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Audophile quality vs 256kbps? Please HELP - Page 2

post #16 of 32
Quote:
Originally Posted by Eisenhower View Post

I wouldn't worry about it...  too much. For IEMs, don't worry. For hi-fi speakers or full size audiophile headphones, it does matter a little.

Not in myexperience. I have a pretty fantastic stereo, and my media serveris all AAC 256 VBR. No need for lossless. Do a controlled test and you'll find out too.
post #17 of 32
Thread Starter 

so im guessing i should choose the highest quality when i go with VBR? i tested it on one song and i had it set to compress to 256 using VBR with highest quality and it changed to 281

post #18 of 32
You don't need to do anything with VBR. It will raise or lower the bitrate as needed.
post #19 of 32
Thread Starter 
Well when i turn on vbr, it asks ne what quality i should choose (lowest to highest) i chose highest of course which compressed the song above 256 so i guess ill stick with that one
post #20 of 32
Quote:
Originally Posted by SlyCarrot View Post

Well when i turn on vbr, it asks ne what quality i should choose (lowest to highest) i chose highest of course which compressed the song above 256 so i guess ill stick with that one

 

Its a good choice, but doesn't work that way really. What you choose is the average bit rate. The algorithm will adjust itself based on the data it needs to compress in that frame. MP3 is usually read in 'frames', something like video. Each frame will have an associated bitrate, essentially limiting the amount of information that can be put in that frame.

The good thing about VBR is that its not hard limited by the bit rate you specify, so it can go both ways from your specified bit rate. Quiet portions use less bitrate, portions with more data use higher bitrates. Of course the hard limit here is the max bit rate supported by the encoder.


Edited by proton007 - 9/24/12 at 10:14pm
post #21 of 32
Quote:
Originally Posted by bigshot View Post


Not in myexperience. I have a pretty fantastic stereo, and my media serveris all AAC 256 VBR. No need for lossless. Do a controlled test and you'll find out too.

 

I have done a blind listening test in a studio with some professional Focal monitors, and although I personally couldn't tell the difference for every song, the people who worked there were able to, and pointed out certain things that I picked up on.

I doubt you have an acoustically treated room (most people don't). It makes a big difference.

 

Hard drive space is dirt cheap. There is no reason to rip at less than the highest quality if you are using expensive hi-fi equipment.


Edited by Eisenhower - 9/25/12 at 8:42am
post #22 of 32

I tried once doing an ABX test with Foobar.

Its really hard to tell which one is which, and you'll develop a certain knack for it after repeated listening. That lasts for one song. confused_face.gif A new song, and there'll be new artifacts you'll need to recognize.

 

However, I'm curious as to whats the golden benchmark I keep hearing about. Usually everywhere I've read, the pros say they can pick the two apart, and the common answer is, "you'll be able to do as well, once you know what to look for."

 

Please can someone impart this wisdom to us uninitiated??

post #23 of 32
Quote:
Originally Posted by Eisenhower View Post

I have done a blind listening test in a studio with some professional Focal monitors, and although I personally couldn't tell the difference for every song, the people who worked there were able to, and pointed out certain things that I picked up on.

I bet that was MP3, not AAC.
post #24 of 32
Quote:
Originally Posted by proton007 View Post

'm curious as to whats the golden benchmark I keep hearing about. Usually everywhere I've read, the pros say they can pick the two apart, and the common answer is, "you'll be able to do as well, once you know what to look for." Please can someone impart this wisdom to us uninitiated??

It either artifacts, or it doesn't.
post #25 of 32
Quote:
Originally Posted by proton007 View Post

 

Its a good choice, but doesn't work that way really. What you choose is the average bit rate. The algorithm will adjust itself based on the data it needs to compress in that frame. MP3 is usually read in 'frames', something like video. Each frame will have an associated bitrate, essentially limiting the amount of information that can be put in that frame.

The good thing about VBR is that its not hard limited by the bit rate you specify, so it can go both ways from your specified bit rate. Quiet portions use less bitrate, portions with more data use higher bitrates. Of course the hard limit here is the max bit rate supported by the encoder.

i once had a VBR codec for MP3 that i could set to go as low as 32kbps and as high as 320. it was pretty cool to watch the bitrate indicator on winamp and see how some musically "bland" stuff would dip down to 32-64kbps while other things with tons of harmonics would send it up to 320.

post #26 of 32
Quote:
Originally Posted by bigshot View Post


I bet that was MP3, not AAC.

 

vbr LAME encoded mp3 (v0 setting), which is as good as AAC.

 

AAC has the biggest advantages over mp3 at low bit rates.

post #27 of 32

What was the difference in sound? Did they level match?

post #28 of 32
Thread Starter 
So im guessing itunes match will be a good inestment for me since s lot of my song are 128kbps
post #29 of 32
Quote:
Originally Posted by bigshot View Post

What was the difference in sound? Did they level match?

 

Yeah the levels were not touched.

They noticed differences in the high frequencies. In particular, snare drums sounded more "crunchy" on the mp3. I actually preferred it during blind listening tests. It was very subtle though. I mean, the addition of air conditioning noise would be enough to mask it.

post #30 of 32

Five bucks says that they were using a track that was peaked out. The drum beats were probably clipping. I've found that certain MP3 encoders tend to shift the volume a little bit as they encode. iTunes generally encodes a little quieter than the original track, but there are some that boost the level a hair. If the track was normalized up to 100% (like almost all rock music is) and they used one of those encoders, it would push the loudest part of the track (the snare drum transients) up into clipping and make them sound crunchy.

 

If you see them again, take a track that is normalized down to 90% off the peak and see if it's still crunchy. I bet that fixes it.

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