[Krutsch]

 
<snip, snip>
 
all that is well known, all that has been researched in detail many years ago, yet people chose to believe advertising instead of the guys who actually invented digital audio. at some point we get bored, some here are very very patient, I am not.
 
<snip, snip>

 
Exactly, so, why do you even bother to post a reply?
As I said earlier, it's always entertaining to read a good rant 

 

[bigshot]

We're always happy to explain the technical details if you don't understand. Just let us know. But otherwise we cut to the chase and point at what is causing the person's mistaken impression, not the underlying theory behind it.

 

[Don Hills]

Why pull ******** out of your ass

Why indeed. Let nature (and gravity) take its course.

 

[sonitus mirus]

 
Why indeed. Let nature (and gravity) take its course.

 
Asstronauts are probably thankful that gravity is not a factor when nature calls. 

 

[francopro]

  And you've done proper level-matched double blind tests to verify that you can actually hear the difference, right?

 
All tests have been done through different setups

 

[francopro]

   
Do you use Foobar to play music?
 
Foobar is an excellent free software that works with windows (and Linux via Wine) and has the ability to add plugins. One such plugin is the SoX resampling tool, which is an excellent resampler. Another tool is the ABX plugin which allows you to use double blind test whether you can detect the difference between two sound clips.
 
I think it would be very helpful to you and to the rest of the community of you can test one of your 192kHz sampled tracks by comparing it to a down-sampled version. You can use the Sox DSP plugin to transcode your full-res track into a 48kHz version, and then perform an ABX test. If there is any detectable difference the two, then you should be able to differentiate between them in a statistically meaningful way. You can post the results from the ABX tool here to demonstrate the audible difference between a 192kHz track and its 48kHz equivalent. You need multiple trials (say 30) for the results to begin to be statistically significant (unless you can nail 20/20 )
 
Furthermore, if you use <30 second clips, you can share the two tracks here so others can give it a try as well!
 
Cheers

I don't trust Foobar anymore coz everytime I reconvert a track , even with hi-res the output file is a real disaster to my ears, as I don't use DSD so Foobar is useless to me
I prefer VLC and Songbird for playback , DBamp for ripping and music conversion on my PC

 

[francopro]

 
I don't trust Foobar anymore coz everytime I reconvert a track , even with hi-res the output file is a real disaster to my ears, as I don't use DSD so Foobar is useless to me
I prefer VLC and Songbird for playback , DBamp for ripping and music conversion on my PC

Furthermore , I do all tests with my desktop setup (Yamaha amp/ Beyerdynamic T5p/ yamaha SACD/AK100/Audioquest cables) , the PC is just useful to purchase and downloading music, it's very hard to setup an audiophile PC even with a good soundcard

 

[ab initio]

I don't trust Foobar anymore coz everytime I reconvert a track , even with hi-res the output file is a real disaster to my ears, as I don't use DSD so Foobar is useless to me
I prefer VLC and Songbird for playback , DBamp for ripping and music conversion on my PC

It sounds like you don't know how to configure foobar. Also, what does DSD have to do with anything? DSD is irrelevant to this issue of PCM formats.

Can you describe your issue? Because Foobar is a free player with realtime upsampling and ABX tool that none of your other options offer, making it the only useful option to you for determining the (in)audibility of hires PCM formats.

If you need help, ask questions. This community is friendly and happy to help.


Cheers

 

[francopro]

It sounds like you don't know how to configure foobar. Also, what does DSD have to do with anything? DSD is irrelevant to this issue of PCM formats.

Can you describe your issue? Because Foobar is a free player with realtime upsampling and ABX tool that none of your other options offer, making it the only useful option to you for determining the (in)audibility of hires PCM formats.

If you need help, ask questions. This community is friendly and happy to help.


Cheers

Yes , I don't know how to configure Foobar the way you mentioned above because i don't use this software !
What I meant regarding DSD , I know that Foobar is one of the rare softwares that support this format, as I don't use DSD because of compatibility issues with my physical setups , that's why I don't need Foobar.
 
Thanks anyway , maybe when I'll buy a monster like Alienware systems I may reconsider Foobar but for now my pc is 3 - 4 years old with outdated soundcard

 

[esldude]

 
Furthermore , I do all tests with my desktop setup (Yamaha amp/ Beyerdynamic T5p/ yamaha SACD/AK100/Audioquest cables) , the PC is just useful to purchase and downloading music, it's very hard to setup an audiophile PC even with a good soundcard

You could buy a USB to SPDIF converter of good quality which will do up to 192/24 for something like $100.  Connect PC (even an older one) via USB to the device, feed the digital signal to your Yamaha or the AK100 and you have an audiophile PC.  So not so hard. 

