Why would 24 bit / 192 khz flac sound any better than 16 bit / 44.1 khz flac if both are lossless (if at all)?

[bigshot]

Thanks, but this is not what we were talking about. The point that I made was that I need time to adapt to a particular sound signature before switching to another for the DBT to be worthwhile, it's different than echoic memory.

I find that for myself, quick switching back and forth between samples is the best way to determine whether a difference exists between two very similar samples. In order to figure out exactly what the difference is, it takes a little longer on each sample. But after a minute or two on any sample, my ears adjust and I'm not getting anything out of it any more.

 

[elmoe]

   
You'll just have to trust that the subjects are doing their best.
 
Have you ever been tested?

 
I've done the test myself, with someone to do the switching for me, which is how I found out that taking my time between switches helped. 

  I think this thread may have gone slightly off topic...

 
Yes sorry about that, but it seems most of the responses above cover what I wanted to know so I won't hijack it much further.
 
 

   
pretty much the opposite of what I experience. I could tell you a list of sonic differences from listening to the same gear with the same music file if you just give enough time to my brain to start making stuff up. and I would believe I'm right.
I wrote something just yesterday about some 3seconds lag that was already enough for me to feel insecure about my analysis (the time needed for me to unplug one source and replug the other one). I didn't know about that echoic memory thing(super interesting), but I guess it just explains what I always felt in practice.
 
 

 
I think you've misunderstood, the time in between the switch (the 3 seconds lag as you called it) is not what I'm referring to. That lag should be as minimal as possible so you get to hear the other source or whatever it is you're testing, right away. I was saying that given a small recording (say 30s), I need to listen to that recording for 5-15 minutes before switching between source (or whatever I'm testing) so that my ears can adapt to its sound signature before I can pick out any differences.
 
 

 
Elmoe,
 
Listen to what folks are telling you.  It has indeed been tested rigorously.  Subjects detect differences at lower levels with fast switching rather than with slower switching.  Yes, the common argument you are making when tested doesn't work out to be true.  You do less well if the switching is very long.  Past 10 seconds and things fall off quite a bit. In speech intelligibility testing somewhere around 200 msec switching was needed for the most discriminating results.  You probably feel much better and more confident with longer times between comparisons, but despite your feelings of confidence your accurate discrimination will be worse.
 
http://www.nousaine.com/nousaine_tech_articles.html
 
On this page, down in the middle of it read the PDF from a magazine article on Flying Blind Long Term Listening.   There are more scholarly works on the subject.  But this gets to the gist of it.  And we'll see if you believe it or just retrench as stv104 and I think you will.  We've been here, and done that about a million times.  It gets old.  You can supply all the credible info you want to most people, and they just refuse to believe it because they don't want to.  Maybe you are different, I hope so.  So far your postings follow right along with someone who we are wasting our time to converse with. Not trying to be lacking in respect for you, just being honest.

 
Ok I've finished reading that article, and what I'm saying is very different from what's argued in it, so let me be clearer. In the article, it's explained that long term listening (we're not talking about 15 minutes, then switch here, but about many many hours/days/weeks of listening without any switching) is not going to accurately help you find differences as opposed to a DBT ABX with switching.
 
I couldn't agree more with that, but that is not what I'm saying. I'm saying that WITHIN AN ABX DBT, instead of listening for only a FEW SECONDS before switching, we should be listening for 5-15 minutes and THEN switch, so our ears can adapt somewhat to the sound signature before we switch, thus making the difference more obvious.
 
That being said, I do think that quick switching can be the better choice for some things (such as this thread for example, determining whether or not 2 samples, one at 116/44 and the other at 24/192, are different or not). However, when comparing gear, for example 2 different DACs, fast switching is more likely to confuse the brain than to help pick out differences. So if you're going to pick out one aspect (such as in the tests from the article, distortion), then quick switching is no doubt most effective and that's logical because you're listening for it during the test, so you don't need prolonged exposure to hear it - you either do or you don't. But comparing 2 different DACs, you're not listening just for distortion or any single aspect, there are many many things to compare and without taking your time with each, there is no way to accurately compare their sound.
 
