Wireless lossless digital audio
latentlaboratories.com
latentlaboratories.com
However, if you intend on getting one of these transmitter/receiver pairs, as one reviewer on Amazon noted[1], you'll likely run into WiFi interference problems.
[1] http://www.amazon.com/review/R13BSAFVTXO6H0/ref=cm_cr_rdp_pe...
FYI you can also get bluetooth-to-RCA modules for not much more: http://www.amazon.com/Fusion-MS-BT100-Bluetooth-Dongle-Marin...
http://www.parts-express.com/bluetooth-v40-audio-receiver-bo...
http://www.parts-express.com/parts-express-bt-1a-bluetooth-r...
http://ronja.twibright.com/about.php
As a side note, this project is interesting for another reason - its a textbook example of a failed open source project:
http://p2pfoundation.net/Ronja
.. which didn't quite hit the sweet spot for mass adoption/production, because of a commercial fork of the project. Very interesting reading both from the perspective of transferring high-rate data over long distances on the cheap, technologically, and then not actually getting off the ground and doing it on a mass level, because: politics.
That being said, the benefits of 24 bits over 16 (which I'm not convinced of in any case) would certainly not survive that setup.
There are a few caveats and I haven't looked at the transmitter/receiver pair he purchased. If it's simply 6MHz of bandwidth transmitted for some specified distance, it will probably work. SPDIF is a digital signal and while it runs at 6Mbps, there are significant harmonics that can be passed through the fiber. Since we can assume the first harmonic will pass through the link, you can expect a signal which has more sinusoid pulses coming back out of the receiver. With a PLL to recover the clock and a good data recovery circuit added to the receiver, I'm guessing he could easily push this link to 50 to 100 feet.
As an aside, the NTSC signal is actually sending far more information as a whole and while it uses an AGC to overcome amplitude changes during transmission (the sync pulse is adjusted to be -0.3V), it also uses a comb filter to recover chroma information and the luminense is AM modulated above at the pixel clock rate. Your eyes are actually pretty low resolution compared to your ears and there are lots of ways to fool the brain's image processing (e.g. interlace makes a 30FPS system look more like 60FPS).
In any case, a very creative build!
"pro audio" uses 24 bits for sample storage, so that the noise floor remains inaudible after processing has happened to each track. 16 bit output is more than adequate for monitoring.
Disappointing this pseudo science makes it on to HN.
There was a nice article here a while back that went through the entire subject in depth, can't recall the site.
This extra headroom is critical for digital systems, as they have no headroom above 0dBFS - most analogue devices distort progressively (and often rather pleasantly) when pushed too hard, but digital clipping sounds horrible. Once you've run out of bits, all hell breaks loose.
16 bit is absolutely fine for domestic audio, but it is totally inadequate in a production environment. Without the extra dynamic range provided by 24 bit, you're stuck with some kind of compromise - either you risk overs, you raise your noise floor, or you have to stick limiters in front of all your ADCs and put up with your peaks getting squashed.
I don't think I'm unique in having both?