It does this by using something called “scalable coding” which means that the file can be played at different bitrates depending on how fancy your speakers are (or if you’re listening on a crappy old phone).

Here’s an example: let’s say we have a 10-second audio clip with some sweet guitar riffs. If we use RFC-8486 to encode it, we can create multiple versions of the file that all sound pretty much the same but take up different amounts of space on your hard drive (or in the cloud).

For example:

1) The “high quality” version might have a bitrate of 320 kbps and be about 5 MB. This is perfect for listening to on your fancy headphones at home, but it’s not so great if you’re trying to stream the same song over a slow internet connection (like when you’re stuck in an airport).

2) The “medium quality” version might have a bitrate of 192 kbps and be about 3 MB. This is still pretty good, but it takes up less space on your device so you can fit more songs on there without running out of storage (or having to delete old ones).

3) The “low quality” version might have a bitrate of 64 kbps and be about 1 MB. This is the one you’d want to use if you’re streaming music over your phone or listening on some crappy earbuds (like when you’re stuck in an airport).

So basically, RFC-8486 lets us create multiple versions of our audio files that all sound pretty much the same but take up different amounts of space. This is great for people who have limited storage or are streaming music over a slow internet connection (like when you’re stuck in an airport). And it’s also really cool because we can use this technology to create “scalable” audio files that adapt to our listening environment, which means they sound better no matter what kind of device we’re using.

It’s a pretty cool technology that I think will become more popular as people start to realize how useful it can be for streaming audio over slow internet connections.