Signal processing is an electrical engineering subfield that focuses on analysing, modifying and synthesizing signals such as sound, images and biological measurements. Electronic signal processing was first revolutionized by the MOSFET and then single-chip digital signal processor (DSP). Digital signal processing is the processing of digitized discrete-time sampled signals. Processing is done by general-purpose computers or by digital circuits such as ASICs, field-programmable gate arrays or specialized digital signal processors (DSP chips).
Hackaday has published an in interesting series of articles on signal processing, and here are some picks from it:
RTFM: ADCs And DACs
https://hackaday.com/2019/10/16/rtfm-adcs-and-dacs/
DSP Spreadsheet: IQ Diagrams
https://hackaday.com/2019/11/15/dsp-spreadsheet-iq-diagrams/
Sensor Filters For Coders
https://hackaday.com/2019/09/06/sensor-filters-for-coders/
DSP Spreadsheet: FIR Filtering
https://hackaday.com/2019/10/03/dsp-spreadsheet-fir-filtering/
Fourier Explained: [3Blue1Brown] Style!
https://hackaday.com/2019/07/13/fourier-explained-3blue1brown-style/
DSP Spreadsheet: Frequency Mixing
https://hackaday.com/2019/11/01/dsp-spreadsheet-frequency-mixing/
Spice With A Sound Card
https://hackaday.com/2019/07/03/spice-with-a-sound-card/
- check also A real-time netlist based audio circuit plugin at https://github.com/thadeuluiz/RTspice
Reverse Engineering The Sound Blaster
https://hackaday.com/2019/06/19/reverse-engineering-the-sound-blaster/
FM Signal Detection The Pulse-Counting Way
https://hackaday.com/2019/08/28/fm-signal-detection-the-pulse-counting-way/
DSP Spreadsheet: IQ Diagrams< https://hackaday.com/2019/11/15/dsp-spreadsheet-iq-diagrams/
Here is an extra, not from Hackaday, but an interesting on-line signal processing tool for generating sounds
https://z.musictools.live/#95
209 Comments
Tomi Engdahl says:
https://hackaday.com/2024/05/16/is-the-frequency-domain-a-real-place/
Tomi Engdahl says:
https://www.digitaldjtips.com/over-to-you-when-is-it-ok-to-go-into-the-red/
Tomi Engdahl says:
https://hackaday.com/2024/06/05/fourier-the-animated-series/
Tomi Engdahl says:
https://hackaday.com/2024/06/23/fixed-point-math-exposed/
Tomi Engdahl says:
Encoding sound in the cochlea: from receptor potential to afferent discharge
https://physoc.onlinelibrary.wiley.com/doi/full/10.1113/JP279189?fbclid=IwZXh0bgNhZW0CMTEAAR0wwGVES_9S9c48HD3VS52LkSU9PqdBSvuBq40Grp3F0Qqm8V1FK9-xy3A_aem_4ucA-ly_KkLCwS5q75TwKw
Tomi Engdahl says:
https://github.com/jaakkopasanen/Impulcifer
Tomi Engdahl says:
https://hackaday.com/2024/10/12/approximating-an-adc-with-successive-approximation/
Tomi Engdahl says:
Reverse engineering the circuit that revolutionized delay effects
https://www.youtube.com/watch?v=4LjP5Y1yxXs
Here’s an interesting problem: how do you create a delayed duplicate of a sound without recording it to some sort of storage medium? Back in the days before digital signal processing and cheap, abundant memory, this was a prime engineering issue. Until two Engineers named Sangster and Teer came up with a deceptively simple solution: the bucket brigade delay.
