ZX Spectrum: Experimenting with 1-Bit Sound
Posted by ibobev 7 hours ago
Comments
Comment by mdp2021 6 hours ago
> Agent X ( https://www.youtube.com/watch?v=T42WuUpBuHE ) or Agent X II ( https://www.youtube.com/watch?v=gNc_xczyGLc ) or Chronos ( https://www.youtube.com/watch?v=u-D24A_N4d4 ) or Raw Recruit ( https://www.youtube.com/watch?v=kl8dAVybwq8 ) or Future Games ( https://www.youtube.com/watch?v=orEXKOBIv_8 )... // A modern attempt, the ON and OFF album by Rich 'Tufty' Hollins ( https://www.youtube.com/watch?v=4nEfO4Yu7Mg )
(Repost originally for submission Furnace - the biggest multi-system chiptune tracker ever made https://news.ycombinator.com/item?id=41609254)
Comment by robviren 3 hours ago
Silver Surfer (https://m.youtube.com/results?sp=mAEA&search_query=silver+su...) Pictionary (https://m.youtube.com/results?sp=mAEA&search_query=pictionar...)
Comment by thatmf 4 hours ago
Comment by asdefghyk 5 hours ago
This mention of one bit sound made me remember the Lis’ner 1000 project of Steve Ciarca , cira 1984 , WHICH WAS speech recognition system of about 30 words. It used a GI SP1000 chip
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Comment by louprado 6 hours ago
Release party: DiHalt 2026 summer Compo: Demo Platform: zx spectrum Ranked: 2nd
Comment by codazoda 6 hours ago
Comment by MichaelRo 3 hours ago
Will post an update after I get back to my parent's home, where I have these beauties that launched me into computer science: https://imgur.com/a/GKrA0vX
Comment by khedoros1 3 hours ago
I've got an Arduino project where I did that. The output is on a single digital pin, audio encoded as PWM. It turns the output off at the beginning of an audio frame, and sets a timer to trigger an interrupt when it's time to turn the output on.
I've got a memory block of event structs, each event representing a write to NES sound hardware. I emulate enough of the features to support everything that the Legend of Zelda title theme uses. 16MHz is more than enough to decide on the sample for 2 square channels, 1 triangle, and 1 noise, outputting a 6-bit output sample at a 31250Hz sample rate. The output is only slightly scratchy; the polyphony is actually surprisingly smooth.
Comment by sehugg 2 hours ago
The NES has a 7-bit DAC, but since we're bit-banging with the CPU we only have enough time between samples for about 13 levels. Each level has its own toggle-wait-toggle-wait routine, and then we shift another bit and decide whether to branch to the next (level+1) routine or the previous (level-1) routine.
https://8bitworkshop.com/v3.12.1/?platform=apple2&file=delta...
Comment by gwbas1c 4 hours ago
Ahem, that's polyphonic sound, not multichannel sound. Multichannel sound is playback through multiple speakers.
Comment by ssl-3 2 hours ago
Questions like: How many channels can a mod file[1] have? Does that number of channels this file has change depending on whether it is being played back with monophonic or stereophonic (or some other arrangement of) speakers? What happens to the count when one or more individual samples within that mod file contain polyphonic data (chords!), themselves?
It also leads to rhetorical statements, such as: When I play a film that has a soundtrack with 8 (ie 7.1) discrete audio channels on a stereophonic pair of speakers with a sane playback chain, I'm definitely still hearing all 8 of those channels. The same happens if I extend that stereophonic system to greater number of output channels, by adding surround speakers (quadraphonic!) or whatever.
Or maybe I take one earbud out and put it back in its charging case, and through the magic of active electronics, good software, and wireless comms, the output deliberately collapses to being monophonic.
However it happens, I've still got an 8 channel of film soundtrack in my ear(s).
And of course, it also works the other way: If the electronic musical keyboard in front of me is said to support 10-voice polyphony, then that's a hard limit on the number of notes it can produce at once regardless of the number of audio output channels.
Multichannel and polyphonic are pretty bendy terms, I think. Maybe that flexibility was wrong at some point, but they've been flexible in this way for so many decades that it definitely seems right to accept the overlapping uses as being correct.
Comment by geori 6 hours ago
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Comment by fallat 6 hours ago
End result of it playing 1-bit music: https://www.youtube.com/watch?v=IjTihhFG03o
Comment by bezko 6 hours ago
Comment by adrian_b 4 hours ago
Delta-sigma modulation generates an inefficient encoding of the audio signal, with much more bits than necessary.
Converting a delta-sigma encoded bit stream to a PCM (pulse-code modulation) stream (e.g. with 24-bit or 16-bit samples at a sampling frequency of 48 kHz or 44.1 kHz) is a method of data compression.
On the other hand, delta-sigma modulation is more efficient for both analog-to-digital and digital-to-analog conversions, in the sense that for a given quality of the conversion it is much easier and much cheaper to make ADCs and DACs with delta-sigma modulation than with pulse-code modulation.
Because of this, for audio signals, normally delta-sigma ADCs and DACs are used at the analog inputs and outputs, but the delta-sigma bit stream is converted by digital filtering to PCM for data storage or for audio processing.
Comment by dreamcompiler 1 hour ago
An ADC that merely samples much faster than Nyquist but with the same number of bits per sample as PCM is not a delta-sigma converter; it's an oversampler. Delta-sigma converters always sample faster and use fewer bits per sample (usually just one bit per sample).
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