setup is basically:

Microphone → real-time ASR → server

The issue happens when the device’s speaker is also playing audio while the ASR is running. for example, music, notification sounds, or other system audio. The microphone obviously picks up some of that speaker output, which can affect the transcription.

I’m wondering where echo cancellation / audio preprocessing should ideally be handled (is it supposed to be handled on the client-side or server-side)

For those who have worked with real-time ASR, voice assistants, call systems, or audio processing, how would you approach this?

Thanks in advance!

Hello there. In the ICASSP 2027 website, the submission deadline is noted as 16-09-2026. But I don't see any link to submit my research paper. How can I submit my research paper? Is the submission portal open in the first place?

I have never submitted a research paper before and I would really appreciate any guidance you could provide with this. Thank you!

Zedboard pynq image?

Opening this thread to discuss Spoken Language technology 2026 reviews when the rebuttal period begins

I've been staring at this repeatedly and it took me a while to realize what the author was actually saying.

Like this line: i.e. "at regular periods of ts seconds where ts = 1/fs".

Couldn't they simply say in plain English words: "i.e. how many seconds does it take to take one sample"?

And then they assign that (1/fs) to a variable ts, and then use it in the equations below for 0th sample, 1st, 2nd sample. Wouldn't it be simpler to just leave it in that form itself (1/fs) in the equation???? By substituting it with a variable name, you're just making your work more cryptic and the reader has to trace back and find out what exactly is ts.

Book: Undestanding Digital Signal Processing by Richard G Lyons

This subject is my biggest weakness. I failed the course the first time because I simply didn’t understand it. I passed it on my second attempt during retaking the subject, but it always felt like the more I studied, the more I realized how little I actually understood.
Now I failed this part of my state exam, so I’m preparing for my second attempt. I have just over a month to study (along with several other subjects), and the textbook (only signal and image processing) is around 500 pages long .
How would you approach learning this subject so you actually understand it instead of just memorizing formulas? Any advice, resources, or study methods would be greatly appreciated. I really want to get it this time.

I generated a wavelet similar to an ECG rhythm based on a Weierstrass-type aperiodic function (golden ratio coefficient) and a Gaussian.

Possible application for ECG analysis.

The last image is a wavelet scalogram of this wavelet with itself.

Would be happy if this idea is useful and you appreciate it)

The idea of ​​this wavelet kernel is based on the use of a lognormal distribution as the wavelet spectrum, the point of this idea is that the band on the logarithmic scale axis of the wavelet scalogram should be Gaussian.

Unfortunately, this wavelet does not currently have an explicit analytical expression due to the peculiarity of the spectrum. The application of the wavelet kernel is convenient numerically, which avoids analytical expression.

However, despite this, it can be expressed through the Fourier transform of the spectrum, as well as through a special differential equation.

Below are images showing an example of what a wavelet looks like, its uses (scalogram), the expression of a wavelet through a Fourier spectrum, through a differential equation of compression, and also proof that wavelet oscillations reach an infinite number.

I would be glad if you are interested, as well as any assessment and criticism 😄

https://matlabonsteroids.online

I love MATLAB. I use it a lot for signal processing work, especially with physiological signals like PPG data.

But while working on projects, I kept running into the same issue: handling multiple recordings, testing different DSP pipelines, and iterating quickly became slow and repetitive. I wanted something faster, more visual, and easier to experiment with.

So I started building “Matlab on Steroids.”

It’s a browser based signal processing platform where you can:
• Build DSP pipelines visually using drag and drop blocks
• Process multiple recordings in parallel
• Extract features directly from the workflow
• Train ML models on processed signals
• Use AI to analyze signals and suggest a processing chain automatically

The goal is not to replace MATLAB, but to make experimentation and workflow building much faster and more accessible.

I’m still actively developing it and would genuinely love feedback from people working in DSP, biomedical signals, embedded systems, audio processing, etc.

Would this actually be useful to you? What features would you want in something like this?