In this video, I attempt to reverse engineer the architecture of a classic BBD, recreate a bare bones version on the breadboard – and then use a proper BBD chip to design a simple DIY audio delay. If you want to build along, here’s the bill of materials:
Tomi Engdahl says:
Most people don’t actually understand what the differences are in bit rates and Sample rates
Your bit rate is an expression of the noise floor
Zero is always zero but the more bits you add the further down and the noise floor you get
So it’s 16 bit you have -96 DB of noise floor
Every bit gains you another -6 DB of noise floor
Once you get below the noise floor of the equipment the bits no longer matter and at that point it just becomes marketing
What is super useful is to use a floating point
If it isn’t floating point then the extra bits won’t matter because you have more self noise in the equipment than the bitrate you are using
So if you have a 32-bit floating point you can take advantage of every single bit in the mix down that is above the noise floor and it allows you to change the clipping point so that you can mix without having to worry about distortion and then when you did their things down you can essentially select where that noise floor ends up at
And all of this is a moot point if you don’t have software that takes advantage of the floating point to dynamically move the clipping point of your content on the Fly
In live audio it just doesn’t matter as your typical noise floor is more in the -70 range at best
The biggest reason to have a 32-bit floating point bit rate is so that when you dither the product back down to 16 bit or 24 bit that you can get rid of the noise floor created by the objects use to record and throw away those bottom eight or 16 bits of noise allowing for a cleaner and less noisy end product
Where your best and biggest differences you can hear happen are in the sample rate
And once you get above 96k there are diminishing returns but you can start to hear a smoothness and the sound with more sample rate
In the critical listening side of things once you get to 384 k sample rate it becomes indistinguishable from analog circuitry as the noise created from each sample becomes quieter than the noise of any analog circuit so you are left with only noise created from The Analog Devices connected to the converter
Tomi Engdahl says:
Why Your 16-bit CDs Might Sound Better Than 24-bit Hi-Res Streaming
https://www.headphonesty.com/2024/11/cds-offer-sound-quality-high-res-streaming/
Tomi Engdahl says:
Concise summary:
Bit Depth: Determines the noise floor in digital audio. Each bit adds -6 dB to the noise floor. For instance, 16-bit audio has a -96 dB noise floor. However, once below the noise floor of the recording equipment, extra bits have no practical effect and are mostly marketing unless using floating point.
Floating Point: A 32-bit floating point system is highly advantageous for mixing because it allows dynamic clipping point adjustments and better utilization of bits above the noise floor. It also ensures cleaner output when dithering back down to 16-bit or 24-bit, as it can eliminate noise introduced by recording equipment.
Sample Rate: Affects the smoothness of the audio. Differences are noticeable as you move above 96 kHz, but returns diminish beyond that. At 384 kHz, the sound becomes almost indistinguishable from analog due to reduced sample noise.
Practical Application: In live audio, bit depth and floating point often don’t matter much due to higher inherent noise floors (-70 dB). The real benefits of high bit depth and sample rates are more relevant in critical listening and post-production.
Tomi Engdahl says:
https://hackaday.com/2024/11/17/analog-shift-register-revealed/
Tomi Engdahl says:
Trivia: The main reason we have 48 kHz sampling (and silly multiples of it ) is that it works with 24 frames/second film. 44.1 kHz does not provide a whole number of samples per picture frame. 48 kHz of course also works well with 50 and 60 frames/s
Tomi Engdahl says:
If your DAC is 32 bit and your source is 24 bit, you can lower (in amplitude) the source/input by 8 bit/48 dB without loosing any bits. Comes in handy if your DAC hasn’t a hardware/analog volume knob.
Tomi Engdahl says:
32bit (float) is useful (kinda) for recording out in the wild, like for documentaries or maybe even movies (although it’s less helpful there since you have sound levels that can usually be anticipated and also a dedicated person for mixing and gain staging).
It basically eliminates the need to adjust the need to adjust your gain level between loud and silent takes.
In post production you can then adjust the levels without raising the noise floor (into an audible level)
Why ever you would need 32bit audio on the end-user side of production is completely beyond me.
Tomi Engdahl says:
a normal person’s “overkill” is an audiophool’s gotta have.
