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Both of these are excellent pieces so I very much appreciate the effort. I would guess that you made this primarily for the early build up into the point between about 0:46 and 1:10 which I find to be exquisite and worth the effort on its own, well done. In the rest, however, I don't hear much synergy. Aerith's theme is very melodic and Vangelis is very atmospheric which I guess is what makes this difficult. The period after 1:10 where Aerith's theme becomes more playful especially with the quicker notes you've added changes the tone a lot and also seems to muddle the meaning of the previous Vangelis inspired part because it is so different. I wish I could offer you more constructive advice, but all I can suggest is that you could try to drastically slow down or even change Aerith's theme after 1:10 so that it works more with the Vangelis tone you have set in the first 1:10. Good luck!

I'm 4fast5u guys. Entry submitted.

If you like chiptunes, I don't see how you could possibly dislike this! So good!

What a freaking epic way to end this competition. Both of these tracks were incredible, and they couldn't be more different in their presentations. Well done everyone. Well done.

So I should amend my statement to be more technically accurate: Sonarworks can not remove reflections from the room, they are still bouncing around, and no amount of DSP can just stop them from propagating. However, the effect is "cancelled" at the exact measured listening position. Sonarworks is an FIR approach, which is another name for convolution style filtering. Deconvolving reflections is totally and absolutely in the wheelhouse of FIR filtering, as reverb is "linear" and "time-invariant" at a fixed listening position, (relatively) fixed monitoring level and fixed positions of objects and materials in the room. So it absolutely necessitates re-running calibration process if you change stuff around in the room, change the gain structure of your system output, etc I can't comment on standing waves but it seems in their paper they noted that it wasn't covered by the filter approach and so they recommended treatment for that. Just from my peanut gallery background in studying EE I think it makes sense that standing waves aren't linear and time invariant and so trying to reverse them through a filter wouldn't go well. Same goes for nulls, if a band is just dying at your sitting position, trying to reverse that via filter is just not smart at all. Regardless, if you were to move your head or walk around, you would again clearly notice how horrible the room sounds (though the fixed general frequency response is still an improvement), because now you've violated the "math assumption", introducing sound difference created by changing your spatial position. This is the disadvantage of relying on DSP calibration (along with latency and slight pre-ring for the linear phase) and is a compelling reason why you wouldn't want to choose it over proper acoustic design in a more commercial/professional studio design (you don't want crap sound for the people sitting next to you in a session!). I think it's a pretty decent trade for home producers and produces much better results than trying to put cheap speakers in a minimally treated room and still having to learn how to compensate for issues. I just see it as more expensive and time-consuming. Compensating isn't fun; its easy on headphones where problems are usually broad, general tonal shifts in frequency ranges. But in a room, and this is shown in the measurement curve, the differences are not broad and predictable, they're pretty random and localized in small bands. In my opinion it's difficult to really build a mental compensation map unless you listen to a metric ton of different sounding music in your room. It is traditional to learn your setup, but I think the tech is there to make the process way simpler nowadays. To be scientifically thorough, I would love to run a measurement test and show the "after" curve of my setup, however sadly I don't think it's really possible, because SW has a stingy requirement that for measurement the I/O for the computer has to be running on the same audio interface and the calibrated system output is a different virtual out, so there's no way I could run the existing calibration and then also measure that in series. All I can do is volunteer my personal anecdotal experience at how it has improved the sound. I'm not trying to literally sell it to you guys, and no, I don't get kickback, I just think it's one of the best investments people should make into their audio before saving up to buy expensive plugins or anything else. Especially because its results are relatively transferrable to any new environment without spending any more money no matter how many times you move, where room treatments would have to be re-done and maybe more money spent depending on the circumstance. And because of the topic of this thread, it shouldn't be understated that SW calibration can drastically improve the viability of using cheaper sound systems to do professional audio work. I've run calibration at my friend's house with incredibly shitty, tiny $100 M-Audio speakers, in just about the worst way to possibly place/orient them, and I'd say the end result really was within the ballpark of sound quality I get at my home room with more expensive monitors and a more symmetrical set up. It wasn't the same, but it was a lot more accurate (sans any decent sub response) than you could roll your eyes at. Stereo field fixing is dope too. @Master Mi I'm not sure what's to be accomplished by linking YouTube videos of the sound of other monitors. They're all being colored by whatever you're watching the YouTube video on. At best, a "flat response" speaker will sound as bad as the speakers you're using to watch the video, and furthermore, a speaker set that has opposite problems that yours do will sound flat, when they aren't flat at all. Listening to recordings of other sound systems is just about the worst possible way to tell what they sound like.