So yes, I believe you and this article absolutely, but that's beside the point and it isn't what's ultimately interesting to me: knowing if different gear brings different sound and how. So you're right, I was too hasty before and only remembered back to the few ABX DBT tests I did myself comparing gear, without thinking about the usefulness of these tests to compare smaller aspects such as "is there or is there not distortion". For these kinds of tests, I completely agree, quick switching is the prerogative. My problem is that the same procedure is used when comparing gear and that seems flawed to me. I'm not trying to be stubborn, it just seems illogical.
 
 

I find that for myself, quick switching back and forth between samples is the best way to determine whether a difference exists between two very similar samples. In order to figure out exactly what the difference is, it takes a little longer on each sample. But after a minute or two on any sample, my ears adjust and I'm not getting anything out of it any more.

 
Yes I can see how that would be the case when comparing 2 short samples. But like I said above, when comparing 2 different DACs or amps or any other piece of gear, wouldn't you prefer to take the time to a) try a variety of recordings and b) listen long enough to register all the different ways it sounds?

 

[limpidglitch]

   
I've done the test myself, with someone to do the switching for me, which is how I found out that taking my time between switches helped. 
 

 
I've had much the same experience as bigshot, but then I don't have much experience with comparing DACs and amplifiers in this way.

If you've had more success in blind tests with longer switching times, there's really not much to argue about.

 

[elmoe]

   
I've had much the same experience as bigshot, but then I don't have much experience with comparing DACs and amplifiers in this way.

If you've had more success in blind tests with longer switching times, there's really not much to argue about.

 
I agree, but it's nice to see what others have experienced doing the same kind of testing.

 

[bigshot]

  Yes I can see how that would be the case when comparing 2 short samples. But like I said above, when comparing 2 different DACs or amps or any other piece of gear, wouldn't you prefer to take the time to a) try a variety of recordings and b) listen long enough to register all the different ways it sounds?

 
A yes, B not more than a couple of minutes between switches.
 
I generally switch back and forth a lot in different parts of the music and with different music. But if I stay on any one sample longer than a couple of minutes, I end up listening to the music, not listening for any difference. My ears and brain just can't hold very similar comparisons that long. If I listen for a long time, I often hear things that I didn't hear before, but when I take that section and switch A/B over it, I realize it was just my faulty memory or not paying attention. The difference is in my memory, not in the test.

 

[elmoe]

   
A yes, B not more than a couple of minutes between switches.
 
I generally switch back and forth a lot in different parts of the music and with different music. But if I stay on any one sample longer than a couple of minutes, I end up listening to the music, not listening for any difference. My ears and brain just can't hold very similar comparisons that long. If I listen for a long time, I often hear things that I didn't hear before, but when I take that section and switch A/B over it, I realize it was just my faulty memory or not paying attention. The difference is in my memory, not in the test.

 
That makes sense actually. I'll have to try another DBT just to confirm when I get another DAC.

 

[Makiah S]

http://people.xiph.org/~xiphmont/demo/neil-young.html
 
Here read through this, I felt it makes a lot of sense, and further supports my opinion that 24/192 formatting is uncessiarly terrible for play back 

 

[gevorg]

What if you're adding DSPs to your playback chain, such as TB Isone. Wouldn't high resolution recordings (in particular, the 24-bit side of them) will give you more "headroom" for DSP processing, the same headroom that pro audio guys use when they record/master in 24/96 and then down-convert to redbook when they're done.

 

[kraken2109]

What if you're adding DSPs to your playback chain, such as TB Isone. Wouldn't high resolution recordings (in particular, the 24-bit side of them) will give you more "headroom" for DSP processing, the same headroom that pro audio guys use when they record/master in 24/96 and then down-convert to redbook when they're done.

It's possible that 24bit would help, but chances are you wouldn't even be using half of what 16bit is capable of.

 

[bigshot]

I would think that if a DSP required that kind of headroom, it would automatically upsample/process/downsample on the fly. It wouldn't require you to have high bitrate music.