A standard single-precision floating point value has only a 23-bit mantissa.
The reason such things make sense is that they are so trivial to implement that smaller word size/integer implementations are pointless; chips with more-hefty circuits are trivial to fabricate, and high-precision/high-dynamic range claims are good for product marketing.
You need a good deal of precision/dynamic range when you’re processing audio (i.e., creation). For playback — not so much.
Tomi Engdahl says:
Here’s Why 32-bit/384 kHz DACs Make Little Sense
https://www.headphonesty.com/2024/01/32-bit-384-khz-dacs-make-little-sense/
Tomi Engdahl says:
https://www.edn.com/earplugs-ready-lets-make-some-noise/#google_vignette
Tomi Engdahl says:
Recording uses 32-bit *floating point*, for playback it is of course completely unnecessary.
See https://tascam.jp/int/feature/32-bit_float
Tomi Engdahl says:
The Infrasonic 17 Hz Tone Experiment Induced Revulsion And Fear In Participants
Concert-goers were subjected to unusual sounds, with unsettling results.
https://www.iflscience.com/the-infrasonic-17-hz-tone-experiment-induced-revulsion-and-fear-in-participants-77093
Tomi Engdahl says:
https://www.kw-masterointi.fi/LUFS-RMS
Tomi Engdahl says:
A beginner’s guide to power of IQ data and beauty of negative frequencies – Part 1
https://www.edn.com/part-1-a-beginners-guide-to-the-power-of-iq-data-and-beauty-of-negative-frequencies/#google_vignette
Part 1 explains the commonly used SDR IQ signal representation and negative frequencies without the complexity of math.
Part 2 (to be published) presents a device that allows you to play with and display live SDR signal spectrums with negative frequencies.
Tomi Engdahl says:
https://www.edn.com/a-beginners-guide-to-power-of-iq-data-and-beauty-of-negative-frequencies-part-2/
Tomi Engdahl says:
Analog elitism debunked: digital is actually analogue anyway
https://youtu.be/30EznMSZMBU?si=6nZP9uDIi4rOrLFn
Tomi Engdahl says:
https://hackaday.com/2025/01/30/digital-paint-mixing-has-been-greatly-improved-with-1930s-math/
Tomi Engdahl says:
It’s not just your imagination. Science proves bigger audio files do not automatically guarantee better sound quality: https://www.headphonesty.com/2025/02/320kbps-mp3s-sound-better-hi-res-files/
Tomi Engdahl says:
Freed At Last From Patents, Does Anyone Still Care About MP3?
https://hackaday.com/2025/02/08/freed-at-last-from-patents-does-anyone-still-care-about-mp3/
The MP3 file format was always encumbered with patents, but as of 2017, the last patent finally expired. Although the format became synonymous with the digital music revolution that started in the late 90s, as an audio compression format there is an argument to be made that it has long since been superseded by better formats and other changes. [Ibrahim Diallo] makes that very argument in a recent blog post. In a world with super fast Internet speeds and the abstracting away of music formats behind streaming services, few people still care about MP3.
The last patents for the MP3 format expired in 2012 in the EU and 2017 in the US, ending many years of incessant legal sniping. For those of us learning of the wonders of MP3 back around ’98 through services like Napster or Limewire, MP3s meant downloading music on 56k dialup in a matter of minutes to hours rather than days to weeks with WAV, and with generally better quality than Microsoft’s WMA format at lower bitrates. When portable media players came onto the scene, they were called ‘MP3 players’, a name that stuck around.
But is MP3 really obsolete and best forgotten in the dustbin of history at this point? Would anyone care if computers dropped support for MP3 tomorrow?
You Didn’t Notice MP3 Is Now Free
Why nobody cares
https://idiallo.com/blog/listen-mp3-is-free
The MP3 format, once the gold standard for digital audio files, is now free. The licensing and patents on MP3 encoders have expired, meaning you can now include them in your applications without paying royalties. For software developers and audio enthusiasts, this might seem like a big deal. But, surprisingly, almost no one noticed. Why? Because the world of technology has changed so drastically that MP3′s significance has faded into the background.