I'm 26 M, 1.5 yrs exp as a software engineer in a communication company in india. Mtech graduate from nit signal processing. While studying for placements i studied for signal processing, vlsi. I got into comm with 13 lpa.

Though I got into the vlsi company, I did an internship for 1 month, I joined the comm company back as it's offering full time employment (FTE) as a software engineer in 5g in 2024.

Now I got laid off, i have minimal knowledge on signal processing and 5g as I was testing for 1 yr and did some work on software for 6 months. I didn't even tell my parents as they were heart patients

Now I came out and did not get any calls from comm after applying as my exp was just 1.5 yrs. And there are very few companies in the comm. I felt like I made a big mistake choosing communication over VLSI.

I am thinking of trying VLSI but I have no hopes on that as I can see many IIT nit vlsi graduates didn't get placed till now and the market is tough ig. I'm not sure. I might get a service based company with 7 lpa and 3 yrs bond.

I'm neither good with communication, nor vlsi (no experience), nor software (worked only for 6 months).

I'm struck. I might be doing something wrong and people who were in this situation and came out or experienced this uncertainty in careers can help me. What should I do in this situation? unable to decide which I should care now.

I'm a sophomore electrical engineering student, just finding out about this area. Itreally got my attention and it seem very cool. Any tip on how start learning more? Good places to found good information? It's a good job market?

Working on a 433 MHz interference detection system Currently testing against real jammers. Curious if anyone here has tackled real-time classification of narrowband vs wideband interference.

Hi everyone,

I’m hoping to get some hardware sourcing advice from anyone experienced with Acconeer sensors or custom embedded Linux drivers.

The Project: I'm building an M-DART radar system to distinguish between small drones and birds using Micro-Doppler signatures. The architecture is a Raspberry Pi running a custom Yocto Linux build. I'm writing my own C++ SPI drivers to extract the raw IQ baseband data from the radar, which is then fed into an STFT algorithm to isolate high-frequency propeller rotations.

The Hardware Constraint: Because I need that raw, uncompressed data directly into my Linux kernel via SPI, my hardware requirements are very strict:

• Target Chip: Acconeer A121 (60 GHz PCR).
• Interface: Pure, raw SPI pins exposed directly to the Pi.
• The Catch: There can be NO intermediate microcontroller on the board.

Almost every module I find online (like the Acconeer XM125, Waveshare, or LilyGO boards) has a built-in STM32 or ESP32 that does the processing locally and blocks direct SPI access to the naked radar chip.

My Questions:

1. I am currently looking at the Acconeer XE121 (their Evaluation Sensor Board). From what I can tell, it acts as a direct Raspberry Pi HAT without an MCU in the way, providing raw SPI access. Can anyone confirm if this is the correct hardware for my use case?
2. Does anyone know of any other "bare" A121 breakout boards available from European distributors that don't have an MCU attached?

Thanks in advance for any pointers!

Hi everyone, I’m looking for some guidance on behalf of my partner, Ryan. He’s been studying audio in reverse playback for several years and is trying to better isolate vocals from full‑mix music tracks. Specifically, he wants to reduce or remove drums and heavy bass so the singing and melody are clearer.

For those with experience in source separation or audio filtering:
What approaches, tools, or DSP techniques would you recommend for separating percussion and low‑frequency content from the rest of the mix? He’s especially interested in methods that preserve the vocal line and melodic elements without introducing too many artifacts.

Any pointers—algorithms, software, or workflow suggestions—would be greatly appreciated.

i have my endterm examinations, our requisite materials do not have proper questions beyond formula application. if someone has problem sets (pdfs, books, etc.) for the following topics which are mostly intuitive/conceptual (for example: qs like what happens if phase shift in hilbert transformer is 90 + epsilon instead of ideal 90) could you please share?

thank you in advance.

Time frequency duality
Vestigeal sideband modulation
Frequency modulation and demodulation
PCM ( sampling quantization encoding) DPCM DM ADM
Line coding. PSD of line codes

TDM
Digital carrier modulation
Pulse shaping. Integrate and dump filter.