 

[castleofargh]

What if you're adding DSPs to your playback chain, such as TB Isone. Wouldn't high resolution recordings (in particular, the 24-bit side of them) will give you more "headroom" for DSP processing, the same headroom that pro audio guys use when they record/master in 24/96 and then down-convert to redbook when they're done.

that's a very good point. I guess it would be hard to make a general statement because each dsp does its own thing. I only get how it works for pictures, where the resulting quality depends a lot on how much data we have before the effect. but I guess it would translate more into sample rate than into bits in our situation right? and as long as the sample is big enough to get the right wave, should it matter?
argghhhh I wish I would know more about all this.

 

[ralphp@optonline]

 
that's a very good point. I guess it would be hard to make a general statement because each dsp does its own thing. I only get how it works for pictures, where the resulting quality depends a lot on how much data we have before the effect. but I guess it would translate more into sample rate than into bits in our situation right? and as long as the sample is big enough to get the right wave, should it matter?
argghhhh I wish I would know more about all this.

There have been links to several very good articles and videos about digital audio posted throughout this thread so if you go back and read/watch some of them then you will "know more about all this". One thing you will learn the difference between sampling rate and bit depth. Another would be that a higher sampling rate does not mean a "smoother" sine wave, which is a common audiophile myth, often passed on by the clowns in the audio press. And yet another would be that a greater bit depth does not add any dynamic range to the recording, greater bit depth just means that more dynamic range is available, however it the actual music on the recording that provides the dynamic range. For example a recording of an acoustic guitar will have less dynamic range than a recording of a full symphony orchestra regardless of whether it is 16 or 24 bit, again a common myth that is not dispelled by the audio press.
 
Lately I've been thinking about how the whole shift from disc based playback, be they black plastic LPs or shiny silver CDs or SACDS, to computer based playback has given the high end audio world a whole new area to exploit with misinformation and lies, very similar to the early days of the high end cable craze. More myths and misinformation basically means more completely useless but highly profitable products to sell to all the kool-aid drinking audiophiles.

 

[castleofargh]

 

 
that's a very good point. I guess it would be hard to make a general statement because each dsp does its own thing. I only get how it works for pictures, where the resulting quality depends a lot on how much data we have before the effect. but I guess it would translate more into sample rate than into bits in our situation right? and as long as the sample is big enough to get the right wave, should it matter?
argghhhh I wish I would know more about all this.

 Another would be that a higher sampling rate does not mean a "smoother" sine wave

yup, that's why I said "and as long as the sample is big enough to get the right wave, should it matter?" having a few less "dots" on the wave wouldn't change the wave unless there was really too
few reference values. but I don't really know how many is enough as music is not juste 1 sine wave.
 
I hungrily eat anything you guys through at me, but unless it's basic and noob oriented (like the videos from xiph.org), I usually don't grasp most of it.

 

[ralphp@optonline]

  yup, that's why I said "and as long as the sample is big enough to get the right wave, should it matter?" having a few less "dots" on the wave wouldn't change the wave unless there was really too
few reference values. but I don't really know how many is enough as music is not juste 1 sine wave.
 
I hungrily eat anything you guys through at me, but unless it's basic and noob oriented (like the videos from xiph.org), I usually don't grasp most of it.

Unfortunately the xiph.org videos, while fairly simple and, as you say, aimed at noobs, could go a long way in saving audiophiles some money if they would only watch them instead of ust believing the lies and misinformation that come out of the high end audio press (and Mr. Neil Young, although I think he's just being misled by others looking to use his good name and money for ill gotten gains).

 

[esldude]

   
 
Lately I've been thinking about how the whole shift from disc based playback, be they black plastic LPs or shiny silver CDs or SACDS, to computer based playback has given the high end audio world a whole new area to exploit with misinformation and lies, very similar to the early days of the high end cable craze. More myths and misinformation basically means more completely useless but highly profitable products to sell to all the kool-aid drinking audiophiles.

Amen to this.  I had hopes that simple good quality playback from a computer would fix all perceived and real ills for digital audio.  That finally much of the high end BS would be out.  Instead, it appears it may become the most corrupted area of music playback ever.  In just the last 5 years stunningly ridiculous products have been put forward on equally stunningly untrue ideas to solve completely non-issues.  Many of them are really crazy, and are already accepted by the mainstream computer based audiophile in the high end realm as known and certain issues.  Which of course can only be reduced by terribly expensive products.  Though the problems they reputedly solve are never quite solved.  Leaving another level of refinement and expense available for the next go around.