 

[SharpEars]

OK, I am going to stick my face into the arena while the kicks are flying and say that I have a reason to prefer 24-bit over 16-bit. Here is why:
 
My workflow consists of taking a 44.1/16 CD and ripping it into FLAC files. I then take the FLAC file for each track and:
 

• Upsample to 24-bits
• Remove DC Offset, if present
• Reduce the volume until there are no clipped samples present
• Possibly do heuristic based automated clip repair on those samples that were clipped
• (optional) Normalize the volume to a reasonable level inline with the other tracks on the CD

 
I perform functions 2-5 in the 24-bit domain and I leave the result as a 24-bit (FLAC) audio file that is ready for listening. I do this with all of my music and I stick with 24-bit, because I believe that the steps above performed in the 24-bit domain make more sense. I leave the resultant file as a 24-bit file, so that I can perform additional steps in the future that I deem will improve the tracks sound quality (e.g., equalization, excitement, whatever).
 
Is my 24-bit workflow nonsense? Could all of this have been done at the 16-bit level with no possible way to tell the difference? Am I stupid in leaving the result at 24-bits and not downsampling it to 16, given that storage costs are negligible (e.g., 4 TB hard drive is $130)?
 
Let the thread experts speak!

 

[bigshot]

It's unlikely that bumping up to 24 bit makes any difference there, except with the correction of the DC offset. Where is that coming from? Do you have a bad sound card? The clipping correction is all operating up at the top of the volume range, not down by the noise floor where 24 bit would make a difference.

 

[esldude]

Well doing processing on sound at the 24 bit level undoubtedly has lower levels of artifacts from DSP done.  It may or may not be audible, but as you said, given the low cost of storage and your desire to perform these operations it makes pretty good sense to me to do it at 24 bit and have no worries over the results. 
 
Of course just for kicks, you could do one in 24 bit and one in 16.  When done ABX them to see if you hear a difference.  Myself I wouldn't bother, I would do it 24 bit like you are.

 

[SharpEars]

  It's unlikely that bumping up to 24 bit makes any difference there, except with the correction of the DC offset. Where is that coming from? Do you have a bad sound card? The clipping correction is all operating up at the top of the volume range, not down by the noise floor where 24 bit would make a difference.

 
The DC offset correction is due to the fact that most (especially pop) recording suffer a DC offset as can be measured over the entire track by any good audio editing program (e.g., Audacity, Audition, etc...). The same is true for determining whether or not there are any clipped samples. I try to make sure that the peak volume levels (not just RMS) do not exceed -0.25 dBFS.
 
How does 24-bit better 16-bit when it comes to DC offset neutralization as you mentioned in your reply?

 

[castleofargh]

  OK, I am going to stick my face into the arena while the kicks are flying and say that I have a reason to prefer 24-bit over 16-bit. Here is why:
 
My workflow consists of taking a 44.1/16 CD and ripping it into FLAC files. I then take the FLAC file for each track and:
 

• Upsample to 24-bits
• Remove DC Offset, if present
• Reduce the volume until there are no clipped samples present
• Possibly do heuristic based automated clip repair on those samples that were clipped
• (optional) Normalize the volume to a reasonable level inline with the other tracks on the CD

 
I perform functions 2-5 in the 24-bit domain and I leave the result as a 24-bit (FLAC) audio file that is ready for listening. I do this with all of my music and I stick with 24-bit, because I believe that the steps above performed in the 24-bit domain make more sense. I leave the resultant file as a 24-bit file, so that I can perform additional steps in the future that I deem will improve the tracks sound quality (e.g., equalization, excitement, whatever).
 
Is my 24-bit workflow nonsense? Could all of this have been done at the 16-bit level with no possible way to tell the difference? Am I stupid in leaving the result at 24-bits and not downsampling it to 16, given that storage costs are negligible (e.g., 4 TB hard drive is $130)?
 
Let the thread experts speak!

that's pretty much why studios work on 24bit (or maybe 32now?) you minimize the size of errors while processing stuff, and you get the dynamic margin to move everything without having to worry about crushing the lowest sounds. I don't think anybody here would prefer to mix in 16bit.
 
about going back to 16bit at the end or not. I have no doubt that there would be no audible difference one way or the other, so the choice is only laziness vs storage space ^_^.