[Don Hills]

  I stand by my statement. On my full 2 ch floor system, most 24-bit files I listen to sound better than my 16-bit files.
 
(hmm maybe I should stop upgrading....)
 
cheers,

No DAC is perfect. (Though some of them are very close.) A DAC may perform (and maybe even sound) different when processing 24/96 versus 16/44.1.
- Select a 24/96 source you are familiar with.
- Convert it to 16/44.1 with a high quality converter, such as Sox.
- Convert the result back  to 24/96.
- Using a tool such as Foobar ABX, compare the original 24/96 and your doubly converted 24/96.

 

[CanadianMaestro]

 
No DAC is perfect. (Though some of them are very close.) A DAC may perform (and maybe even sound) different when processing 24/96 versus 16/44.1.
- Select a 24/96 source you are familiar with.
- Convert it to 16/44.1 with a high quality converter, such as Sox.
- Convert the result back  to 24/96.
- Using a tool such as Foobar ABX, compare the original 24/96 and your doubly converted 24/96.

Thanks, will try this this WE. Never heard of Sox, but will add it to my software board!

 

[Music Alchemist]

  Thanks, will try this this WE. Never heard of Sox, but will add it to my software board!

 
You can also use dBpoweramp (my personal favorite) or foobar2000.

 

[CanadianMaestro]

   
You can also use dBpoweramp (my personal favorite) or foobar2000.

I use Mac.

 

[Music Alchemist]

  I use Mac.

 
dBpoweramp works with Apple now.

 

[CanadianMaestro]

   
dBpoweramp works with Apple now.

OK didn't know that. I use XLD to inter-convert file types, but not bit depth.
Anyways, my "experimental" phase with music files is long past. So don't hold your breath.
cheers

 

[Music Alchemist]

  OK didn't know that. I use XLD to inter-convert file types, but not bit depth.
Anyways, my "experimental" phase with music files is long past. So don't hold your breath.
cheers

 
In case you do decide to use dBpoweramp, here's a brief tutorial. Right-click the file. Click Convert To. Next to Converting To, select Wave. Next to Uncompressed, select 16 bit and 44.1 kHz. Click Convert. Then label the two files (original and new) and listen. Or do the extra step of converting the new file back to high-res if your gear is screwy.

 

[CanadianMaestro]

   
In case you do decide to use dBpoweramp, here's a brief tutorial. Right-click the file. Click Convert To. Next to Converting To, select Wave. Next to Uncompressed, select 16 bit and 44.1 kHz. Click Convert. Then label the two files (original and new) and listen. Or do the extra step of converting the new file back to high-res if your gear is screwy.

 
I listen to all files on my BDP-1, thru a USB stick/drive. All I do is take the files, transfer them to a stick, plug the stick into my BDP-1, then playback thru my floor system. Flawless playback thru BDA-1. If the sound is "screwy", it's not because of my hardware.
Thanks for the tips.

 

[cjl]

   
I was kind of saying, "I dare you to try to pass an ABX test of 256 kbps AAC vs lossless." Because as far as I know, it's never been done.

It probably could be done, given a specific, carefully chosen sample, lots of fast, level-matched a/b switching, and careful practice on what to listen for. Keep in mind that lossy compression has never been claimed to perfectly audibly reproduce a lossless file - it merely tries to minimize the audibility of the pieces it throws away. Modern lossy compression is darn good at it, but I wouldn't be surprised if 256 AAC could be beaten with the right sample. I know there have been successful ABXs between uncompressed PCM and LAME 320 before. Yes, lossy audio is more than adequate for almost any possible application, but in order to claim audible perfection, you really do need lossless.

 

[bigshot]

  It probably could be done, given a specific, carefully chosen sample, lots of fast, level-matched a/b switching, and careful practice on what to listen for.

 
I wouldn't bet on that.
 
It may seem possible in theory because the zeros and ones are different, but in practice our ears are the limiting factor, and lossy compression has been *designed* to take advantage of those limitations. MP4 compression is VERY sophisticated and is designed to be transparent at high bitrates. I've got over a year and a half of music encoded AAC 256 VBR and I haven't found a single file yet where there is any audible artifacting at all.

