This afternoon I woke up from a slumber, in my dark quiet room, and started listening to some music.
I was listening for a while, and then thought "this sounds too good, what's wrong?"
My current 'listening studio' is simplistic, here's a photo of it:
Music on laptop -> USB -> m2tech Hiface 24/192 -> Sabre ES9023 -> LME49610 -> RCA -> IEM/HP of choice.
Alternatively:
Music on laptop -> USB -> m2tech Hiface 24/192 -> Sabre ES9023 -> LME49610 -> RCA -> stereo receiver -> speakers.
Sure enough, when something sounds "too good" or "unusual" I check the DSP settings in foobar:
Right so SoX was turned on, makes sense.
I turn it on and off on the fly, while listening to the track, with these arrows:
I think the difference is quite faint, but I like the way the upsampling sounds, and I think it shouldn't be very difficult to ABX it,
so I proceed to right click the track, and click on convert, here we have the conversion settings:
and here:
This time I selected upsampling x4 (176 kHz) instead of 192 kHz, however as usual, the replaygain scan is all stuffed up, which invalidates the testing:
There's more issues I have with the testing, I have no way of knowing if the ABX comparator is successfully playing the files the same way as they are played to me in foobar, and after the testing the ABX comparator doesn't tell me which is which (A/B are constant, X/Y differ), so it's only purpose is localising a difference, when executed correctly, yet it doesn't show if the listener knew which
difference was which.
I mean hypothetically, if someone uses ASIO in the foobar program, how do we know the ABX comparator is using ASIO as well, with the same settings? It just takes a chunk of data and plays it, and checks if you can hear the difference, which is useful, but I think it could be more advanced.
Here is what a downsample looks like, by the way, so as you can see, data is cut off and lost in the downsample from 192kHz to 44.1kHz, (user "gregorio" kept on insisting to me that no sound exists above 20kHz).
IMHO gregorio was a good example of a "cynical scientist", where instead of practicing impartiality, they chose the side with the least snake-oil, and then defend it violently.
I kept on telling him I agree with him that no one can hear above 20kHz, and he kept on saying the fact we can't hear above 20kHz is "simple proof", and implying that people that think they can hear above 20kHz are "idiots".
Personally, I think there's a possibility that sounds above 20kHz may have some effect on the audible, sub-20kHz spectrum, via interaction, overtones, filtering, etc.
Either way, as is demonstrated in history over and over, and by looking at "scentific studies" that are completely devoid of intution, discussing facts on paper sometimes leads nowhere, or to error, and as you can see in politics, humans are extremely partial to their personal circumstances.
I think the best thing to do is to find out for yourself.
I think it's nice that the developers of audio equipment keep on pushing the envelope, yes sometimes it might look like it's being pushed too far, yet along that path insightful discoveries often come along, so why not?
Anyway, my experience so far has made me lean towards the conviction that upsampling is more important than bit perfect.
The reasons are simple, I think the difference between 44.1 and 192kHz
is there, yet faint, the difference in size of the files is colossal, no portable devices AFAIK support 24/192 playback, and you'd fill the entire device too quickly, at least now in 2011 when we are limited to 64GB DAP's which are not sufficient, yet perhaps in 2016 with 640GB DAP's, then native 24/192 would be ok, (purely talking in a sense of space here).
In other words, I don't want native 24bit/192kHz files on my computer, when upsampling sounds just as good if not better, from what I can tell?
The next reason, how much data is in native 24/192? Hardly any at all... so if upsampling sounds almost just as good, then what's the point in native?
I'd rather watch youtube and DVD's all upsampled into 24/192, than watch youtube... in native bit perfect, right?!
So yeah, what my testing and personal listening experience has led me to, is that I quite enjoy the faint effects of upsampling, via the SoX plugin in foobar, so I definitely wouldn't mind having a USB DAC/Amp that upsampled
everything I fed it, instead of the one I currently have which only supports native.
The next point I have though, is the quality of the upsampling, which is a very complicated, instrinsic process, right?, so if a DAC only advertises "upsampling" and that's it, how do I know what the quality of the upsampling, is like?
Here's an example:
Here I'm upsampling 22.05kHz (which sounds horrible) to 64kHz, as you can see the information in 22.05kHz is being heavily cut off, due to the flatness at the top, and yet the 64kHz upsampling still has that flatness at the top, shouldn't it predict what the information will look like above 11kHz and create some fake information up there?
Finally, here is that Rihanna test again:
In this test I have
not ticked the box, since when I perform this test and I
have ticked the box, I learnt how to listen for slight volume differences, and after a while it became really easy.
Then I unticked the box and started failing a lot, but I'm getting better at it now... and I am
not listening for volume differences anymore, I am listening for air between notes, effortlessness, fine-grain/coarseness (for example in vocals), and difference in emotion.
