I am converting these "electronic music" genre files @ 128k but the encoder was going as high as 150k consistently. Is the encoder struggling? Should I go higher? I am using 128k only because opus wiki says 128k is "pretty" transparent, for me such an assurance is enough for me.
For some reason no one answered this question, which is pretty fundamental to understanding the encoder, I think. Yes, many types of electronic music are difficult to encode, due to the synthetic instruments, with chiptunes being the most difficult genre to encode that isn't random noise. (But some chips do use random noise to stand in for sounds....) I have a library of favorite chiptunes that I'll regenerate and encode as codecs change, and for every single one, the bitrate is far higher than anything else I encode. The hard jumps just don't fit with an MDCT at all.
Opus's bitrate selection is really more like Vorbis's quality parameter, it's a quality-shaping profile that's given a bitrate based on extensive testing. Overly simple inputs, like pure vocal samples, or overly difficult, like chiptunes, will throw off the encoder significantly.
There is a way to constrain Opus to specific maximum bitrates, but unless you have an absolute need, you should let the encoder use as much or as little bitrate at the quality you're happy with.
Last post by phryxo -
Thanks, EpicForever. I guess I'll install it back to how it was, but that can't be all of it. Do people drag and drop their whole library every time they launch foobar? That would be weird...
EDIT; I reinstalled fb2k once again with the normal installation and... my library still isn't showing. Damn it. Everything worked perfectly until I decided I wanted to have a go at customizing my player.... So what else might be causing this? It says it's monitoring my music folder, but still there's nothing showing up.
Last post by greynol -
...and I'm wondering what all this sophistication is attempting to recreate and whether it addresses something as fundamental as Fletcher-Munson. I'm also wondering how it can make questionable EQ mastering decisions sound subjectively better.
In the meantime I'll be enjoying music at any listening level without worrying about how it looks on a computer screen or having my enjoyment be dictated by numerical figures on a piece of paper or how much I spent on the equipment.
What's the point of being concerned about "unbalancing the sound spectrum amplitude" without also being concerned with the way the human auditory system works?
I'm actually not opposed to room correction, but it is certainly not a substitute for a good set of tone controls (and vice versa, though I'd bet you'd be surprised how well your hearing is able to adapt and compensate for a less than perfect listening environment). Speaking of which, have the room correction gods manged to figure out how to compensate for my turning my head or moving to another seat?
Last post by silverprout -
I've got some loudspeakers that have great abilities in the bass domain and it is easy to perform a low THD measurements outdoor, and indoor. On the figures 6 to 9 here : http://www.geocities.ws/kreskovs/Box-Q.html and especially the figures 15 and 16 here : http://www.geocities.ws/kreskovs/Box-Qa.html the input signal tracking is extremely poor, the output signal is not really in phase, wich mathematically tend to a very low number of statistical correspondance. A Rice Kellogg loudspeaker is a sheet of paper (or someting else) putted in a box (if you want a punchy bass rendering) acuated by an electric motor... therfore it sounds like sheet of paper (or someting else) putted in a box (if you want a punchy bass rendering) acuated by an electric motor, trying to match the input signal to the output of acoustic energy in the air is a good place to start IMO.
RE: Tone controls are badly implemented Not those that are well-designed (in terms of how they alter tonal quality). To suggest that all tone controls are badly implemented is idiotic. Furthermore I contend that the average listener doesn't know how to use them. This includes amps with only bass and treble controls, as the average listener also doesn't understand that volume is an additional tone control as well.
A wide bands equalizer applied on loudspeakers designed to sound flat in an anechoic chamber will unbalance the spectrum amplitude of your direct singal. Solve the soundfield problems by modifying imperceptibly the direct sound requires virtuosis competences and advanced tools, i'm dubitative about how ignorant people with simplistic tools can perfom it.
Last post by EpicForever -
Wonder who writes those "guides"... The fact that program is not recognized by Windows as installed is the main goal of creating portable installation. It "keeps system clean" and allows you to move it wherever you want on USB flashdrive. But also has downsides - you can't manage file associations via control panel.
So generally if your goals are NOT "keeping Windows clean" and walking with foobar on pendrive, simply install it normally. Then you can go along those guides about rest of configuration steps. It is the best possible solution of your problem.
With Foobar2000 it defaults to basically the best mode for Apple AAC and Opus which is VBR for Opus and TVBR for Apple AAC. TVBR or CVBR are all one really needs to use for Apple AAC and VBR for Opus is typical.
I prefer TVBR over CVBR in general because sound quality wise they are pretty much the same (some basically claim CVBR is about the same, maybe a hair better than TVBR (sound quality wise) but usually at a noticeable increase to bit rates) but TVBR generally gives you smaller files (not always, but typically does) which makes it a bit more efficient and being sound quality is pretty much the same, I would rather choose the one that generally uses less storage space since it's more efficient.
like on some of my music, TVBR(Apple AAC) at 96kbps can shoot down into the 6x kbps range (sometimes lower) and I have seen it hit roughly as high as the 12x kbps range (but usually in the ball park of 96kbps give or take 10kbps or so) where as doing the same thing with CVBR it tends to be more stable around the 96kbps range as CVBR seems to limit the bit rate from going too low etc.
but basically... use the default mode and choose a bit rate that suits you (likely 96kbps or 128kbps or 160 kbps) and just use it as that's going to be more of a determiner of quality than the whole TVBR/CVBR stuff. but in my opinion using anything higher than 128kbps is pretty much a waste of storage space as efficiency starts to drop off quite a bit because of little gains in sound quality for a solid increase to file size (I am basing this info around sound test and other info I noticed around here). but I usually go for 96kbps since at that rate your getting a great balance of sound quality/file size and I would imagine people will be using these lossy files on-the-go which makes it that much less likely you need anything TOO high on the sound quality side of things (plus, not to mention... most people probably don't have any fancy headphones etc). hence, I tend to see Apple AAC and Opus to stick with 128kbps or less for the vast majority of people. I would start with 96kbps and then go from there but if you don't want to mess with it and want to play it pretty safe, one can't go wrong with 128kbps.
just some thoughts
p.s. but IgorC pretty much summed it up in much fewer words.