[HiResSndWizzard]

what is the motive behind this test. Give me a total run down for the reasons behind what you want me to do so blindly, not knowing who you are, or why you want to know information gathered on your' playing field?

 

[Music Alchemist]

  what is the motive behind this test. Give me a total run down for the reasons behind what you want me to do so blindly, not knowing who you are, or why you want to know information gathered on your' playing field?

 
I thought you knew what an ABX test was.
 
https://en.wikipedia.org/wiki/ABX_test
 
The purpose is to determine whether you really can distinguish between the two files. If you cannot pass at least 15 out of 20 trials, then your results are no better than random guessing, meaning that it was the placebo effect and you were merely imagining the difference.
 
Also, you need to convert the files properly, without doing any extra processing. I thought you knew how to do that as well. If you need help doing this, just ask.

 

[HiResSndWizzard]

I can pass any test I set out to complete, but I want to know for what reward. Do you benefit from this test, do I benefit, or do all of us get a hamberger & milk lunch for the effort?

 

[HiResSndWizzard]

well hell no, I don't know what this test is all about?
 
Are you saying weather I can distinguish, or simply anyone can distinguish between the start file through a standard compression & back to the original as a progression, not just copying the file and submitting it twice.
 
Is this what the test is looking to confirm?

 

[Music Alchemist]

  well hell no, I don't know what this test is all about?
 
Are you saying weather I can distinguish, or simply anyone can distinguish between the start file through a standard compression & back to the original as a progression, not just copying the file and submitting it twice.
 
Is this what the test is looking to confirm?

 
I will break it down into steps so there is no confusion.
 
Start with a 16-bit / 44.1 kHz file.
 
Convert it to...let's say 24-bit / 96 kHz using a program like dBpoweramp or foobar2000.
 
http://dbpoweramp.com
http://foobar2000.org
 
If you need instructions on how to do the conversion, let me know. But since you already claimed that you were able to do these conversions, I would imagine that you would not need a tutorial.
 
Then you do the ABX test in foobar2000 and publish the results. I linked you to a tutorial on how to do that.
 
The ABX test has two known variables and one unknown variable. A and B are known, whereas X is unknown. For example, A can be the 16-bit file and B can be the 24-bit file. A and B will play one after the other, then something else (X) will play at random, out of the two, but without telling you which one it is. You choose whether you think X is A or B. I already linked you to the Wikipedia article explaining what an ABX test is. You should click and read the information on the links I provide.
 
As for "between the start file through a standard compression & back to the original as a progression, not just copying the file and submitting it twice.", I have no idea what that is referring to.
 
The ABX test is to determine whether you can really tell the two files apart. It is the only way to back up your claim.

 

[upstateguy]

 

  I can't hear a difference between a high rez file and flac from a CD, but I'd like to play around with his upsampling technique, if I could figure out how to do it.

 
Just use a program like dBpoweramp or foobar2000 and convert a 16-bit / 44.1 kHz lossless file to a 24-bit / whatever kHz lossless file. With dBpoweramp, you'll need to use uncompressed lossless to do it, though.
 
As for hi-res vs CD, that depends on the master. Some hi-res downloads sound very different from their CD counterparts; others sound the same. This is why you need to convert the files yourself to make sure you are comparing the formats and not merely different masters.

 
 

 
There is no way this testing could be a double blind test of the file integrity. Without my knowing the artifact data integrity has not been altered, I could not predict any outcome of upsample. I have to assume first of all this is a trick to see what I know, vs what you can hide from my knowing in this trial testing stunt.
 
You want to do it on my terms? I will record a test file at 48k24b, compress it to 44.1 16 bit, then return that file plus or minus a noise figure increase, which I can remove as well, with a special FFT I personally created. I have done it time and time again, so I have total confidence I can do it again this time.
 
The meer fact that a lot of current files available have been altered in the various areas I previously mentioned. Those files will, for what it is worth, do absolutely nothing for the upsample, leaving me looking like a fool.
 