 

[CanadianMaestro]

   
In case you do decide to use dBpoweramp, here's a brief tutorial. Right-click the file. Click Convert To. Next to Converting To, select Wave. Next to Uncompressed, select 16 bit and 44.1 kHz. Click Convert. Then label the two files (original and new) and listen. Or do the extra step of converting the new file back to high-res if your gear is screwy.

Actually, I can do the conversions with my XLD software.
FLAC 96/24 --> WAV 96/24 --> WAV 96/16
Then compare playback quality of the two WAV files. And of the WAV-16 vs. FLAC-24.

 

[cjl]

   
I wouldn't bet on that.
 
It may seem possible in theory because the zeros and ones are different, but in practice our ears are the limiting factor, and lossy compression has been *designed* to take advantage of those limitations. MP4 compression is VERY sophisticated and is designed to be transparent at high bitrates. I've got over a year and a half of music encoded AAC 256 VBR and I haven't found a single file yet where there is any audible artifacting at all.

You'd never hear it unless you did level matched fast-switch comparisons of hand selected sections of audio (selected to be the most challenging to the psychoacoustic model used to develop the codec). You wouldn't hear an artifact when just listening to the file in isolation - you need the comparison, since any errors, distortion, or artifacting would be so quiet compared to the RMS level of the signal. That having been said, I wouldn't be surprised at all to learn that it is possible to detect the difference between 256k AAC and lossless given a particularly difficult sample with fast-switch a/b comparison.
 
Not that any of that is terribly relevant to normal music listening...

 

[bigshot]

I level matched all the samples in the test I share, and with most digital players you can jump around the timeline to compare. The samples are at even increments. Also the music consists of two samples, one choral that is extremely prone to artifacting, and the other orchestral with a broad dynamic range and wide frequency response.
 
I've done direct A/B switched comparisons of lossless and 256 AAC with particularly difficult tracks to encode, and if there is a human being on earth that can tell the difference, I'd sure like to meet them!

 

[thelostMIDrange]

I'm with the OP, and feel WAV sounds more natural compared to FLAC. I dis-covered this on my own before I had even spent a second looking it up on the internet to find that they should sound the same. But it just didn't seem so to me. I've got no reason to want to prefer WAV, in fact I have plenty of reason not to, i.e. extra storage space. But I've since reripped all my vinyl into WAV which is also a task in itself so I must be delusional to spend all that time when it doesn't matter. Discussing the topic with the local hifi shop, the fellow I spoke with agreed and suggested that at least in Windows operating system, he found WAV sounded better. I originally came upon the belief when using a sansa player. I had many FLACs and some WAVs and over time I felt the WAV's always sounded different. So who knows, some suggest there is some processing going on with FLAC even though it should be all done prior to the bits coming into the auditory realm, but there might be something going on that results in meaningful difference. I can understand those who feel passionately that it's the same info for FLAC and WAV and it's all in our heads and am envious of all the extra storage space they have because of it. Oh well

 

[Music Alchemist]

  I'm with the OP, and feel WAV sounds more natural compared to FLAC. I dis-covered this on my own before I had even spent a second looking it up on the internet to find that they should sound the same. But it just didn't seem so to me. I've got no reason to want to prefer WAV, in fact I have plenty of reason not to, i.e. extra storage space. But I've since reripped all my vinyl into WAV which is also a task in itself so I must be delusional to spend all that time when it doesn't matter. Discussing the topic with the local hifi shop, the fellow I spoke with agreed and suggested that at least in Windows operating system, he found WAV sounded better. I originally came upon the belief when using a sansa player. I had many FLACs and some WAVs and over time I felt the WAV's always sounded different. So who knows, some suggest there is some processing going on with FLAC even though it should be all done prior to the bits coming into the auditory realm, but there might be something going on that results in meaningful difference. I can understand those who feel passionately that it's the same info for FLAC and WAV and it's all in our heads and am envious of all the extra storage space they have because of it. Oh well

 
Would you be willing to do an ABX test to document whether you can distinguish between FLAC and WAV with statistical significance?