I noticed the change because of my habit of downloading Audacity, the open-source audio editing software. For years, Audacity required users to download an external MP3 encoder, like LAME, because the MP3 format was proprietary. This extra step was a constant reminder of the legal and technical restrictions surrounding MP3. But now, that step is unnecessary. And yet, no one seems to care.
The reality is that MP3, while still relevant in certain niche areas, has largely been eclipsed by a combination of faster internet speeds, changing software paradigms, and a shift away from dealing with files altogether.
Tomi Engdahl says:
https://www.edn.com/converting-from-average-to-rms-values/
Tomi Engdahl says:
https://www.edn.com/basic-oscilloscope-operation/
Tomi Engdahl says:
https://www.headphonesty.com/2024/01/32-bit-384-khz-dacs-make-little-sense/
Tomi Engdahl says:
Tiny data-over-sound library
https://github.com/ggerganov/ggwave
Tomi Engdahl says:
https://hackaday.com/2022/07/06/ggwave-sings-the-songs-of-your-data/
Tomi Engdahl says:
https://github.com/ggerganov/ggwave-arduino
Tomi Engdahl says:
New Chirp tool uses audio tones to transfer data between devices
https://www.bleepingcomputer.com/news/software/new-chirp-tool-uses-audio-tones-to-transfer-data-between-devices/
A new open-source tool named ‘Chirp’ transmits data, such as text messages, between computers (and smartphones) through different audio tones.
The tool, developed by cybersecurity researcher solst/ICE, maps each character into a specific sound frequency and plays it along with real-time visualization.
Other microphone-equipped computers running Chirp may capture the sound and translate the message back into text.
The project allows users to “sneak” messages between devices in a fun way, and it’s available both online and as a standalone app available for free through GitHub.
https://github.com/solst-ice/chirp
Tomi Engdahl says:
https://hackaday.com/2025/03/10/satellite-imagery-you-can-play-with/
Tomi Engdahl says:
Formula
Creativity unleashed
VST3, AU, Standalone | Windows, macOS, Linux | Free
https://soundspear.com/product/formula
Tomi Engdahl says:
Putting Audio Effects on Text
https://m.youtube.com/watch?v=47jlny15IEc
Tomi Engdahl says:
cursed EQs
https://m.youtube.com/watch?v=gNhRXtBvkfs
Tomi Engdahl says:
https://hackaday.com/2025/04/10/using-integer-addition-to-approximate-float-multiplication/
Tomi Engdahl says:
https://www.edn.com/single-sideband-generation/#google_vignette
Obeydul says:
What is Transient Response in Audio?
https://obeydulhaque.com/
https://en.wikipedia.org/wiki/Transient_(acoustics)
https://majormixing.com/what-are-transients/
Tomi Engdahl says:
https://hackaday.com/2025/06/04/passive-saturation-box-is-a-cheap-way-to-distort-your-sound/
Tomi Engdahl says:
https://hackaday.com/2025/06/12/learning-the-basics-of-astrophotography-editing/
Tomi Engdahl says:
A simple to use implementation of networking across ultrasonic frequencies, by making use of Gnuradio and a microphone and speakers. This allows you to use TCP/IP,UDP across an audio link.
https://www.anfractuosity.com/projects/ultrasound-networking/
#2600net #irc #hackers #hamradio #ultrasound #networking #2600
Tomi Engdahl says:
https://www.facebook.com/share/17ESgmSAUk/
A German engineer was tasked with a goal that sounded impossible: shrink a digital music file by nearly 95% without ruining the sound.
This engineer was Karlheinz Brandenburg. He and his team at the Fraunhofer Institute in Germany took on the challenge in the late 1980s.
The problem was that high-quality digital audio files were massive, making them impractical to store or send over the internet at the time.