No. You can save your' joy ride through technology lane for someone else. I have no reason to put forth the effort for no reward.
 
I suspect the possiblity, you know something is there, but don't know the real secret how upsampling is done correctly. It stares you right in the face everytime you play a file. I'm not going to be the one to show you simply for one reason. Intelectual property rights, vs you, the website, & the general public would grab it & run with it, leaving me sitting here scratching my head as to what I did wrong?
 
No, I don't buy into that deal. But, in all fairness, I will take a look at this site to see what hooks are hiding there to snag intelectual property from me.
 
I really am not into ripping tagging, but ocassionally transcode a video file.
 
Somehow I invision you sitting there wondering just what I might know, that you don't. Considering the hidden secret I found years ago, relating to upsampling from a compression to a higher res file, I am not so sure anyone else has ever thought of it. It is for sure I have never told, or explained to anyone over the years what I found by pure accident.
 
Besides that fact, there are other artifacts that come with the upsample, which is in some ways just as bad as other unwanted artifacts. I wrote an FFT for the removal of those artifacts.
 
When I get done with the file, it will sometimes sound better than the original master, considering some of the same artifacts are in the original master, along with the new ones created during the upsample.
 
And, yes, you can call upsampling the same thing as transcoding to some extent, but that is where the simularity ends. There is a lot more to it than just resampling with pro tools. That is as far as I am willing to go to explain the technic.

 
If you're not going to show us your technique for upsampling files why are you wasting our time babble rapping about it?
 
I would love to play around with your upsampling technique, so if you change your mind and want to share it please PM me.
 
Until I try your technique myself, I have no alternative but to consider your posts nothing but hot air.  I'm sure you would feel the same way if the situation was reversed.
 
Best Regards

 

[HiResSndWizzard]

Ok so I am zeroing in on some of the perameters of this test, finally. Well, I will say, the deal sounds sort of intreging. But when you say without any aditional processing.
 
Am I to assume the order of of processing includes only a down sample, then an upsample with no stopping at a bar along the way for any other processes. Just plane Jane sample down, sample up? It can't happen that way because of how you have to unlock the alias, isn't just that simple.
 
I would conseed right here & now, under the your rules for this test. My first answer to that test patch, "It is not possible" to recover the original file using your' idea of standard sampling programs, say like pro tools , or sound forge, or any number of other editors.
 
That is straight from the live horses mouth!

 

[Music Alchemist]

  If you're not going to show us your technique for upsampling files why are you wasting our time babble rapping about it?
 
I would love to play around with your upsampling technique, so if you change your mind and want to share it please PM me.
 
Until I try your technique myself, I have no alternative but to consider your posts nothing but hot air.  I'm sure you would feel the same way if the situation was reversed.
 
Best Regards

 
If it sounds different, then he is doing extra processing to the files, not merely upsampling them. Upsampling would be converting a file from 16-bit to 24-bit, for example. That is a very simple file conversion that many programs can do within moments.
 

  Ok so I am zeroing in on some of the perameters of this test, finally. Well, I will say, the deal sounds sort of intreging. But when you say without any aditional processing.
 
Am I to assume the order of of processing includes only a down sample, then an upsample with no stopping at a bar along the way for any other processes. Just plane Jane sample down, sample up? It can't happen that way because of how you have to unlock the alias, isn't just that simple.
 
I would conseed right here & now, under the your rules for this test. My first answer to that test patch, "It is not possible" to recover the original file using your' idea of standard sampling programs, say like pro tools , or sound forge, or any number of other editors.
 
That is straight from the live horses mouth!

Ah, so you admit that you are doing extra processing to the file! Well then of course it sounds different.

 
If you simply convert a 16-bit / 44.1 kHz file to 24-bit / 96 (or whatever) kHz, or vice versa, there is no audible difference between the two files.

