What throughput can you get with audible sound?
It would take a sophisticated modulation scheme, like a modem. The sound would be similar to white noise. Software that's getting 20bit/s or whatever is using old-school tone-based modulation, but it is quite robust in the presence of other sounds.
That assumes:
- Frequency response ~20kHz.
- Audio ADC/DAC sample rates configurable well in excess of the Nuqyist limit of the frequency response range of the speakers and mics.
- Good signal to noise (~90dB), which equates to ~15 bits at max volume range.
- Not playing at max volume, but a reasonable level ~18dB down, so ~12 bits.
- A very quiet environment, or one where the background sound is very predictable.
- Stereo laptop speakers and stereo mics, to make a 2x2 MIMO spatially modulated channel.
- Good channel separation (~80dB).
- Great linearity, which might be optimistic.
Thanks!
Interesting...
If white noise “exists” and is constantly being received does it have an energetic value? Hmm.
Interesting if there is a particular reason our bodies converged to this state because of a link to sound and resiliency to information communication. I need to think on that one.
It's not a coincidence that human range is similar, as the Mac was designed for humans.
In theory minimodem could be put into web assembly right?
Cool stuff to understand. Thanks. Now I must learn more.
Also, weird: https://en.m.wikipedia.org/wiki/Parametric_array
How do I reason about the range of frequencies that air itself has a threshold of vibration? There’s a particular range within that type of matter itself I’d imagine. Sound is only a human description of a waveform perturbation within the air... I’d imagine if the air vibrates too much it could explode or some other effects.
If in audible range a faster song would have more data capability - techno vs hip hop? Ha.
It also brings up an interesting thought I never had before. With early home computers, and I assume before that as well, you would place a phone handset onto a coupler to receive data. It is odd to think now that there is a little air in there that is transferring sound between the handset speaker and the coupler microphone. I assume that without that little bit of air it wouldn't function.
Also if air was such an issue then the dynamics of a song being played live would be of series inconsistent patterns leading to error in rate based on environmental factors. Rate would be whatever air density allows but whether you received that rate is different.
Song recorded vs song played vs song heard when recorded vs song heard when played over speakers.
The waveform of both should be highly similar in a way that wouldn’t be considered error besides amplitude or echoes from acoustics.
So it’s a distributed message sensing problem?
In my perspective I just saw the waveform of a techno song == the rate of data being communicated.
120 bpm is 120 bpm. Loss over distance is it’s own factor based on ability to ack.
Physical waves depend on a fluid?
EM seems to be affected by a medium/fluid.
Wonder where the distinctions are more poignant. Maybe degree of affects?
I've used it on a few occasions to send out a text email or some other kind of small text document. Sending out something like a several megabyte image would be very slow.
The analogue phone line signal is digitised at the exchange, and the digital channel is explicitly 64kbit/s.
Modems can only do "56k" by co-operating with the digital system; one of the modems actually has a digital ISDN connection.
Without that ISDN connection, analogue modems reach up to 33.6k. Theoretically they can do more (but never more than 64k), but in practice that was the last standard produced prior to the 56k, semi-analogue-semi-digital standard.