Their solution was based on psychoacoustics, the study of how we perceive sound. They realized the human ear can’t hear every single frequency in a recording.
The algorithm they created cleverly removes the parts of the audio that are inaudible to most people, dramatically reducing the file size.
This new format was standardized in 1991 and officially named MPEG-1 Audio Layer III. We know it today as the MP3.
The file extension “.mp3″ was officially adopted in 1995, and it completely changed the world.
Suddenly, you could store hundreds of songs on a device that fit in your pocket. The age of digital music and the iPod was born directly from this breakthrough.
Brandenburg’s work paved the way for the entire digital music revolution, fundamentally changing how we buy, store, and listen to music forever
Sources: Fraunhofer Institute, The Conversation#MP3Revolution #DigitalMusic #InnovationInAudio
Tomi Engdahl says:
The success of the MP3 wasn’t guaranteed from the start.
In the early 1990s, the new format faced major resistance from established industry giants.
Companies like Philips were heavily promoting their own format, called MP2, and didn’t see the need for the more aggressive compression of the MP3.
Despite the lack of initial industry support, Brandenburg and the Fraunhofer team persevered.
They released the first software to create MP3 files in 1994, making it accessible to the public and programmers.
This decision to release the encoder software allowed the format to grow organically, eventually becoming the standard that everyone knows today.
Tomi Engdahl says:
The History Archives Sony insisted that the compression they used for their Mini Disc (ATRAC) was superior to mp3, at a higher bitrate however. I believe late MD devices could also playback mp3.
Tomi Engdahl says:
Colin Mee That’s not always the case. While it’s true that music today is more digital than ever, it’s important to consider how it is ultimately rendered. A significant portion of the audio content remains analog in origin—such as vocals and acoustic instruments. Even digitally generated elements are typically recorded and mixed in high-resolution formats during mastering, far exceeding the resolution delivered through consumer digital platforms.
Given that human hearing is analog by nature, the final perception of digitally distributed music is the result of multiple stages of transformation: analog sources are digitized, all components are typically compressed, and ultimately reconverted to analog through a DAC. Each of these stages introduces some degree of loss or alteration relative to the artist’s original intent.
Vinyl, by contrast, involves fewer conversion stages. As a fully analog medium, it preserves a more direct and continuous signal path from the master to playback. This can result in a final listening experience that more accurately reflects the artistic intention, regardless of whether the original production was analog or digital.
Tomi Engdahl says:
Wade Sun Yep, most radio stations will not use MP3. They demand 44.1K 16 bit sample rate. I can tell the difference.
Especially when the signal keeps getting compressed and processed by other steps in the studio and transmitter chain. By the time you have compressed it, converted it to VOIP, with jitter and latencies, you get a noticablly reduced audio quality. Stick with the real thing at the head end, and what you get out of the back end will sound much better.
Tomi Engdahl says:
The programmers from Zagreb made a significant contribution to the development of the MP3 format through their work on AMP (Advanced Multimedia Products) software, which became a key part of the popular Winamp and other MP3 players in the late 1990s.
1. The company PlayMedia Systems from Zagreb developed the AMP decoder, one of the first and most efficient software decoders for the MP3 format.
2. AMP became the core decoder for playing MP3 files in Winamp, the most popular music player of that era.
3. By doing so, Croatian programmers enabled the widespread availability and adoption of the MP3 format globally – as Winamp was crucial for popularizing digital music.
4. Their decoder was also used by other major companies, including Microsoft (in some versions of Windows Media Player).
Why is this important?
Without a fast and reliable software decoder like AMP, the MP3 format would not have gained popularity among users as quickly in the late 1990s.
The Zagreb team were pioneers in optimizing decoding for the relatively slow processors of the time.
So, although they did not invent the MP3 format itself (it was developed by the Fraunhofer Institute in Germany), their software was crucial for the mass adoption of the format among end users.