 

[HiResSndWizzard]

Tell you what. I am going to take a break for sleep, but when I get back, I will take a look at all of this. To be honest, it sounds interesting.
 
Hold your breath, I will surely get back to you on this test subject, ok?
 
In the mean time, I will catch up with you sooner than later.
 
The only thing I can say straight up, is the upsample from a compression involves a lot of if's in the whereabouts of file artifacts, & my exclusive right to unlock the alias in the file before transcoding.
 
No thanks on the fubar2000, or db power amp editors. I need the right appz's to make it happen, which I already have installed in this terminal. Even with 12 terabytes, this terminal is overloaded with appz's now. If sound forge isn't good enough for you, the deal is off.
 
Good night.

 

[Music Alchemist]

  Tell you what. I am going to take a break for sleep, but when I get back, I will take a look at all of this. To be honest, it sounds interesting.
 
Hold your breath, I will surely get back to you on this test subject, ok?
 
In the mean time, I will catch up with you sooner than later.
 
The only thing I can say straight up, is the upsample from a compression involves a lot of if's in the whereabouts of file artifacts, & my exclusive right to unlock the alias in the file before transcoding.
 
No thanks on the fubar2000, or db power amp editors. I need the right appz's to make it happen, which I already have installed in this terminal. Even with 12 terabytes, this terminal is overloaded with appz's now. If sound forge isn't good enough for you, the deal is off.
 
Good night.

 
How about this, then...as per @upstateguy's suggestion, you share a before and after file (done your way) by uploading them to a file sharing site, so we can hear for ourselves.

 

[upstateguy]

 

  If you're not going to show us your technique for upsampling files why are you wasting our time babble rapping about it?
 
I would love to play around with your upsampling technique, so if you change your mind and want to share it please PM me.
 
Until I try your technique myself, I have no alternative but to consider your posts nothing but hot air.  I'm sure you would feel the same way if the situation was reversed.
 
Best Regards

 
If it sounds different, then he is doing extra processing to the files, not merely upsampling them. Upsampling would be converting a file from 16-bit to 24-bit, for example. That is a very simple file conversion that many programs can do within moments.
 

  Ok so I am zeroing in on some of the perameters of this test, finally. Well, I will say, the deal sounds sort of intreging. But when you say without any aditional processing.
 
Am I to assume the order of of processing includes only a down sample, then an upsample with no stopping at a bar along the way for any other processes. Just plane Jane sample down, sample up? It can't happen that way because of how you have to unlock the alias, isn't just that simple.
 
I would conseed right here & now, under the your rules for this test. My first answer to that test patch, "It is not possible" to recover the original file using your' idea of standard sampling programs, say like pro tools , or sound forge, or any number of other editors.
 
That is straight from the live horses mouth!

Ah, so you admit that you are doing extra processing to the file! Well then of course it sounds different.

 
If you simply convert a 16-bit / 44.1 kHz file to 24-bit / 96 (or whatever) kHz, or vice versa, there is no audible difference between the two files.

 
I think there's no point of pursuing this any further.
 
He's not going let us in on his imaginary technique and as far as I'm concerned, he's giving it to us straight from the horses other end.

 

[FFBookman]

  16-bit / 44.1 kHz is the highest resolution you will ever benefit from. There is no loss in the sense of anything that is audible.
 
Read this article for background info.
 
Since you asked about upsampling, you may be interested in HQPlayer. It does oversampling, dithering, noise shaping, modulation, and other DSP. To me, it sounded much better than other players, but it's expensive and the interface is awful.

Wow is that false.  Even 24/44 sounds better than 16/44, given a proper transfer from tape.  See The Cars and The Beatles for 24/44 masters.
 
It's like saying no one ever can see more than 640x480 resolution, because you can't see it on your singular screen.  Stop it. Stop it. Stop it.  1984 called, it wants it's argument back.
 
