For some time now I've been using AAC as the default codec for portable usage (128GB OnePlus 5) and I've been quite happy with it. Because I didn't want to mess with Winamp or iTunes (or any method of extracting the needed files from their installers) I've always used AAC (FDK) encoder included with foobar2000.
The other day I thought of bumping the encoding quality a little (since a slight size increase would not be a problem) and decided to re-encode my library with the VBR 5 (instead of VBR 4). Halfway through the encoding process I checked the results and the size increase is quite big. By sampling 6 albums (89 tracks) VBR 5 is about 1.78 times bigger than VBR 4. (131 kbps vs 233 kbps average bitrate).
So my question comes...
Is there a way to encode with some profile between VBR 4 and VBR 5?
I have to mention that I tried tweaking the settings from foobar2000 and looked at fdkaac.exe's parameters without figuring out anything useful... VBR 4.5 results in VBR 4
once again, and for the last time, i need to say i just trusted the high res audio hype. from vhs to dvd to bluray the quality (resolution) of visual content got better, so i just believed high res audio will be this step for audio content. but i need to figure this out for myself for sure, but i was definitely planning to buy better soundcard, soundsystem, etc..
at the end of the day I AM happy with my audio cd collection. there are all in the same specs, 16bits 44,1 khz, and this somehow is giving me a good feeling. but i would go for a better sound if there "would be" one.
finally it seems to be very tricky. i've listened to a guano apes record from 1997 that got released in highr es audio last year.
yes! it sounded better, but after some hours finding out why i have to admit it seems it is just a different mix of the tracks.
the same with cd and vinyl. the effect when someone says "hey, vinyl sounds better because of this and that", and you listen to vinyl, you somehow maybe listen to it with different ears.
finally. having my audio cds ripped in .flac became "the standard" for me, and i am absolutly happy with the sound as it is.
Last post by Mustardman -
Back again after several months, and I see that no-body has made any follow-up comments, so I will make some of my own in case anyone is interested (or is even listening!).
The suggestion of using DVDisaster (using multiple drives and the feature that allows filling of "blank" areas of a poor read with good samples from the same disc in another drive) sounded very promising. However, the software specifically does NOT support audio CDs. Whether this is a technical limitation or one that has been put in by the author (to prevent being sued, and I can understand his stance) is unknown.
I did try the suggested 'abandonware' product "PerfectRip" on my own CD-Rs, hoping it could spit out the C2 error flags (so they would help me pinpoint errors), but because of either a problem with my drive/s, or an incorrect interpretation by [mjb2006] on the software, or some setup/ini file settings that are not explained in any documentation (or webpage) that I have found - it did NOT output C2 error markers. Sigh, my hopes had been so high
Life has got in the way of recovering my CD-Rs, but I now have multiple copies of them as rips done by EAC with different drives, stored away on both on-site and off-site hard disks, and I have stored my bad CD-Rs in the cellar in a cool, dry and dark place.
could it just be that a lot is lost between what he's trying to say and what you think he said? because I have a hard time imagining a math professor saying something along the lines of: sound has only one dimension. even considering a punctual source or punctual recording position, we're still very much getting amplitude over time.
just here to contradict myself like a boss. amplitude over time is actually a one dimensional signal. Saratoga is right and reading his post got the me from more than 20 years ago instantly punching today's me in the face. I first went for a 2 dimensions space for the graph showing a signal, but time is the only independent variable in the function.
sorry math guy, sorry younger me. I'm a fool who forgot all his math.
Also, the Fourier transform can be used to convert audio data to a frequency domain representation, which is a 2D array of frequency bins that vary in intensity over time, so in that sense audio could be considered 2D.
The FFT of a signal returns a function which itself returns a two-dimensinal value for each input component. It is therefore one-dimensional as well.