Let's cut to the chase --- stereo PCM file -- effective bandwidth -- are you claiming you can hear a difference between 256k and 1400k, but you can't hear a difference between 1400k and 2800k, or 1400k and 5800k?
 
Listen to reverb trails. Listen to air in the room. Listen for how many voices (sounds) can be presented at once. Vocal choruses, choirs. Listen to room shape, accuracy of delays, timing cues, width, depth, even timbre of instruments -- all of that improves with higher bitrate.
 
Yes bad gear in the signal chain can mask it. If you play through laptop-level circuits then 16/44 might be all you need. But as soon as you use a real DAC with good analog, it's easy to hear beyond that. Especially if you are projecting the sound to a large room.
 
Btw -- my mastering engineer said he takes everything to 24/192 immediately upon delivery, regardless of resolution it is delivered as, and has for over 15 years now.  He then applies his processes (mainly EQ), masters the material, and then down samples and dithers to whatever resolution / format is required by the client.
 
His reason?  "There's more there there", and what some call headroom he calls air, and it's critical to the sound of the music (assuming it's not modern pop/rock squashed to all high hell).
 
When you create music from scratch in a studio it's very easy to hear these differences. Perhaps on the train with an iPhone it's not as easy, but the differences are still critical.

 

[cjl]

   
Let's cut to the chase --- stereo PCM file -- effective bandwidth -- are you claiming you can hear a difference between 256k and 1400k, but you can't hear a difference between 1400k and 2800k, or 1400k and 5800k?

Oh god, not this argument again.

 

[Steve Eddy]

Oh god, not this argument again.

In order for it to be "this argument again," it first has to be an argument.

se

 

[Music Alchemist]

  Wow is that false.  Even 24/44 sounds better than 16/44, given a proper transfer from tape.  See The Cars and The Beatles for 24/44 masters.
 
It's like saying no one ever can see more than 640x480 resolution, because you can't see it on your singular screen.  Stop it. Stop it. Stop it.  1984 called, it wants it's argument back.
 
Let's cut to the chase --- stereo PCM file -- effective bandwidth -- are you claiming you can hear a difference between 256k and 1400k, but you can't hear a difference between 1400k and 2800k, or 1400k and 5800k?
 
Listen to reverb trails. Listen to air in the room. Listen for how many voices (sounds) can be presented at once. Vocal choruses, choirs. Listen to room shape, accuracy of delays, timing cues, width, depth, even timbre of instruments -- all of that improves with higher bitrate.
 
Yes bad gear in the signal chain can mask it. If you play through laptop-level circuits then 16/44 might be all you need. But as soon as you use a real DAC with good analog, it's easy to hear beyond that. Especially if you are projecting the sound to a large room.
 
Btw -- my mastering engineer said he takes everything to 24/192 immediately upon delivery, regardless of resolution it is delivered as, and has for over 15 years now.  He then applies his processes (mainly EQ), masters the material, and then down samples and dithers to whatever resolution / format is required by the client.
 
His reason?  "There's more there there", and what some call headroom he calls air, and it's critical to the sound of the music (assuming it's not modern pop/rock squashed to all high hell).
 
When you create music from scratch in a studio it's very easy to hear these differences. Perhaps on the train with an iPhone it's not as easy, but the differences are still critical.

 
Take a 24-bit file. Convert it to lossless 16-bit / 44.1 kHz. Sounds exactly the same. Always.
 
The only reason some hi-res files sound different is because they came from a different master. Isolate the variables by converting the files yourself (to ensure you are comparing two different resolutions instead of merely two different masters) and those differences disappear.
 
As has been stated countless times elsewhere, 24-bit just adds more dynamic range, but 16-bit can already handle the dynamic range of all recordings. And frequencies above roughly 20 kHz are inaudible to humans. These things have no possible benefit in terms of audio playback.
 
If you would like to prove otherwise, simply publish a proper ABX test. Many people here can walk you through how to do so.